diff options
| author | Android Chromium Automerger <chromium-automerger@android> | 2014-09-24 16:39:47 +0000 |
|---|---|---|
| committer | Android Chromium Automerger <chromium-automerger@android> | 2014-09-24 16:39:47 +0000 |
| commit | 8af00ea2e38cd722cdc20c41427cdab7e165591e (patch) | |
| tree | b14d4c5686ee0fc58d848cb6b00960c51595199d | |
| parent | 81b7993e9ee6acc9e003ee943373c90221ef6f70 (diff) | |
| parent | 291035ed1d8ec308ffbc81e9cd119e2f53f92f86 (diff) | |
| download | webrtc-8af00ea2e38cd722cdc20c41427cdab7e165591e.tar.gz | |
Merge third_party/webrtc from https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git at 291035ed1d8ec308ffbc81e9cd119e2f53f92f86
This commit was generated by merge_from_chromium.py.
Change-Id: I8de5d3b4724dd14ebda167e51683b554ddb5e024
125 files changed, 4134 insertions, 1334 deletions
diff --git a/base/BUILD.gn b/base/BUILD.gn index 003fc1bc..fbed1a52 100644 --- a/base/BUILD.gn +++ b/base/BUILD.gn @@ -127,6 +127,7 @@ static_library("rtc_base_approved") { "stringencode.h", "stringutils.cc", "stringutils.h", + "thread_annotations.h", "timeutils.cc", "timeutils.h", ] diff --git a/base/asynchttprequest_unittest.cc b/base/asynchttprequest_unittest.cc index 0bfd795b..4748de76 100644 --- a/base/asynchttprequest_unittest.cc +++ b/base/asynchttprequest_unittest.cc @@ -14,6 +14,7 @@ #include "webrtc/base/httpserver.h" #include "webrtc/base/socketstream.h" #include "webrtc/base/thread.h" +#include "webrtc/test/testsupport/gtest_disable.h" namespace rtc { @@ -126,7 +127,7 @@ class AsyncHttpRequestTest : public testing::Test, TestHttpServer server_; }; -TEST_F(AsyncHttpRequestTest, TestGetSuccess) { +TEST_F(AsyncHttpRequestTest, DISABLED_ON_MAC(TestGetSuccess)) { AsyncHttpRequest* req = CreateGetRequest( kServerHostnameAddr.hostname(), server().address().port(), kServerGetPath); @@ -143,7 +144,7 @@ TEST_F(AsyncHttpRequestTest, TestGetSuccess) { req->Release(); } -TEST_F(AsyncHttpRequestTest, TestGetNotFound) { +TEST_F(AsyncHttpRequestTest, DISABLED_ON_MAC(TestGetNotFound)) { AsyncHttpRequest* req = CreateGetRequest( kServerHostnameAddr.hostname(), server().address().port(), "/bad"); @@ -157,7 +158,7 @@ TEST_F(AsyncHttpRequestTest, TestGetNotFound) { req->Release(); } -TEST_F(AsyncHttpRequestTest, TestGetToNonServer) { +TEST_F(AsyncHttpRequestTest, DISABLED_ON_MAC(TestGetToNonServer)) { AsyncHttpRequest* req = CreateGetRequest( "127.0.0.1", server().address().port(), kServerGetPath); @@ -187,7 +188,7 @@ TEST_F(AsyncHttpRequestTest, DISABLED_TestGetToInvalidHostname) { req->Release(); } -TEST_F(AsyncHttpRequestTest, TestPostSuccess) { +TEST_F(AsyncHttpRequestTest, DISABLED_ON_MAC(TestPostSuccess)) { AsyncHttpRequest* req = CreatePostRequest( kServerHostnameAddr.hostname(), server().address().port(), kServerPostPath, "text/plain", new MemoryStream("abcd1234")); @@ -203,7 +204,7 @@ TEST_F(AsyncHttpRequestTest, TestPostSuccess) { } // Ensure that we shut down properly even if work is outstanding. -TEST_F(AsyncHttpRequestTest, TestCancel) { +TEST_F(AsyncHttpRequestTest, DISABLED_ON_MAC(TestCancel)) { AsyncHttpRequest* req = CreateGetRequest( kServerHostnameAddr.hostname(), server().address().port(), kServerGetPath); @@ -211,7 +212,7 @@ TEST_F(AsyncHttpRequestTest, TestCancel) { req->Destroy(true); } -TEST_F(AsyncHttpRequestTest, TestGetSuccessDelay) { +TEST_F(AsyncHttpRequestTest, DISABLED_ON_MAC(TestGetSuccessDelay)) { AsyncHttpRequest* req = CreateGetRequest( kServerHostnameAddr.hostname(), server().address().port(), kServerGetPath); diff --git a/base/autodetectproxy_unittest.cc b/base/autodetectproxy_unittest.cc index 80f220f2..782bc258 100644 --- a/base/autodetectproxy_unittest.cc +++ b/base/autodetectproxy_unittest.cc @@ -12,6 +12,7 @@ #include "webrtc/base/gunit.h" #include "webrtc/base/httpcommon.h" #include "webrtc/base/httpcommon-inl.h" +#include "webrtc/test/testsupport/gtest_disable.h" namespace rtc { @@ -99,26 +100,26 @@ class AutoDetectProxyTest : public testing::Test, public sigslot::has_slots<> { bool done_; }; -TEST_F(AutoDetectProxyTest, TestDetectUnresolvedProxy) { +TEST_F(AutoDetectProxyTest, DISABLED_ON_MAC(TestDetectUnresolvedProxy)) { TestCopesWithProxy(rtc::SocketAddress("localhost", 9999)); } -TEST_F(AutoDetectProxyTest, TestDetectUnresolvableProxy) { +TEST_F(AutoDetectProxyTest, DISABLED_ON_MAC(TestDetectUnresolvableProxy)) { TestCopesWithProxy(rtc::SocketAddress("invalid", 9999)); } -TEST_F(AutoDetectProxyTest, TestDetectIPv6Proxy) { +TEST_F(AutoDetectProxyTest, DISABLED_ON_MAC(TestDetectIPv6Proxy)) { TestCopesWithProxy(rtc::SocketAddress("::1", 9999)); } -TEST_F(AutoDetectProxyTest, TestDetectIPv4Proxy) { +TEST_F(AutoDetectProxyTest, DISABLED_ON_MAC(TestDetectIPv4Proxy)) { TestCopesWithProxy(rtc::SocketAddress("127.0.0.1", 9999)); } // Test that proxy detection completes successfully. (Does not actually verify // the correct detection result since we don't know what proxy to expect on an // arbitrary machine.) -TEST_F(AutoDetectProxyTest, TestProxyDetection) { +TEST_F(AutoDetectProxyTest, DISABLED_ON_MAC(TestProxyDetection)) { ASSERT_TRUE(Create(kUserAgent, kPath, kHost, diff --git a/base/base.gyp b/base/base.gyp index f5fad01f..fa92707c 100644 --- a/base/base.gyp +++ b/base/base.gyp @@ -50,6 +50,7 @@ 'stringencode.h', 'stringutils.cc', 'stringutils.h', + 'thread_annotations.h', 'timeutils.cc', 'timeutils.h', ], diff --git a/base/base_tests.gyp b/base/base_tests.gyp index 5ad3d2da..2d99e810 100644 --- a/base/base_tests.gyp +++ b/base/base_tests.gyp @@ -47,110 +47,111 @@ 'direct_dependent_settings': { 'sources': [ 'asynchttprequest_unittest.cc', - 'atomicops_unittest.cc', - 'autodetectproxy_unittest.cc', - 'bandwidthsmoother_unittest.cc', - 'base64_unittest.cc', - 'basictypes_unittest.cc', - 'bind_unittest.cc', - 'buffer_unittest.cc', - 'bytebuffer_unittest.cc', - 'byteorder_unittest.cc', - 'callback_unittest.cc', - 'cpumonitor_unittest.cc', - 'crc32_unittest.cc', - 'criticalsection_unittest.cc', - 'event_unittest.cc', - 'exp_filter_unittest.cc', - 'filelock_unittest.cc', - 'fileutils_unittest.cc', - 'helpers_unittest.cc', - 'httpbase_unittest.cc', - 'httpcommon_unittest.cc', - 'httpserver_unittest.cc', - 'ipaddress_unittest.cc', - 'logging_unittest.cc', - 'md5digest_unittest.cc', - 'messagedigest_unittest.cc', - 'messagequeue_unittest.cc', - 'multipart_unittest.cc', - 'nat_unittest.cc', - 'network_unittest.cc', - 'nullsocketserver_unittest.cc', - 'optionsfile_unittest.cc', - 'pathutils_unittest.cc', - 'physicalsocketserver_unittest.cc', - 'profiler_unittest.cc', - 'proxy_unittest.cc', - 'proxydetect_unittest.cc', - 'ratelimiter_unittest.cc', - 'ratetracker_unittest.cc', - 'referencecountedsingletonfactory_unittest.cc', - 'rollingaccumulator_unittest.cc', - 'scopedptrcollection_unittest.cc', - 'sha1digest_unittest.cc', - 'sharedexclusivelock_unittest.cc', - 'signalthread_unittest.cc', - 'sigslot_unittest.cc', - 'sigslottester.h', - 'sigslottester.h.pump', - 'socket_unittest.cc', - 'socket_unittest.h', - 'socketaddress_unittest.cc', - 'stream_unittest.cc', - 'stringencode_unittest.cc', - 'stringutils_unittest.cc', + #'atomicops_unittest.cc', + #'autodetectproxy_unittest.cc', + #'bandwidthsmoother_unittest.cc', + #'base64_unittest.cc', + #'basictypes_unittest.cc', + #'bind_unittest.cc', + #'buffer_unittest.cc', + #'bytebuffer_unittest.cc', + #'byteorder_unittest.cc', + #'callback_unittest.cc', + #'cpumonitor_unittest.cc', + #'crc32_unittest.cc', + #'criticalsection_unittest.cc', + #'event_unittest.cc', + #'exp_filter_unittest.cc', + #'filelock_unittest.cc', + #'fileutils_unittest.cc', + #'helpers_unittest.cc', + #'httpbase_unittest.cc', + #'httpcommon_unittest.cc', + #'httpserver_unittest.cc', + #'ipaddress_unittest.cc', + #'logging_unittest.cc', + #'md5digest_unittest.cc', + #'messagedigest_unittest.cc', + #'messagequeue_unittest.cc', + #'multipart_unittest.cc', + #'nat_unittest.cc', + #'network_unittest.cc', + #'nullsocketserver_unittest.cc', + #'optionsfile_unittest.cc', + #'pathutils_unittest.cc', + #'physicalsocketserver_unittest.cc', + #'profiler_unittest.cc', + #'proxy_unittest.cc', + #'proxydetect_unittest.cc', + #'ratelimiter_unittest.cc', + #'ratetracker_unittest.cc', + #'referencecountedsingletonfactory_unittest.cc', + #'rollingaccumulator_unittest.cc', + #'scopedptrcollection_unittest.cc', + #'sha1digest_unittest.cc', + #'sharedexclusivelock_unittest.cc', + #'signalthread_unittest.cc', + #'sigslot_unittest.cc', + #'sigslottester.h', + #'sigslottester.h.pump', + #'socket_unittest.cc', + #'socket_unittest.h', + #'socketaddress_unittest.cc', + #'stream_unittest.cc', + #'stringencode_unittest.cc', + #'stringutils_unittest.cc', # TODO(ronghuawu): Reenable this test. # 'systeminfo_unittest.cc', - 'task_unittest.cc', - 'testclient_unittest.cc', + #'task_unittest.cc', + #'testclient_unittest.cc', 'thread_checker_unittest.cc', - 'thread_unittest.cc', - 'timeutils_unittest.cc', - 'urlencode_unittest.cc', - 'versionparsing_unittest.cc', - 'virtualsocket_unittest.cc', + #'thread_unittest.cc', + #'timeutils_unittest.cc', + #'urlencode_unittest.cc', + #'versionparsing_unittest.cc', + #'virtualsocket_unittest.cc', # TODO(ronghuawu): Reenable this test. # 'windowpicker_unittest.cc', ], 'conditions': [ ['OS=="linux"', { 'sources': [ - 'latebindingsymboltable_unittest.cc', + #'latebindingsymboltable_unittest.cc', # TODO(ronghuawu): Reenable this test. # 'linux_unittest.cc', - 'linuxfdwalk_unittest.cc', + #'linuxfdwalk_unittest.cc', ], }], ['OS=="win"', { 'sources': [ - 'win32_unittest.cc', - 'win32regkey_unittest.cc', - 'win32socketserver_unittest.cc', - 'win32toolhelp_unittest.cc', - 'win32window_unittest.cc', - 'win32windowpicker_unittest.cc', - 'winfirewall_unittest.cc', + #'win32_unittest.cc', + #'win32regkey_unittest.cc', + #'win32socketserver_unittest.cc', + #'win32toolhelp_unittest.cc', + #'win32window_unittest.cc', + #'win32windowpicker_unittest.cc', + #'winfirewall_unittest.cc', ], 'sources!': [ # TODO(ronghuawu): Fix TestUdpReadyToSendIPv6 on windows bot # then reenable these tests. - 'physicalsocketserver_unittest.cc', - 'socket_unittest.cc', - 'win32socketserver_unittest.cc', - 'win32windowpicker_unittest.cc', + #'physicalsocketserver_unittest.cc', + #'socket_unittest.cc', + #'win32socketserver_unittest.cc', + #'win32windowpicker_unittest.cc', ], }], ['OS=="mac"', { 'sources': [ - 'macsocketserver_unittest.cc', - 'macutils_unittest.cc', + #'macsocketserver_unittest.cc', + #'macutils_unittest.cc', ], }], ['os_posix==1', { 'sources': [ - 'sslidentity_unittest.cc', - 'sslstreamadapter_unittest.cc', + #'ssladapter_unittest.cc', + #'sslidentity_unittest.cc', + #'sslstreamadapter_unittest.cc', ], }], ['OS=="ios" or (OS=="mac" and target_arch!="ia32")', { diff --git a/base/cpumonitor_unittest.cc b/base/cpumonitor_unittest.cc index 6d9af5ae..1546f558 100644 --- a/base/cpumonitor_unittest.cc +++ b/base/cpumonitor_unittest.cc @@ -23,6 +23,7 @@ #include "webrtc/base/thread.h" #include "webrtc/base/timeutils.h" #include "webrtc/base/timing.h" +#include "webrtc/test/testsupport/gtest_disable.h" namespace rtc { @@ -337,7 +338,7 @@ TEST(CpuMonitorTest, TestGetSystemLoadForceFallback) { #endif // Tests both process and system functions in use at same time. -TEST(CpuMonitorTest, TestGetBothLoad) { +TEST(CpuMonitorTest, DISABLED_ON_MAC(TestGetBothLoad)) { TestCpuSampler(true, true, false); } diff --git a/base/criticalsection.h b/base/criticalsection.h index a950a47f..a2d9bca0 100644 --- a/base/criticalsection.h +++ b/base/criticalsection.h @@ -12,6 +12,7 @@ #define WEBRTC_BASE_CRITICALSECTION_H__ #include "webrtc/base/constructormagic.h" +#include "webrtc/base/thread_annotations.h" #if defined(WEBRTC_WIN) #include "webrtc/base/win32.h" @@ -34,7 +35,7 @@ namespace rtc { #if defined(WEBRTC_WIN) -class CriticalSection { +class LOCKABLE CriticalSection { public: CriticalSection() { InitializeCriticalSection(&crit_); @@ -44,18 +45,18 @@ class CriticalSection { ~CriticalSection() { DeleteCriticalSection(&crit_); } - void Enter() { + void Enter() EXCLUSIVE_LOCK_FUNCTION() { EnterCriticalSection(&crit_); TRACK_OWNER(thread_ = GetCurrentThreadId()); } - bool TryEnter() { + bool TryEnter() EXCLUSIVE_TRYLOCK_FUNCTION(true) { if (TryEnterCriticalSection(&crit_) != FALSE) { TRACK_OWNER(thread_ = GetCurrentThreadId()); return true; } return false; } - void Leave() { + void Leave() UNLOCK_FUNCTION() { TRACK_OWNER(thread_ = 0); LeaveCriticalSection(&crit_); } @@ -71,7 +72,7 @@ class CriticalSection { #endif // WEBRTC_WIN #if defined(WEBRTC_POSIX) -class CriticalSection { +class LOCKABLE CriticalSection { public: CriticalSection() { pthread_mutexattr_t mutex_attribute; @@ -84,18 +85,18 @@ class CriticalSection { ~CriticalSection() { pthread_mutex_destroy(&mutex_); } - void Enter() { + void Enter() EXCLUSIVE_LOCK_FUNCTION() { pthread_mutex_lock(&mutex_); TRACK_OWNER(thread_ = pthread_self()); } - bool TryEnter() { + bool TryEnter() EXCLUSIVE_TRYLOCK_FUNCTION(true) { if (pthread_mutex_trylock(&mutex_) == 0) { TRACK_OWNER(thread_ = pthread_self()); return true; } return false; } - void Leave() { + void Leave() UNLOCK_FUNCTION() { TRACK_OWNER(thread_ = 0); pthread_mutex_unlock(&mutex_); } @@ -111,13 +112,13 @@ class CriticalSection { #endif // WEBRTC_POSIX // CritScope, for serializing execution through a scope. -class CritScope { +class SCOPED_LOCKABLE CritScope { public: - explicit CritScope(CriticalSection *pcrit) { + explicit CritScope(CriticalSection *pcrit) EXCLUSIVE_LOCK_FUNCTION(pcrit) { pcrit_ = pcrit; pcrit_->Enter(); } - ~CritScope() { + ~CritScope() UNLOCK_FUNCTION() { pcrit_->Leave(); } private: diff --git a/base/criticalsection_unittest.cc b/base/criticalsection_unittest.cc index e1b05cb0..69f74fa2 100644 --- a/base/criticalsection_unittest.cc +++ b/base/criticalsection_unittest.cc @@ -16,6 +16,7 @@ #include "webrtc/base/gunit.h" #include "webrtc/base/scopedptrcollection.h" #include "webrtc/base/thread.h" +#include "webrtc/test/testsupport/gtest_disable.h" namespace rtc { @@ -119,7 +120,7 @@ TEST(AtomicOpsTest, Simple) { EXPECT_EQ(0, value); } -TEST(AtomicOpsTest, Increment) { +TEST(AtomicOpsTest, DISABLED_ON_MAC(Increment)) { // Create and start lots of threads. AtomicOpRunner<IncrementOp> runner(0); ScopedPtrCollection<Thread> threads; @@ -131,7 +132,7 @@ TEST(AtomicOpsTest, Increment) { EXPECT_EQ(kOperationsToRun * kNumThreads, runner.value()); } -TEST(AtomicOpsTest, Decrement) { +TEST(AtomicOpsTest, DISABLED_ON_MAC(Decrement)) { // Create and start lots of threads. AtomicOpRunner<DecrementOp> runner(kOperationsToRun * kNumThreads); ScopedPtrCollection<Thread> threads; diff --git a/base/filelock_unittest.cc b/base/filelock_unittest.cc index eecbf07d..536204ec 100644 --- a/base/filelock_unittest.cc +++ b/base/filelock_unittest.cc @@ -17,6 +17,7 @@ #include "webrtc/base/pathutils.h" #include "webrtc/base/scoped_ptr.h" #include "webrtc/base/thread.h" +#include "webrtc/test/testsupport/gtest_disable.h" namespace rtc { @@ -76,7 +77,7 @@ TEST_F(FileLockTest, TestLockX2) { EXPECT_TRUE(lock2.get() == NULL); } -TEST_F(FileLockTest, TestThreadedLock) { +TEST_F(FileLockTest, DISABLED_ON_MAC(TestThreadedLock)) { scoped_ptr<FileLock> lock(FileLock::TryLock(temp_file_.pathname())); EXPECT_TRUE(lock.get() != NULL); diff --git a/base/logging_unittest.cc b/base/logging_unittest.cc index 59630d74..aa438376 100644 --- a/base/logging_unittest.cc +++ b/base/logging_unittest.cc @@ -14,6 +14,7 @@ #include "webrtc/base/pathutils.h" #include "webrtc/base/stream.h" #include "webrtc/base/thread.h" +#include "webrtc/test/testsupport/gtest_disable.h" namespace rtc { @@ -82,7 +83,7 @@ class LogThread : public Thread { } }; -TEST(LogTest, MultipleThreads) { +TEST(LogTest, DISABLED_ON_MAC(MultipleThreads)) { int sev = LogMessage::GetLogToStream(NULL); LogThread thread1, thread2, thread3; diff --git a/base/macsocketserver_unittest.cc b/base/macsocketserver_unittest.cc index e98be918..4c0432af 100644 --- a/base/macsocketserver_unittest.cc +++ b/base/macsocketserver_unittest.cc @@ -33,7 +33,7 @@ class WakeThread : public Thread { #ifndef CARBON_DEPRECATED // Test that MacCFSocketServer::Wait works as expected. -TEST(MacCFSocketServerTest, TestWait) { +TEST(MacCFSocketServerTest, DISABLED_TestWait) { MacCFSocketServer server; uint32 start = Time(); server.Wait(1000, true); @@ -41,7 +41,7 @@ TEST(MacCFSocketServerTest, TestWait) { } // Test that MacCFSocketServer::Wakeup works as expected. -TEST(MacCFSocketServerTest, TestWakeup) { +TEST(MacCFSocketServerTest, DISABLED_TestWakeup) { MacCFSocketServer server; WakeThread thread(&server); uint32 start = Time(); @@ -51,7 +51,7 @@ TEST(MacCFSocketServerTest, TestWakeup) { } // Test that MacCarbonSocketServer::Wait works as expected. -TEST(MacCarbonSocketServerTest, TestWait) { +TEST(MacCarbonSocketServerTest, DISABLED_TestWait) { MacCarbonSocketServer server; uint32 start = Time(); server.Wait(1000, true); @@ -59,7 +59,7 @@ TEST(MacCarbonSocketServerTest, TestWait) { } // Test that MacCarbonSocketServer::Wakeup works as expected. -TEST(MacCarbonSocketServerTest, TestWakeup) { +TEST(MacCarbonSocketServerTest, DISABLED_TestWakeup) { MacCarbonSocketServer server; WakeThread thread(&server); uint32 start = Time(); @@ -69,7 +69,7 @@ TEST(MacCarbonSocketServerTest, TestWakeup) { } // Test that MacCarbonAppSocketServer::Wait works as expected. -TEST(MacCarbonAppSocketServerTest, TestWait) { +TEST(MacCarbonAppSocketServerTest, DISABLED_TestWait) { MacCarbonAppSocketServer server; uint32 start = Time(); server.Wait(1000, true); @@ -77,7 +77,7 @@ TEST(MacCarbonAppSocketServerTest, TestWait) { } // Test that MacCarbonAppSocketServer::Wakeup works as expected. -TEST(MacCarbonAppSocketServerTest, TestWakeup) { +TEST(MacCarbonAppSocketServerTest, DISABLED_TestWakeup) { MacCarbonAppSocketServer server; WakeThread thread(&server); uint32 start = Time(); @@ -102,19 +102,19 @@ class MacAsyncSocketTest : public SocketTest { SocketServerScope scope_; }; -TEST_F(MacAsyncSocketTest, TestConnectIPv4) { +TEST_F(MacAsyncSocketTest, DISABLED_TestConnectIPv4) { SocketTest::TestConnectIPv4(); } -TEST_F(MacAsyncSocketTest, TestConnectIPv6) { +TEST_F(MacAsyncSocketTest, DISABLED_TestConnectIPv6) { SocketTest::TestConnectIPv6(); } -TEST_F(MacAsyncSocketTest, TestConnectWithDnsLookupIPv4) { +TEST_F(MacAsyncSocketTest, DISABLED_TestConnectWithDnsLookupIPv4) { SocketTest::TestConnectWithDnsLookupIPv4(); } -TEST_F(MacAsyncSocketTest, TestConnectWithDnsLookupIPv6) { +TEST_F(MacAsyncSocketTest, DISABLED_TestConnectWithDnsLookupIPv6) { SocketTest::TestConnectWithDnsLookupIPv6(); } @@ -123,7 +123,7 @@ TEST_F(MacAsyncSocketTest, DISABLED_TestConnectFailIPv4) { SocketTest::TestConnectFailIPv4(); } -TEST_F(MacAsyncSocketTest, TestConnectFailIPv6) { +TEST_F(MacAsyncSocketTest, DISABLED_TestConnectFailIPv6) { SocketTest::TestConnectFailIPv6(); } @@ -136,11 +136,11 @@ TEST_F(MacAsyncSocketTest, DISABLED_TestConnectWithDnsLookupFailIPv6) { SocketTest::TestConnectWithDnsLookupFailIPv6(); } -TEST_F(MacAsyncSocketTest, TestConnectWithClosedSocketIPv4) { +TEST_F(MacAsyncSocketTest, DISABLED_TestConnectWithClosedSocketIPv4) { SocketTest::TestConnectWithClosedSocketIPv4(); } -TEST_F(MacAsyncSocketTest, TestConnectWithClosedSocketIPv6) { +TEST_F(MacAsyncSocketTest, DISABLED_TestConnectWithClosedSocketIPv6) { SocketTest::TestConnectWithClosedSocketIPv6(); } @@ -155,51 +155,51 @@ TEST_F(MacAsyncSocketTest, DISABLED_TestServerCloseDuringConnectIPv6) { } // Flaky at the moment (0.5% failure rate). Seems the client doesn't get // signalled in a timely manner... -TEST_F(MacAsyncSocketTest, TestClientCloseDuringConnectIPv4) { +TEST_F(MacAsyncSocketTest, DISABLED_TestClientCloseDuringConnectIPv4) { SocketTest::TestClientCloseDuringConnectIPv4(); } -TEST_F(MacAsyncSocketTest, TestClientCloseDuringConnectIPv6) { +TEST_F(MacAsyncSocketTest, DISABLED_TestClientCloseDuringConnectIPv6) { SocketTest::TestClientCloseDuringConnectIPv6(); } -TEST_F(MacAsyncSocketTest, TestServerCloseIPv4) { +TEST_F(MacAsyncSocketTest, DISABLED_TestServerCloseIPv4) { SocketTest::TestServerCloseIPv4(); } -TEST_F(MacAsyncSocketTest, TestServerCloseIPv6) { +TEST_F(MacAsyncSocketTest, DISABLED_TestServerCloseIPv6) { SocketTest::TestServerCloseIPv6(); } -TEST_F(MacAsyncSocketTest, TestCloseInClosedCallbackIPv4) { +TEST_F(MacAsyncSocketTest, DISABLED_TestCloseInClosedCallbackIPv4) { SocketTest::TestCloseInClosedCallbackIPv4(); } -TEST_F(MacAsyncSocketTest, TestCloseInClosedCallbackIPv6) { +TEST_F(MacAsyncSocketTest, DISABLED_TestCloseInClosedCallbackIPv6) { SocketTest::TestCloseInClosedCallbackIPv6(); } -TEST_F(MacAsyncSocketTest, TestSocketServerWaitIPv4) { +TEST_F(MacAsyncSocketTest, DISABLED_TestSocketServerWaitIPv4) { SocketTest::TestSocketServerWaitIPv4(); } -TEST_F(MacAsyncSocketTest, TestSocketServerWaitIPv6) { +TEST_F(MacAsyncSocketTest, DISABLED_TestSocketServerWaitIPv6) { SocketTest::TestSocketServerWaitIPv6(); } -TEST_F(MacAsyncSocketTest, TestTcpIPv4) { +TEST_F(MacAsyncSocketTest, DISABLED_TestTcpIPv4) { SocketTest::TestTcpIPv4(); } -TEST_F(MacAsyncSocketTest, TestTcpIPv6) { +TEST_F(MacAsyncSocketTest, DISABLED_TestTcpIPv6) { SocketTest::TestTcpIPv6(); } -TEST_F(MacAsyncSocketTest, TestSingleFlowControlCallbackIPv4) { +TEST_F(MacAsyncSocketTest, DISABLED_TestSingleFlowControlCallbackIPv4) { SocketTest::TestSingleFlowControlCallbackIPv4(); } -TEST_F(MacAsyncSocketTest, TestSingleFlowControlCallbackIPv6) { +TEST_F(MacAsyncSocketTest, DISABLED_TestSingleFlowControlCallbackIPv6) { SocketTest::TestSingleFlowControlCallbackIPv6(); } @@ -226,11 +226,11 @@ class MacCarbonAppAsyncSocketTest : public MacAsyncSocketTest { }; }; -TEST_F(MacCarbonAppAsyncSocketTest, TestSocketServerWaitIPv4) { +TEST_F(MacCarbonAppAsyncSocketTest, DISABLED_TestSocketServerWaitIPv4) { SocketTest::TestSocketServerWaitIPv4(); } -TEST_F(MacCarbonAppAsyncSocketTest, TestSocketServerWaitIPv6) { +TEST_F(MacCarbonAppAsyncSocketTest, DISABLED_TestSocketServerWaitIPv6) { SocketTest::TestSocketServerWaitIPv6(); } #endif diff --git a/base/messagequeue_unittest.cc b/base/messagequeue_unittest.cc index 78024e0b..2027d91e 100644 --- a/base/messagequeue_unittest.cc +++ b/base/messagequeue_unittest.cc @@ -16,6 +16,7 @@ #include "webrtc/base/thread.h" #include "webrtc/base/timeutils.h" #include "webrtc/base/nullsocketserver.h" +#include "webrtc/test/testsupport/gtest_disable.h" using namespace rtc; @@ -79,7 +80,7 @@ TEST_F(MessageQueueTest, DelayedPostsWithIdenticalTimesAreProcessedInFifoOrder(&q_nullss); } -TEST_F(MessageQueueTest, DisposeNotLocked) { +TEST_F(MessageQueueTest, DISABLED_ON_MAC(DisposeNotLocked)) { bool was_locked = true; bool deleted = false; DeletedLockChecker* d = new DeletedLockChecker(this, &was_locked, &deleted); diff --git a/base/nat_unittest.cc b/base/nat_unittest.cc index 0e16259a..19c1225c 100644 --- a/base/nat_unittest.cc +++ b/base/nat_unittest.cc @@ -19,6 +19,7 @@ #include "webrtc/base/physicalsocketserver.h" #include "webrtc/base/testclient.h" #include "webrtc/base/virtualsocketserver.h" +#include "webrtc/test/testsupport/gtest_disable.h" using namespace rtc; @@ -236,11 +237,11 @@ void TestPhysicalInternal(const SocketAddress& int_addr) { TestFilters(int_pss.get(), int_addr, ext_pss.get(), ext_addrs); } -TEST(NatTest, TestPhysicalIPv4) { +TEST(NatTest, DISABLED_ON_MAC(TestPhysicalIPv4)) { TestPhysicalInternal(SocketAddress("127.0.0.1", 0)); } -TEST(NatTest, TestPhysicalIPv6) { +TEST(NatTest, DISABLED_ON_MAC(TestPhysicalIPv6)) { if (HasIPv6Enabled()) { TestPhysicalInternal(SocketAddress("::1", 0)); } else { @@ -278,11 +279,11 @@ void TestVirtualInternal(int family) { TestFilters(int_vss.get(), int_addr, ext_vss.get(), ext_addrs); } -TEST(NatTest, TestVirtualIPv4) { +TEST(NatTest, DISABLED_ON_MAC(TestVirtualIPv4)) { TestVirtualInternal(AF_INET); } -TEST(NatTest, TestVirtualIPv6) { +TEST(NatTest, DISABLED_ON_MAC(TestVirtualIPv6)) { if (HasIPv6Enabled()) { TestVirtualInternal(AF_INET6); } else { diff --git a/base/nssstreamadapter.cc b/base/nssstreamadapter.cc index f4b2d311..40c017fb 100644 --- a/base/nssstreamadapter.cc +++ b/base/nssstreamadapter.cc @@ -486,6 +486,8 @@ int NSSStreamAdapter::BeginSSL() { return -1; } + // TODO(juberti): Check for client_auth_enabled() + rv = SSL_OptionSet(ssl_fd_, SSL_REQUIRE_CERTIFICATE, PR_TRUE); if (rv != SECSuccess) { Error("BeginSSL", -1, false); diff --git a/base/nullsocketserver_unittest.cc b/base/nullsocketserver_unittest.cc index fe21f6ad..6d3eb52b 100644 --- a/base/nullsocketserver_unittest.cc +++ b/base/nullsocketserver_unittest.cc @@ -10,6 +10,7 @@ #include "webrtc/base/gunit.h" #include "webrtc/base/nullsocketserver.h" +#include "webrtc/test/testsupport/gtest_disable.h" namespace rtc { @@ -27,7 +28,7 @@ class NullSocketServerTest NullSocketServer ss_; }; -TEST_F(NullSocketServerTest, WaitAndSet) { +TEST_F(NullSocketServerTest, DISABLED_ON_MAC(WaitAndSet)) { Thread thread; EXPECT_TRUE(thread.Start()); thread.Post(this, 0); diff --git a/base/opensslstreamadapter.cc b/base/opensslstreamadapter.cc index ed5ac740..070a948b 100644 --- a/base/opensslstreamadapter.cc +++ b/base/opensslstreamadapter.cc @@ -743,8 +743,15 @@ SSL_CTX* OpenSSLStreamAdapter::SetupSSLContext() { SSL_CTX_set_info_callback(ctx, OpenSSLAdapter::SSLInfoCallback); #endif - SSL_CTX_set_verify(ctx, SSL_VERIFY_PEER |SSL_VERIFY_FAIL_IF_NO_PEER_CERT, - SSLVerifyCallback); + int mode = SSL_VERIFY_PEER; + if (client_auth_enabled()) { + // Require a certificate from the client. + // Note: Normally this is always true in production, but it may be disabled + // for testing purposes (e.g. SSLAdapter unit tests). + mode |= SSL_VERIFY_FAIL_IF_NO_PEER_CERT; + } + + SSL_CTX_set_verify(ctx, mode, SSLVerifyCallback); SSL_CTX_set_verify_depth(ctx, 4); SSL_CTX_set_cipher_list(ctx, "ALL:!ADH:!LOW:!EXP:!MD5:@STRENGTH"); diff --git a/base/physicalsocketserver_unittest.cc b/base/physicalsocketserver_unittest.cc index df750adb..4d57e730 100644 --- a/base/physicalsocketserver_unittest.cc +++ b/base/physicalsocketserver_unittest.cc @@ -18,6 +18,7 @@ #include "webrtc/base/socket_unittest.h" #include "webrtc/base/testutils.h" #include "webrtc/base/thread.h" +#include "webrtc/test/testsupport/gtest_disable.h" namespace rtc { @@ -106,11 +107,11 @@ TEST_F(PhysicalSocketTest, TestCloseInClosedCallbackIPv6) { SocketTest::TestCloseInClosedCallbackIPv6(); } -TEST_F(PhysicalSocketTest, TestSocketServerWaitIPv4) { +TEST_F(PhysicalSocketTest, DISABLED_ON_MAC(TestSocketServerWaitIPv4)) { SocketTest::TestSocketServerWaitIPv4(); } -TEST_F(PhysicalSocketTest, TestSocketServerWaitIPv6) { +TEST_F(PhysicalSocketTest, DISABLED_ON_MAC(TestSocketServerWaitIPv6)) { SocketTest::TestSocketServerWaitIPv6(); } @@ -206,7 +207,7 @@ Thread *PosixSignalDeliveryTest::signaled_thread_ = NULL; // Test receiving a synchronous signal while not in Wait() and then entering // Wait() afterwards. -TEST_F(PosixSignalDeliveryTest, RaiseThenWait) { +TEST_F(PosixSignalDeliveryTest, DISABLED_ON_MAC(RaiseThenWait)) { ASSERT_TRUE(ss_->SetPosixSignalHandler(SIGTERM, &RecordSignal)); raise(SIGTERM); EXPECT_TRUE(ss_->Wait(0, true)); @@ -216,7 +217,7 @@ TEST_F(PosixSignalDeliveryTest, RaiseThenWait) { // Test that we can handle getting tons of repeated signals and that we see all // the different ones. -TEST_F(PosixSignalDeliveryTest, InsanelyManySignals) { +TEST_F(PosixSignalDeliveryTest, DISABLED_ON_MAC(InsanelyManySignals)) { ss_->SetPosixSignalHandler(SIGTERM, &RecordSignal); ss_->SetPosixSignalHandler(SIGINT, &RecordSignal); for (int i = 0; i < 10000; ++i) { @@ -231,7 +232,7 @@ TEST_F(PosixSignalDeliveryTest, InsanelyManySignals) { } // Test that a signal during a Wait() call is detected. -TEST_F(PosixSignalDeliveryTest, SignalDuringWait) { +TEST_F(PosixSignalDeliveryTest, DISABLED_ON_MAC(SignalDuringWait)) { ss_->SetPosixSignalHandler(SIGALRM, &RecordSignal); alarm(1); EXPECT_TRUE(ss_->Wait(1500, true)); @@ -256,7 +257,7 @@ class RaiseSigTermRunnable : public Runnable { // Test that it works no matter what thread the kernel chooses to give the // signal to (since it's not guaranteed to be the one that Wait() runs on). -TEST_F(PosixSignalDeliveryTest, SignalOnDifferentThread) { +TEST_F(PosixSignalDeliveryTest, DISABLED_ON_MAC(SignalOnDifferentThread)) { ss_->SetPosixSignalHandler(SIGTERM, &RecordSignal); // Mask out SIGTERM so that it can't be delivered to this thread. sigset_t mask; diff --git a/base/proxy_unittest.cc b/base/proxy_unittest.cc index d8a523fe..a1ade0c7 100644 --- a/base/proxy_unittest.cc +++ b/base/proxy_unittest.cc @@ -17,6 +17,7 @@ #include "webrtc/base/testclient.h" #include "webrtc/base/testechoserver.h" #include "webrtc/base/virtualsocketserver.h" +#include "webrtc/test/testsupport/gtest_disable.h" using rtc::Socket; using rtc::Thread; @@ -74,7 +75,7 @@ class ProxyTest : public testing::Test { }; // Tests whether we can use a SOCKS5 proxy to connect to a server. -TEST_F(ProxyTest, TestSocks5Connect) { +TEST_F(ProxyTest, DISABLED_ON_MAC(TestSocks5Connect)) { rtc::AsyncSocket* socket = ss()->CreateAsyncSocket(kSocksProxyIntAddr.family(), SOCK_STREAM); rtc::AsyncSocksProxySocket* proxy_socket = @@ -118,7 +119,7 @@ TEST_F(ProxyTest, TestHttpsConnect) { */ // Tests whether we can autodetect a SOCKS5 proxy. -TEST_F(ProxyTest, TestAutoDetectSocks5) { +TEST_F(ProxyTest, DISABLED_ON_MAC(TestAutoDetectSocks5)) { EXPECT_EQ(rtc::PROXY_SOCKS5, DetectProxyType(kSocksProxyIntAddr)); } @@ -130,6 +131,6 @@ TEST_F(ProxyTest, TestAutoDetectHttps) { */ // Tests whether we fail properly for no proxy. -TEST_F(ProxyTest, TestAutoDetectBogus) { +TEST_F(ProxyTest, DISABLED_ON_MAC(TestAutoDetectBogus)) { EXPECT_EQ(rtc::PROXY_UNKNOWN, DetectProxyType(kBogusProxyIntAddr)); } diff --git a/base/sharedexclusivelock.h b/base/sharedexclusivelock.h index f64d7cf5..aaaba3b8 100644 --- a/base/sharedexclusivelock.h +++ b/base/sharedexclusivelock.h @@ -19,14 +19,14 @@ namespace rtc { // This class provides shared-exclusive lock. It can be used in cases like // multiple-readers/single-writer model. -class SharedExclusiveLock { +class LOCKABLE SharedExclusiveLock { public: SharedExclusiveLock(); // Locking/unlocking methods. It is encouraged to use SharedScope or // ExclusiveScope for protection. - void LockExclusive(); - void UnlockExclusive(); + void LockExclusive() EXCLUSIVE_LOCK_FUNCTION(); + void UnlockExclusive() UNLOCK_FUNCTION(); void LockShared(); void UnlockShared(); @@ -39,15 +39,14 @@ class SharedExclusiveLock { DISALLOW_COPY_AND_ASSIGN(SharedExclusiveLock); }; -class SharedScope { +class SCOPED_LOCKABLE SharedScope { public: - explicit SharedScope(SharedExclusiveLock* lock) : lock_(lock) { + explicit SharedScope(SharedExclusiveLock* lock) SHARED_LOCK_FUNCTION(lock) + : lock_(lock) { lock_->LockShared(); } - ~SharedScope() { - lock_->UnlockShared(); - } + ~SharedScope() UNLOCK_FUNCTION() { lock_->UnlockShared(); } private: SharedExclusiveLock* lock_; @@ -55,15 +54,15 @@ class SharedScope { DISALLOW_COPY_AND_ASSIGN(SharedScope); }; -class ExclusiveScope { +class SCOPED_LOCKABLE ExclusiveScope { public: - explicit ExclusiveScope(SharedExclusiveLock* lock) : lock_(lock) { + explicit ExclusiveScope(SharedExclusiveLock* lock) + EXCLUSIVE_LOCK_FUNCTION(lock) + : lock_(lock) { lock_->LockExclusive(); } - ~ExclusiveScope() { - lock_->UnlockExclusive(); - } + ~ExclusiveScope() UNLOCK_FUNCTION() { lock_->UnlockExclusive(); } private: SharedExclusiveLock* lock_; diff --git a/base/sharedexclusivelock_unittest.cc b/base/sharedexclusivelock_unittest.cc index 42334af7..ea406cf9 100644 --- a/base/sharedexclusivelock_unittest.cc +++ b/base/sharedexclusivelock_unittest.cc @@ -16,6 +16,7 @@ #include "webrtc/base/sharedexclusivelock.h" #include "webrtc/base/thread.h" #include "webrtc/base/timeutils.h" +#include "webrtc/test/testsupport/gtest_disable.h" namespace rtc { @@ -157,7 +158,7 @@ TEST_F(SharedExclusiveLockTest, DISABLED_TestSharedShared) { EXPECT_LE(reader1.waiting_time_in_ms(), kNoWaitThresholdInMs); } -TEST_F(SharedExclusiveLockTest, TestSharedExclusive) { +TEST_F(SharedExclusiveLockTest, DISABLED_ON_MAC(TestSharedExclusive)) { bool done; WriteTask writer(shared_exclusive_lock_.get(), &value_, &done); @@ -176,7 +177,7 @@ TEST_F(SharedExclusiveLockTest, TestSharedExclusive) { EXPECT_GE(writer.waiting_time_in_ms(), kWaitThresholdInMs); } -TEST_F(SharedExclusiveLockTest, TestExclusiveShared) { +TEST_F(SharedExclusiveLockTest, DISABLED_ON_MAC(TestExclusiveShared)) { int value; bool done; ReadTask reader(shared_exclusive_lock_.get(), &value_, &done); @@ -196,7 +197,7 @@ TEST_F(SharedExclusiveLockTest, TestExclusiveShared) { EXPECT_GE(reader.waiting_time_in_ms(), kWaitThresholdInMs); } -TEST_F(SharedExclusiveLockTest, TestExclusiveExclusive) { +TEST_F(SharedExclusiveLockTest, DISABLED_ON_MAC(TestExclusiveExclusive)) { bool done; WriteTask writer(shared_exclusive_lock_.get(), &value_, &done); diff --git a/base/signalthread.h b/base/signalthread.h index a97bda1a..8e18be61 100644 --- a/base/signalthread.h +++ b/base/signalthread.h @@ -115,16 +115,17 @@ class SignalThread DISALLOW_IMPLICIT_CONSTRUCTORS(Worker); }; - class EnterExit { + class SCOPED_LOCKABLE EnterExit { public: - explicit EnterExit(SignalThread* t) : t_(t) { + explicit EnterExit(SignalThread* t) EXCLUSIVE_LOCK_FUNCTION(t->cs_) + : t_(t) { t_->cs_.Enter(); // If refcount_ is zero then the object has already been deleted and we // will be double-deleting it in ~EnterExit()! (shouldn't happen) ASSERT(t_->refcount_ != 0); ++t_->refcount_; } - ~EnterExit() { + ~EnterExit() UNLOCK_FUNCTION() { bool d = (0 == --t_->refcount_); t_->cs_.Leave(); if (d) diff --git a/base/signalthread_unittest.cc b/base/signalthread_unittest.cc index e0ea54eb..3f534312 100644 --- a/base/signalthread_unittest.cc +++ b/base/signalthread_unittest.cc @@ -11,6 +11,7 @@ #include "webrtc/base/gunit.h" #include "webrtc/base/signalthread.h" #include "webrtc/base/thread.h" +#include "webrtc/test/testsupport/gtest_disable.h" using namespace rtc; @@ -133,7 +134,7 @@ class OwnerThread : public Thread, public sigslot::has_slots<> { // Test for when the main thread goes away while the // signal thread is still working. This may happen // when shutting down the process. -TEST_F(SignalThreadTest, OwnerThreadGoesAway) { +TEST_F(SignalThreadTest, DISABLED_ON_MAC(OwnerThreadGoesAway)) { { scoped_ptr<OwnerThread> owner(new OwnerThread(this)); main_thread_ = owner.get(); @@ -156,7 +157,7 @@ TEST_F(SignalThreadTest, OwnerThreadGoesAway) { EXPECT_EQ(stopped, thread_stopped_); \ EXPECT_EQ(deleted, thread_deleted_); -TEST_F(SignalThreadTest, ThreadFinishes) { +TEST_F(SignalThreadTest, DISABLED_ON_MAC(ThreadFinishes)) { thread_->Start(); EXPECT_STATE(1, 0, 0, 0, 0); Thread::SleepMs(500); @@ -165,7 +166,7 @@ TEST_F(SignalThreadTest, ThreadFinishes) { EXPECT_STATE(1, 1, 1, 0, 1); } -TEST_F(SignalThreadTest, ReleasedThreadFinishes) { +TEST_F(SignalThreadTest, DISABLED_ON_MAC(ReleasedThreadFinishes)) { thread_->Start(); EXPECT_STATE(1, 0, 0, 0, 0); thread_->Release(); @@ -177,7 +178,7 @@ TEST_F(SignalThreadTest, ReleasedThreadFinishes) { EXPECT_STATE(1, 1, 1, 0, 1); } -TEST_F(SignalThreadTest, DestroyedThreadCleansUp) { +TEST_F(SignalThreadTest, DISABLED_ON_MAC(DestroyedThreadCleansUp)) { thread_->Start(); EXPECT_STATE(1, 0, 0, 0, 0); thread_->Destroy(true); @@ -186,7 +187,7 @@ TEST_F(SignalThreadTest, DestroyedThreadCleansUp) { EXPECT_STATE(1, 0, 0, 1, 1); } -TEST_F(SignalThreadTest, DeferredDestroyedThreadCleansUp) { +TEST_F(SignalThreadTest, DISABLED_ON_MAC(DeferredDestroyedThreadCleansUp)) { thread_->Start(); EXPECT_STATE(1, 0, 0, 0, 0); thread_->Destroy(false); diff --git a/base/ssladapter_unittest.cc b/base/ssladapter_unittest.cc new file mode 100644 index 00000000..6d4536d2 --- /dev/null +++ b/base/ssladapter_unittest.cc @@ -0,0 +1,342 @@ +/* + * Copyright 2014 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#include <string> + +#include "webrtc/base/gunit.h" +#include "webrtc/base/ipaddress.h" +#include "webrtc/base/socketstream.h" +#include "webrtc/base/ssladapter.h" +#include "webrtc/base/sslstreamadapter.h" +#include "webrtc/base/stream.h" +#include "webrtc/base/virtualsocketserver.h" + +static const int kTimeout = 5000; + +static rtc::AsyncSocket* CreateSocket(const rtc::SSLMode& ssl_mode) { + rtc::SocketAddress address(rtc::IPAddress(INADDR_ANY), 0); + + rtc::AsyncSocket* socket = rtc::Thread::Current()-> + socketserver()->CreateAsyncSocket( + address.family(), (ssl_mode == rtc::SSL_MODE_DTLS) ? + SOCK_DGRAM : SOCK_STREAM); + socket->Bind(address); + + return socket; +} + +static std::string GetSSLProtocolName(const rtc::SSLMode& ssl_mode) { + return (ssl_mode == rtc::SSL_MODE_DTLS) ? "DTLS" : "TLS"; +} + +class SSLAdapterTestDummyClient : public sigslot::has_slots<> { + public: + explicit SSLAdapterTestDummyClient(const rtc::SSLMode& ssl_mode) + : ssl_mode_(ssl_mode) { + rtc::AsyncSocket* socket = CreateSocket(ssl_mode_); + + ssl_adapter_.reset(rtc::SSLAdapter::Create(socket)); + + // Ignore any certificate errors for the purpose of testing. + // Note: We do this only because we don't have a real certificate. + // NEVER USE THIS IN PRODUCTION CODE! + ssl_adapter_->set_ignore_bad_cert(true); + + ssl_adapter_->SignalReadEvent.connect(this, + &SSLAdapterTestDummyClient::OnSSLAdapterReadEvent); + ssl_adapter_->SignalCloseEvent.connect(this, + &SSLAdapterTestDummyClient::OnSSLAdapterCloseEvent); + } + + rtc::AsyncSocket::ConnState GetState() const { + return ssl_adapter_->GetState(); + } + + const std::string& GetReceivedData() const { + return data_; + } + + int Connect(const std::string& hostname, const rtc::SocketAddress& address) { + LOG(LS_INFO) << "Starting " << GetSSLProtocolName(ssl_mode_) + << " handshake with " << hostname; + + if (ssl_adapter_->StartSSL(hostname.c_str(), false) != 0) { + return -1; + } + + LOG(LS_INFO) << "Initiating connection with " << address; + + return ssl_adapter_->Connect(address); + } + + int Close() { + return ssl_adapter_->Close(); + } + + int Send(const std::string& message) { + LOG(LS_INFO) << "Client sending '" << message << "'"; + + return ssl_adapter_->Send(message.data(), message.length()); + } + + void OnSSLAdapterReadEvent(rtc::AsyncSocket* socket) { + char buffer[4096] = ""; + + // Read data received from the server and store it in our internal buffer. + int read = socket->Recv(buffer, sizeof(buffer) - 1); + if (read != -1) { + buffer[read] = '\0'; + + LOG(LS_INFO) << "Client received '" << buffer << "'"; + + data_ += buffer; + } + } + + void OnSSLAdapterCloseEvent(rtc::AsyncSocket* socket, int error) { + // OpenSSLAdapter signals handshake failure with a close event, but without + // closing the socket! Let's close the socket here. This way GetState() can + // return CS_CLOSED after failure. + if (socket->GetState() != rtc::AsyncSocket::CS_CLOSED) { + socket->Close(); + } + } + + private: + const rtc::SSLMode ssl_mode_; + + rtc::scoped_ptr<rtc::SSLAdapter> ssl_adapter_; + + std::string data_; +}; + +class SSLAdapterTestDummyServer : public sigslot::has_slots<> { + public: + explicit SSLAdapterTestDummyServer(const rtc::SSLMode& ssl_mode) + : ssl_mode_(ssl_mode) { + // Generate a key pair and a certificate for this host. + ssl_identity_.reset(rtc::SSLIdentity::Generate(GetHostname())); + + server_socket_.reset(CreateSocket(ssl_mode_)); + + server_socket_->SignalReadEvent.connect(this, + &SSLAdapterTestDummyServer::OnServerSocketReadEvent); + + server_socket_->Listen(1); + + LOG(LS_INFO) << ((ssl_mode_ == rtc::SSL_MODE_DTLS) ? "UDP" : "TCP") + << " server listening on " << server_socket_->GetLocalAddress(); + } + + rtc::SocketAddress GetAddress() const { + return server_socket_->GetLocalAddress(); + } + + std::string GetHostname() const { + // Since we don't have a real certificate anyway, the value here doesn't + // really matter. + return "example.com"; + } + + const std::string& GetReceivedData() const { + return data_; + } + + int Send(const std::string& message) { + if (ssl_stream_adapter_ == NULL + || ssl_stream_adapter_->GetState() != rtc::SS_OPEN) { + // No connection yet. + return -1; + } + + LOG(LS_INFO) << "Server sending '" << message << "'"; + + size_t written; + int error; + + rtc::StreamResult r = ssl_stream_adapter_->Write(message.data(), + message.length(), &written, &error); + if (r == rtc::SR_SUCCESS) { + return written; + } else { + return -1; + } + } + + void OnServerSocketReadEvent(rtc::AsyncSocket* socket) { + if (ssl_stream_adapter_ != NULL) { + // Only a single connection is supported. + return; + } + + rtc::SocketAddress address; + rtc::AsyncSocket* new_socket = socket->Accept(&address); + rtc::SocketStream* stream = new rtc::SocketStream(new_socket); + + ssl_stream_adapter_.reset(rtc::SSLStreamAdapter::Create(stream)); + ssl_stream_adapter_->SetServerRole(); + + // SSLStreamAdapter is normally used for peer-to-peer communication, but + // here we're testing communication between a client and a server + // (e.g. a WebRTC-based application and an RFC 5766 TURN server), where + // clients are not required to provide a certificate during handshake. + // Accordingly, we must disable client authentication here. + ssl_stream_adapter_->set_client_auth_enabled(false); + + ssl_stream_adapter_->SetIdentity(ssl_identity_->GetReference()); + + // Set a bogus peer certificate digest. + unsigned char digest[20]; + size_t digest_len = sizeof(digest); + ssl_stream_adapter_->SetPeerCertificateDigest(rtc::DIGEST_SHA_1, digest, + digest_len); + + ssl_stream_adapter_->StartSSLWithPeer(); + + ssl_stream_adapter_->SignalEvent.connect(this, + &SSLAdapterTestDummyServer::OnSSLStreamAdapterEvent); + } + + void OnSSLStreamAdapterEvent(rtc::StreamInterface* stream, int sig, int err) { + if (sig & rtc::SE_READ) { + char buffer[4096] = ""; + + size_t read; + int error; + + // Read data received from the client and store it in our internal + // buffer. + rtc::StreamResult r = stream->Read(buffer, + sizeof(buffer) - 1, &read, &error); + if (r == rtc::SR_SUCCESS) { + buffer[read] = '\0'; + + LOG(LS_INFO) << "Server received '" << buffer << "'"; + + data_ += buffer; + } + } + } + + private: + const rtc::SSLMode ssl_mode_; + + rtc::scoped_ptr<rtc::AsyncSocket> server_socket_; + rtc::scoped_ptr<rtc::SSLStreamAdapter> ssl_stream_adapter_; + + rtc::scoped_ptr<rtc::SSLIdentity> ssl_identity_; + + std::string data_; +}; + +class SSLAdapterTestBase : public testing::Test, + public sigslot::has_slots<> { + public: + explicit SSLAdapterTestBase(const rtc::SSLMode& ssl_mode) + : ssl_mode_(ssl_mode), + ss_scope_(new rtc::VirtualSocketServer(NULL)), + server_(new SSLAdapterTestDummyServer(ssl_mode_)), + client_(new SSLAdapterTestDummyClient(ssl_mode_)), + handshake_wait_(kTimeout) { + } + + static void SetUpTestCase() { + rtc::InitializeSSL(); + } + + static void TearDownTestCase() { + rtc::CleanupSSL(); + } + + void SetHandshakeWait(int wait) { + handshake_wait_ = wait; + } + + void TestHandshake(bool expect_success) { + int rv; + + // The initial state is CS_CLOSED + ASSERT_EQ(rtc::AsyncSocket::CS_CLOSED, client_->GetState()); + + rv = client_->Connect(server_->GetHostname(), server_->GetAddress()); + ASSERT_EQ(0, rv); + + // Now the state should be CS_CONNECTING + ASSERT_EQ(rtc::AsyncSocket::CS_CONNECTING, client_->GetState()); + + if (expect_success) { + // If expecting success, the client should end up in the CS_CONNECTED + // state after handshake. + EXPECT_EQ_WAIT(rtc::AsyncSocket::CS_CONNECTED, client_->GetState(), + handshake_wait_); + + LOG(LS_INFO) << GetSSLProtocolName(ssl_mode_) << " handshake complete."; + + } else { + // On handshake failure the client should end up in the CS_CLOSED state. + EXPECT_EQ_WAIT(rtc::AsyncSocket::CS_CLOSED, client_->GetState(), + handshake_wait_); + + LOG(LS_INFO) << GetSSLProtocolName(ssl_mode_) << " handshake failed."; + } + } + + void TestTransfer(const std::string& message) { + int rv; + + rv = client_->Send(message); + ASSERT_EQ(static_cast<int>(message.length()), rv); + + // The server should have received the client's message. + EXPECT_EQ_WAIT(message, server_->GetReceivedData(), kTimeout); + + rv = server_->Send(message); + ASSERT_EQ(static_cast<int>(message.length()), rv); + + // The client should have received the server's message. + EXPECT_EQ_WAIT(message, client_->GetReceivedData(), kTimeout); + + LOG(LS_INFO) << "Transfer complete."; + } + + private: + const rtc::SSLMode ssl_mode_; + + const rtc::SocketServerScope ss_scope_; + + rtc::scoped_ptr<SSLAdapterTestDummyServer> server_; + rtc::scoped_ptr<SSLAdapterTestDummyClient> client_; + + int handshake_wait_; +}; + +class SSLAdapterTestTLS : public SSLAdapterTestBase { + public: + SSLAdapterTestTLS() : SSLAdapterTestBase(rtc::SSL_MODE_TLS) {} +}; + + +#if SSL_USE_OPENSSL + +// Basic tests: TLS + +// Test that handshake works +TEST_F(SSLAdapterTestTLS, TestTLSConnect) { + TestHandshake(true); +} + +// Test transfer between client and server +TEST_F(SSLAdapterTestTLS, TestTLSTransfer) { + TestHandshake(true); + TestTransfer("Hello, world!"); +} + +#endif // SSL_USE_OPENSSL + diff --git a/base/sslstreamadapter.h b/base/sslstreamadapter.h index ffe6b2f7..ea966c54 100644 --- a/base/sslstreamadapter.h +++ b/base/sslstreamadapter.h @@ -48,11 +48,15 @@ class SSLStreamAdapter : public StreamAdapterInterface { static SSLStreamAdapter* Create(StreamInterface* stream); explicit SSLStreamAdapter(StreamInterface* stream) - : StreamAdapterInterface(stream), ignore_bad_cert_(false) { } + : StreamAdapterInterface(stream), ignore_bad_cert_(false), + client_auth_enabled_(true) { } void set_ignore_bad_cert(bool ignore) { ignore_bad_cert_ = ignore; } bool ignore_bad_cert() const { return ignore_bad_cert_; } + void set_client_auth_enabled(bool enabled) { client_auth_enabled_ = enabled; } + bool client_auth_enabled() const { return client_auth_enabled_; } + // Specify our SSL identity: key and certificate. Mostly this is // only used in the peer-to-peer mode (unless we actually want to // provide a client certificate to a server). @@ -151,10 +155,16 @@ class SSLStreamAdapter : public StreamAdapterInterface { static bool HaveDtlsSrtp(); static bool HaveExporter(); + private: // If true, the server certificate need not match the configured // server_name, and in fact missing certificate authority and other // verification errors are ignored. bool ignore_bad_cert_; + + // If true (default), the client is required to provide a certificate during + // handshake. If no certificate is given, handshake fails. This applies to + // server mode only. + bool client_auth_enabled_; }; } // namespace rtc diff --git a/base/sslstreamadapter_unittest.cc b/base/sslstreamadapter_unittest.cc index af78bfff..f7dcb71e 100644 --- a/base/sslstreamadapter_unittest.cc +++ b/base/sslstreamadapter_unittest.cc @@ -21,6 +21,7 @@ #include "webrtc/base/sslidentity.h" #include "webrtc/base/sslstreamadapter.h" #include "webrtc/base/stream.h" +#include "webrtc/test/testsupport/gtest_disable.h" static const int kBlockSize = 4096; static const char kAES_CM_HMAC_SHA1_80[] = "AES_CM_128_HMAC_SHA1_80"; @@ -750,7 +751,7 @@ TEST_F(SSLStreamAdapterTestDTLS, }; // Test a handshake with small MTU -TEST_F(SSLStreamAdapterTestDTLS, TestDTLSConnectWithSmallMtu) { +TEST_F(SSLStreamAdapterTestDTLS, DISABLED_ON_MAC(TestDTLSConnectWithSmallMtu)) { MAYBE_SKIP_TEST(HaveDtls); SetMtu(700); SetHandshakeWait(20000); diff --git a/base/stream_unittest.cc b/base/stream_unittest.cc index a6664d71..5c2cb576 100644 --- a/base/stream_unittest.cc +++ b/base/stream_unittest.cc @@ -10,6 +10,7 @@ #include "webrtc/base/gunit.h" #include "webrtc/base/stream.h" +#include "webrtc/test/testsupport/gtest_disable.h" namespace rtc { @@ -434,7 +435,7 @@ TEST(FifoBufferTest, WriteOffsetAndReadOffset) { EXPECT_EQ(SR_BLOCK, buf.ReadOffset(out, 10, 16, NULL)); } -TEST(AsyncWriteTest, TestWrite) { +TEST(AsyncWriteTest, DISABLED_ON_MAC(TestWrite)) { FifoBuffer* buf = new FifoBuffer(100); AsyncWriteStream stream(buf, Thread::Current()); EXPECT_EQ(SS_OPEN, stream.GetState()); diff --git a/base/task_unittest.cc b/base/task_unittest.cc index 8831259c..b74c29fe 100644 --- a/base/task_unittest.cc +++ b/base/task_unittest.cc @@ -27,6 +27,7 @@ #include "webrtc/base/taskrunner.h" #include "webrtc/base/thread.h" #include "webrtc/base/timeutils.h" +#include "webrtc/test/testsupport/gtest_disable.h" namespace rtc { @@ -288,7 +289,7 @@ class TaskTest : public sigslot::has_slots<> { TaskInfo happy_[HAPPY_TASK_COUNT]; }; -TEST(start_task_test, Timeout) { +TEST(start_task_test, DISABLED_ON_MAC(Timeout)) { TaskTest task_test; task_test.Start(); task_test.check_passed(); @@ -335,7 +336,7 @@ class TaskAbortTest : public sigslot::has_slots<> { DISALLOW_EVIL_CONSTRUCTORS(TaskAbortTest); }; -TEST(start_task_test, Abort) { +TEST(start_task_test, DISABLED_ON_MAC(Abort)) { TaskAbortTest abort_test; abort_test.Start(); } @@ -398,7 +399,7 @@ class AbortShouldWakeTest : public sigslot::has_slots<> { DISALLOW_EVIL_CONSTRUCTORS(AbortShouldWakeTest); }; -TEST(start_task_test, AbortShouldWake) { +TEST(start_task_test, DISABLED_ON_MAC(AbortShouldWake)) { AbortShouldWakeTest abort_should_wake_test; abort_should_wake_test.Start(); } @@ -479,7 +480,7 @@ class TimeoutChangeTest : public sigslot::has_slots<> { DISALLOW_EVIL_CONSTRUCTORS(TimeoutChangeTest); }; -TEST(start_task_test, TimeoutChange) { +TEST(start_task_test, DISABLED_ON_MAC(TimeoutChange)) { TimeoutChangeTest timeout_change_test; timeout_change_test.Start(); } @@ -496,7 +497,7 @@ class DeleteTestTaskRunner : public TaskRunner { DISALLOW_EVIL_CONSTRUCTORS(DeleteTestTaskRunner); }; -TEST(unstarted_task_test, DeleteTask) { +TEST(unstarted_task_test, DISABLED_ON_MAC(DeleteTask)) { // This test ensures that we don't // crash if a task is deleted without running it. DeleteTestTaskRunner task_runner; @@ -511,7 +512,7 @@ TEST(unstarted_task_test, DeleteTask) { task_runner.RunTasks(); } -TEST(unstarted_task_test, DoNotDeleteTask1) { +TEST(unstarted_task_test, DISABLED_ON_MAC(DoNotDeleteTask1)) { // This test ensures that we don't // crash if a task runner is deleted without // running a certain task. @@ -525,7 +526,7 @@ TEST(unstarted_task_test, DoNotDeleteTask1) { // Never run the tasks } -TEST(unstarted_task_test, DoNotDeleteTask2) { +TEST(unstarted_task_test, DISABLED_ON_MAC(DoNotDeleteTask2)) { // This test ensures that we don't // crash if a taskrunner is delete with a // task that has never been started. diff --git a/base/testclient_unittest.cc b/base/testclient_unittest.cc index c2826686..21a70555 100644 --- a/base/testclient_unittest.cc +++ b/base/testclient_unittest.cc @@ -14,6 +14,7 @@ #include "webrtc/base/testclient.h" #include "webrtc/base/testechoserver.h" #include "webrtc/base/thread.h" +#include "webrtc/test/testsupport/gtest_disable.h" using namespace rtc; @@ -51,11 +52,11 @@ void TestTcpInternal(const SocketAddress& loopback) { } // Tests whether the TestClient can send UDP to itself. -TEST(TestClientTest, TestUdpIPv4) { +TEST(TestClientTest, DISABLED_ON_MAC(TestUdpIPv4)) { TestUdpInternal(SocketAddress("127.0.0.1", 0)); } -TEST(TestClientTest, TestUdpIPv6) { +TEST(TestClientTest, DISABLED_ON_MAC(TestUdpIPv6)) { if (HasIPv6Enabled()) { TestUdpInternal(SocketAddress("::1", 0)); } else { @@ -64,11 +65,11 @@ TEST(TestClientTest, TestUdpIPv6) { } // Tests whether the TestClient can connect to a server and exchange data. -TEST(TestClientTest, TestTcpIPv4) { +TEST(TestClientTest, DISABLED_ON_MAC(TestTcpIPv4)) { TestTcpInternal(SocketAddress("127.0.0.1", 0)); } -TEST(TestClientTest, TestTcpIPv6) { +TEST(TestClientTest, DISABLED_ON_MAC(TestTcpIPv6)) { if (HasIPv6Enabled()) { TestTcpInternal(SocketAddress("::1", 0)); } else { diff --git a/system_wrappers/interface/thread_annotations.h b/base/thread_annotations.h index 612242d6..612242d6 100644 --- a/system_wrappers/interface/thread_annotations.h +++ b/base/thread_annotations.h diff --git a/base/thread_checker_unittest.cc b/base/thread_checker_unittest.cc index 87e69c03..25b9f503 100644 --- a/base/thread_checker_unittest.cc +++ b/base/thread_checker_unittest.cc @@ -15,6 +15,7 @@ #include "webrtc/base/thread.h" #include "webrtc/base/thread_checker.h" #include "webrtc/base/scoped_ptr.h" +#include "webrtc/test/testsupport/gtest_disable.h" // Duplicated from base/threading/thread_checker.h so that we can be // good citizens there and undef the macro. @@ -104,7 +105,7 @@ class DeleteThreadCheckerClassOnThread : public Thread { } // namespace -TEST(ThreadCheckerTest, CallsAllowedOnSameThread) { +TEST(ThreadCheckerTest, DISABLED_ON_MAC(CallsAllowedOnSameThread)) { scoped_ptr<ThreadCheckerClass> thread_checker_class( new ThreadCheckerClass); @@ -115,7 +116,7 @@ TEST(ThreadCheckerTest, CallsAllowedOnSameThread) { thread_checker_class.reset(); } -TEST(ThreadCheckerTest, DestructorAllowedOnDifferentThread) { +TEST(ThreadCheckerTest, DISABLED_ON_MAC(DestructorAllowedOnDifferentThread)) { scoped_ptr<ThreadCheckerClass> thread_checker_class( new ThreadCheckerClass); @@ -128,7 +129,7 @@ TEST(ThreadCheckerTest, DestructorAllowedOnDifferentThread) { delete_on_thread.Join(); } -TEST(ThreadCheckerTest, DetachFromThread) { +TEST(ThreadCheckerTest, DISABLED_ON_MAC(DetachFromThread)) { scoped_ptr<ThreadCheckerClass> thread_checker_class( new ThreadCheckerClass); diff --git a/base/thread_unittest.cc b/base/thread_unittest.cc index 6a687574..4229df28 100644 --- a/base/thread_unittest.cc +++ b/base/thread_unittest.cc @@ -15,6 +15,7 @@ #include "webrtc/base/physicalsocketserver.h" #include "webrtc/base/socketaddress.h" #include "webrtc/base/thread.h" +#include "webrtc/test/testsupport/gtest_disable.h" #if defined(WEBRTC_WIN) #include <comdef.h> // NOLINT @@ -198,7 +199,7 @@ TEST(ThreadTest, DISABLED_Main) { // Test that setting thread names doesn't cause a malfunction. // There's no easy way to verify the name was set properly at this time. -TEST(ThreadTest, Names) { +TEST(ThreadTest, DISABLED_ON_MAC(Names)) { // Default name Thread *thread; thread = new Thread(); @@ -221,7 +222,7 @@ TEST(ThreadTest, Names) { // Test that setting thread priorities doesn't cause a malfunction. // There's no easy way to verify the priority was set properly at this time. -TEST(ThreadTest, Priorities) { +TEST(ThreadTest, DISABLED_ON_MAC(Priorities)) { Thread *thread; thread = new Thread(); EXPECT_TRUE(thread->SetPriority(PRIORITY_HIGH)); @@ -246,7 +247,7 @@ TEST(ThreadTest, Priorities) { } -TEST(ThreadTest, Wrap) { +TEST(ThreadTest, DISABLED_ON_MAC(Wrap)) { CustomThread* cthread = new CustomThread(); EXPECT_TRUE(cthread->WrapCurrent()); EXPECT_TRUE(cthread->RunningForTest()); @@ -256,7 +257,7 @@ TEST(ThreadTest, Wrap) { delete cthread; } -TEST(ThreadTest, Invoke) { +TEST(ThreadTest, DISABLED_ON_MAC(Invoke)) { // Create and start the thread. Thread thread; thread.Start(); @@ -304,7 +305,7 @@ class AsyncInvokeTest : public testing::Test { Thread* expected_thread_; }; -TEST_F(AsyncInvokeTest, FireAndForget) { +TEST_F(AsyncInvokeTest, DISABLED_FireAndForget) { AsyncInvoker invoker; // Create and start the thread. Thread thread; @@ -315,7 +316,7 @@ TEST_F(AsyncInvokeTest, FireAndForget) { EXPECT_TRUE_WAIT(called, kWaitTimeout); } -TEST_F(AsyncInvokeTest, WithCallback) { +TEST_F(AsyncInvokeTest, DISABLED_WithCallback) { AsyncInvoker invoker; // Create and start the thread. Thread thread; @@ -328,7 +329,7 @@ TEST_F(AsyncInvokeTest, WithCallback) { EXPECT_EQ_WAIT(42, int_value_, kWaitTimeout); } -TEST_F(AsyncInvokeTest, CancelInvoker) { +TEST_F(AsyncInvokeTest, DISABLED_CancelInvoker) { // Create and start the thread. Thread thread; thread.Start(); @@ -344,7 +345,7 @@ TEST_F(AsyncInvokeTest, CancelInvoker) { EXPECT_EQ(0, int_value_); } -TEST_F(AsyncInvokeTest, CancelCallingThread) { +TEST_F(AsyncInvokeTest, DISABLED_CancelCallingThread) { AsyncInvoker invoker; { // Create and start the thread. Thread thread; @@ -361,7 +362,7 @@ TEST_F(AsyncInvokeTest, CancelCallingThread) { EXPECT_EQ(0, int_value_); } -TEST_F(AsyncInvokeTest, KillInvokerBeforeExecute) { +TEST_F(AsyncInvokeTest, DISABLED_KillInvokerBeforeExecute) { Thread thread; thread.Start(); { @@ -378,7 +379,7 @@ TEST_F(AsyncInvokeTest, KillInvokerBeforeExecute) { EXPECT_EQ(0, int_value_); } -TEST_F(AsyncInvokeTest, Flush) { +TEST_F(AsyncInvokeTest, DISABLED_Flush) { AsyncInvoker invoker; bool flag1 = false; bool flag2 = false; @@ -396,7 +397,7 @@ TEST_F(AsyncInvokeTest, Flush) { EXPECT_TRUE(flag2); } -TEST_F(AsyncInvokeTest, FlushWithIds) { +TEST_F(AsyncInvokeTest, DISABLED_FlushWithIds) { AsyncInvoker invoker; bool flag1 = false; bool flag2 = false; diff --git a/base/virtualsocket_unittest.cc b/base/virtualsocket_unittest.cc index 253d2c5b..0a26cc9d 100644 --- a/base/virtualsocket_unittest.cc +++ b/base/virtualsocket_unittest.cc @@ -21,6 +21,7 @@ #include "webrtc/base/thread.h" #include "webrtc/base/timeutils.h" #include "webrtc/base/virtualsocketserver.h" +#include "webrtc/test/testsupport/gtest_disable.h" using namespace rtc; @@ -775,37 +776,37 @@ class VirtualSocketServerTest : public testing::Test { const SocketAddress kIPv6AnyAddress; }; -TEST_F(VirtualSocketServerTest, basic_v4) { +TEST_F(VirtualSocketServerTest, DISABLED_ON_MAC(basic_v4)) { SocketAddress ipv4_test_addr(IPAddress(INADDR_ANY), 5000); BasicTest(ipv4_test_addr); } -TEST_F(VirtualSocketServerTest, basic_v6) { +TEST_F(VirtualSocketServerTest,DISABLED_ON_MAC( basic_v6)) { SocketAddress ipv6_test_addr(IPAddress(in6addr_any), 5000); BasicTest(ipv6_test_addr); } -TEST_F(VirtualSocketServerTest, connect_v4) { +TEST_F(VirtualSocketServerTest, DISABLED_ON_MAC(connect_v4)) { ConnectTest(kIPv4AnyAddress); } -TEST_F(VirtualSocketServerTest, connect_v6) { +TEST_F(VirtualSocketServerTest, DISABLED_ON_MAC(connect_v6)) { ConnectTest(kIPv6AnyAddress); } -TEST_F(VirtualSocketServerTest, connect_to_non_listener_v4) { +TEST_F(VirtualSocketServerTest, DISABLED_ON_MAC(connect_to_non_listener_v4)) { ConnectToNonListenerTest(kIPv4AnyAddress); } -TEST_F(VirtualSocketServerTest, connect_to_non_listener_v6) { +TEST_F(VirtualSocketServerTest, DISABLED_ON_MAC(connect_to_non_listener_v6)) { ConnectToNonListenerTest(kIPv6AnyAddress); } -TEST_F(VirtualSocketServerTest, close_during_connect_v4) { +TEST_F(VirtualSocketServerTest, DISABLED_ON_MAC(close_during_connect_v4)) { CloseDuringConnectTest(kIPv4AnyAddress); } -TEST_F(VirtualSocketServerTest, close_during_connect_v6) { +TEST_F(VirtualSocketServerTest, DISABLED_ON_MAC(close_during_connect_v6)) { CloseDuringConnectTest(kIPv6AnyAddress); } @@ -817,11 +818,11 @@ TEST_F(VirtualSocketServerTest, close_v6) { CloseTest(kIPv6AnyAddress); } -TEST_F(VirtualSocketServerTest, tcp_send_v4) { +TEST_F(VirtualSocketServerTest, DISABLED_ON_MAC(tcp_send_v4)) { TcpSendTest(kIPv4AnyAddress); } -TEST_F(VirtualSocketServerTest, tcp_send_v6) { +TEST_F(VirtualSocketServerTest, DISABLED_ON_MAC(tcp_send_v6)) { TcpSendTest(kIPv6AnyAddress); } @@ -833,17 +834,17 @@ TEST_F(VirtualSocketServerTest, TcpSendsPacketsInOrder_v6) { TcpSendsPacketsInOrderTest(kIPv6AnyAddress); } -TEST_F(VirtualSocketServerTest, bandwidth_v4) { +TEST_F(VirtualSocketServerTest, DISABLED_ON_MAC(bandwidth_v4)) { SocketAddress ipv4_test_addr(IPAddress(INADDR_ANY), 1000); BandwidthTest(ipv4_test_addr); } -TEST_F(VirtualSocketServerTest, bandwidth_v6) { +TEST_F(VirtualSocketServerTest, DISABLED_ON_MAC(bandwidth_v6)) { SocketAddress ipv6_test_addr(IPAddress(in6addr_any), 1000); BandwidthTest(ipv6_test_addr); } -TEST_F(VirtualSocketServerTest, delay_v4) { +TEST_F(VirtualSocketServerTest, DISABLED_ON_MAC(delay_v4)) { SocketAddress ipv4_test_addr(IPAddress(INADDR_ANY), 1000); DelayTest(ipv4_test_addr); } @@ -855,104 +856,104 @@ TEST_F(VirtualSocketServerTest, DISABLED_delay_v6) { } // Works, receiving socket sees 127.0.0.2. -TEST_F(VirtualSocketServerTest, CanConnectFromMappedIPv6ToIPv4Any) { +TEST_F(VirtualSocketServerTest, DISABLED_ON_MAC(CanConnectFromMappedIPv6ToIPv4Any)) { CrossFamilyConnectionTest(SocketAddress("::ffff:127.0.0.2", 0), SocketAddress("0.0.0.0", 5000), true); } // Fails. -TEST_F(VirtualSocketServerTest, CantConnectFromUnMappedIPv6ToIPv4Any) { +TEST_F(VirtualSocketServerTest, DISABLED_ON_MAC(CantConnectFromUnMappedIPv6ToIPv4Any)) { CrossFamilyConnectionTest(SocketAddress("::2", 0), SocketAddress("0.0.0.0", 5000), false); } // Fails. -TEST_F(VirtualSocketServerTest, CantConnectFromUnMappedIPv6ToMappedIPv6) { +TEST_F(VirtualSocketServerTest, DISABLED_ON_MAC(CantConnectFromUnMappedIPv6ToMappedIPv6)) { CrossFamilyConnectionTest(SocketAddress("::2", 0), SocketAddress("::ffff:127.0.0.1", 5000), false); } // Works. receiving socket sees ::ffff:127.0.0.2. -TEST_F(VirtualSocketServerTest, CanConnectFromIPv4ToIPv6Any) { +TEST_F(VirtualSocketServerTest, DISABLED_ON_MAC(CanConnectFromIPv4ToIPv6Any)) { CrossFamilyConnectionTest(SocketAddress("127.0.0.2", 0), SocketAddress("::", 5000), true); } // Fails. -TEST_F(VirtualSocketServerTest, CantConnectFromIPv4ToUnMappedIPv6) { +TEST_F(VirtualSocketServerTest, DISABLED_ON_MAC(CantConnectFromIPv4ToUnMappedIPv6)) { CrossFamilyConnectionTest(SocketAddress("127.0.0.2", 0), SocketAddress("::1", 5000), false); } // Works. Receiving socket sees ::ffff:127.0.0.1. -TEST_F(VirtualSocketServerTest, CanConnectFromIPv4ToMappedIPv6) { +TEST_F(VirtualSocketServerTest, DISABLED_ON_MAC(CanConnectFromIPv4ToMappedIPv6)) { CrossFamilyConnectionTest(SocketAddress("127.0.0.1", 0), SocketAddress("::ffff:127.0.0.2", 5000), true); } // Works, receiving socket sees a result from GetNextIP. -TEST_F(VirtualSocketServerTest, CanConnectFromUnboundIPv6ToIPv4Any) { +TEST_F(VirtualSocketServerTest, DISABLED_ON_MAC(CanConnectFromUnboundIPv6ToIPv4Any)) { CrossFamilyConnectionTest(SocketAddress("::", 0), SocketAddress("0.0.0.0", 5000), true); } // Works, receiving socket sees whatever GetNextIP gave the client. -TEST_F(VirtualSocketServerTest, CanConnectFromUnboundIPv4ToIPv6Any) { +TEST_F(VirtualSocketServerTest, DISABLED_ON_MAC(CanConnectFromUnboundIPv4ToIPv6Any)) { CrossFamilyConnectionTest(SocketAddress("0.0.0.0", 0), SocketAddress("::", 5000), true); } -TEST_F(VirtualSocketServerTest, CanSendDatagramFromUnboundIPv4ToIPv6Any) { +TEST_F(VirtualSocketServerTest, DISABLED_ON_MAC(CanSendDatagramFromUnboundIPv4ToIPv6Any)) { CrossFamilyDatagramTest(SocketAddress("0.0.0.0", 0), SocketAddress("::", 5000), true); } -TEST_F(VirtualSocketServerTest, CanSendDatagramFromMappedIPv6ToIPv4Any) { +TEST_F(VirtualSocketServerTest, DISABLED_ON_MAC(CanSendDatagramFromMappedIPv6ToIPv4Any)) { CrossFamilyDatagramTest(SocketAddress("::ffff:127.0.0.1", 0), SocketAddress("0.0.0.0", 5000), true); } -TEST_F(VirtualSocketServerTest, CantSendDatagramFromUnMappedIPv6ToIPv4Any) { +TEST_F(VirtualSocketServerTest, DISABLED_ON_MAC(CantSendDatagramFromUnMappedIPv6ToIPv4Any)) { CrossFamilyDatagramTest(SocketAddress("::2", 0), SocketAddress("0.0.0.0", 5000), false); } -TEST_F(VirtualSocketServerTest, CantSendDatagramFromUnMappedIPv6ToMappedIPv6) { +TEST_F(VirtualSocketServerTest, DISABLED_ON_MAC(CantSendDatagramFromUnMappedIPv6ToMappedIPv6)) { CrossFamilyDatagramTest(SocketAddress("::2", 0), SocketAddress("::ffff:127.0.0.1", 5000), false); } -TEST_F(VirtualSocketServerTest, CanSendDatagramFromIPv4ToIPv6Any) { +TEST_F(VirtualSocketServerTest, DISABLED_ON_MAC(CanSendDatagramFromIPv4ToIPv6Any)) { CrossFamilyDatagramTest(SocketAddress("127.0.0.2", 0), SocketAddress("::", 5000), true); } -TEST_F(VirtualSocketServerTest, CantSendDatagramFromIPv4ToUnMappedIPv6) { +TEST_F(VirtualSocketServerTest, DISABLED_ON_MAC(CantSendDatagramFromIPv4ToUnMappedIPv6)) { CrossFamilyDatagramTest(SocketAddress("127.0.0.2", 0), SocketAddress("::1", 5000), false); } -TEST_F(VirtualSocketServerTest, CanSendDatagramFromIPv4ToMappedIPv6) { +TEST_F(VirtualSocketServerTest, DISABLED_ON_MAC(CanSendDatagramFromIPv4ToMappedIPv6)) { CrossFamilyDatagramTest(SocketAddress("127.0.0.1", 0), SocketAddress("::ffff:127.0.0.2", 5000), true); } -TEST_F(VirtualSocketServerTest, CanSendDatagramFromUnboundIPv6ToIPv4Any) { +TEST_F(VirtualSocketServerTest, DISABLED_ON_MAC(CanSendDatagramFromUnboundIPv6ToIPv4Any)) { CrossFamilyDatagramTest(SocketAddress("::", 0), SocketAddress("0.0.0.0", 5000), true); diff --git a/build/common.gypi b/build/common.gypi index 92404fde..fdf1abe1 100644 --- a/build/common.gypi +++ b/build/common.gypi @@ -148,11 +148,6 @@ 'include_tests%': 1, 'restrict_webrtc_logging%': 0, }], - ['OS=="mac" or OS=="ios"', { - # Set the minimum SDK level to 10.7 since we need AVFoundation. - # See ../build/common.gypi for more info about this variable. - 'mac_sdk_min%': '10.7', - }], ['OS=="ios"', { 'build_libjpeg%': 0, 'enable_protobuf%': 0, @@ -92,8 +92,7 @@ class Call { virtual VideoSendStream* CreateVideoSendStream( const VideoSendStream::Config& config, - const std::vector<VideoStream>& video_streams, - const void* encoder_settings) = 0; + const VideoEncoderConfig& encoder_config) = 0; virtual void DestroyVideoSendStream(VideoSendStream* send_stream) = 0; @@ -108,6 +108,20 @@ struct VideoStream { std::vector<int> temporal_layers; }; +struct VideoEncoderConfig { + enum ContentType { + kRealtimeVideo, + kScreenshare, + }; + + VideoEncoderConfig() + : content_type(kRealtimeVideo), encoder_specific_settings(NULL) {} + + std::vector<VideoStream> streams; + ContentType content_type; + void* encoder_specific_settings; +}; + } // namespace webrtc #endif // WEBRTC_CONFIG_H_ diff --git a/modules/audio_coding/codecs/isac/fix/interface/isacfix.h b/modules/audio_coding/codecs/isac/fix/interface/isacfix.h index 7cd29544..942727ab 100644 --- a/modules/audio_coding/codecs/isac/fix/interface/isacfix.h +++ b/modules/audio_coding/codecs/isac/fix/interface/isacfix.h @@ -130,7 +130,7 @@ extern "C" { int16_t WebRtcIsacfix_Encode(ISACFIX_MainStruct *ISAC_main_inst, const int16_t *speechIn, - int16_t *encoded); + uint8_t* encoded); diff --git a/modules/audio_coding/codecs/isac/fix/source/isacfix.c b/modules/audio_coding/codecs/isac/fix/source/isacfix.c index e855daea..118fd2fe 100644 --- a/modules/audio_coding/codecs/isac/fix/source/isacfix.c +++ b/modules/audio_coding/codecs/isac/fix/source/isacfix.c @@ -17,6 +17,7 @@ #include "webrtc/modules/audio_coding/codecs/isac/fix/interface/isacfix.h" +#include <assert.h> #include <stdlib.h> #include "webrtc/modules/audio_coding/codecs/isac/fix/source/bandwidth_estimator.h" @@ -355,10 +356,10 @@ int16_t WebRtcIsacfix_EncoderInit(ISACFIX_MainStruct *ISAC_main_inst, int16_t WebRtcIsacfix_Encode(ISACFIX_MainStruct *ISAC_main_inst, const int16_t *speechIn, - int16_t *encoded) + uint8_t* encoded) { ISACFIX_SubStruct *ISAC_inst; - int16_t stream_len; + int16_t stream_len, stream_len_even; #ifndef WEBRTC_ARCH_BIG_ENDIAN int k; #endif @@ -382,16 +383,24 @@ int16_t WebRtcIsacfix_Encode(ISACFIX_MainStruct *ISAC_main_inst, return -1; } + /* One would think that only even stream lengths would make sense here. We do + in fact observe odd lengths, however, and in those cases we copy an extra + byte. */ + stream_len_even = stream_len % 2 == 0 ? stream_len : stream_len + 1; - /* convert from bytes to int16_t */ #ifndef WEBRTC_ARCH_BIG_ENDIAN - for (k=0;k<(stream_len+1)>>1;k++) { - encoded[k] = (int16_t)( ( (uint16_t)(ISAC_inst->ISACenc_obj.bitstr_obj).stream[k] >> 8 ) - | (((ISAC_inst->ISACenc_obj.bitstr_obj).stream[k] & 0x00FF) << 8)); + /* The encoded data vector is supposesd to be big-endian, but our internal + representation is little-endian. So byteswap. */ + for (k = 0; k < stream_len_even / 2; ++k) { + uint16_t s = ISAC_inst->ISACenc_obj.bitstr_obj.stream[k]; + /* In big-endian, we have... */ + encoded[2 * k] = s >> 8; /* ...most significant byte at low address... */ + encoded[2 * k + 1] = s; /* ...least significant byte at high address. */ } - #else - WEBRTC_SPL_MEMCPY_W16(encoded, (ISAC_inst->ISACenc_obj.bitstr_obj).stream, (stream_len + 1)>>1); + /* The encoded data vector and our internal representation are both + big-endian. */ + memcpy(encoded, ISAC_inst->ISACenc_obj.bitstr_obj.stream, stream_len_even); #endif diff --git a/modules/audio_coding/codecs/isac/fix/test/isac_speed_test.cc b/modules/audio_coding/codecs/isac/fix/test/isac_speed_test.cc index 35827511..207ee8c3 100644 --- a/modules/audio_coding/codecs/isac/fix/test/isac_speed_test.cc +++ b/modules/audio_coding/codecs/isac/fix/test/isac_speed_test.cc @@ -71,7 +71,7 @@ float IsacSpeedTest::EncodeABlock(int16_t* in_data, uint8_t* bit_stream, size_t pointer = 0; for (int idx = 0; idx < subblocks; idx++, pointer += subblock_length) { value = WebRtcIsacfix_Encode(ISACFIX_main_inst_, &in_data[pointer], - reinterpret_cast<int16_t*>(bit_stream)); + bit_stream); } clocks = clock() - clocks; EXPECT_GT(value, 0); diff --git a/modules/audio_coding/codecs/isac/fix/test/kenny.cc b/modules/audio_coding/codecs/isac/fix/test/kenny.cc index c3c6f135..91c4d76a 100644 --- a/modules/audio_coding/codecs/isac/fix/test/kenny.cc +++ b/modules/audio_coding/codecs/isac/fix/test/kenny.cc @@ -565,7 +565,7 @@ int main(int argc, char* argv[]) /* Encode */ stream_len = WebRtcIsacfix_Encode(ISAC_main_inst, shortdata, - (int16_t*)streamdata); + (uint8_t*)streamdata); /* If packet is ready, and CE testing, call the different API functions from the internal API. */ diff --git a/modules/audio_coding/codecs/isac/fix/test/test_iSACfixfloat.c b/modules/audio_coding/codecs/isac/fix/test/test_iSACfixfloat.c index d5682b21..965f2bc1 100644 --- a/modules/audio_coding/codecs/isac/fix/test/test_iSACfixfloat.c +++ b/modules/audio_coding/codecs/isac/fix/test/test_iSACfixfloat.c @@ -439,7 +439,9 @@ int main(int argc, char* argv[]) /* iSAC encoding */ if (mode==0 || mode ==1) { - stream_len = WebRtcIsac_Encode(ISAC_main_inst, shortdata, streamdata); + stream_len = WebRtcIsac_Encode(ISAC_main_inst, + shortdata, + (uint8_t*)streamdata); if (stream_len < 0) { /* exit if returned with error */ errtype=WebRtcIsac_GetErrorCode(ISAC_main_inst); @@ -449,7 +451,10 @@ int main(int argc, char* argv[]) } else if (mode==2 || mode==3) { /* iSAC encoding */ if (nbTest != 1) - stream_len = WebRtcIsacfix_Encode(ISACFIX_main_inst, shortdata, streamdata); + stream_len = WebRtcIsacfix_Encode( + ISACFIX_main_inst, + shortdata, + (uint8_t*)streamdata); else stream_len = WebRtcIsacfix_EncodeNb(ISACFIX_main_inst, shortdata, streamdata); diff --git a/modules/audio_coding/codecs/isac/main/interface/isac.h b/modules/audio_coding/codecs/isac/main/interface/isac.h index 76a61e6d..40670580 100644 --- a/modules/audio_coding/codecs/isac/main/interface/isac.h +++ b/modules/audio_coding/codecs/isac/main/interface/isac.h @@ -147,7 +147,7 @@ extern "C" { int16_t WebRtcIsac_Encode( ISACStruct* ISAC_main_inst, const int16_t* speechIn, - int16_t* encoded); + uint8_t* encoded); /****************************************************************************** diff --git a/modules/audio_coding/codecs/isac/main/source/isac.c b/modules/audio_coding/codecs/isac/main/source/isac.c index d47eb80b..13170a08 100644 --- a/modules/audio_coding/codecs/isac/main/source/isac.c +++ b/modules/audio_coding/codecs/isac/main/source/isac.c @@ -496,7 +496,7 @@ int16_t WebRtcIsac_EncoderInit(ISACStruct* ISAC_main_inst, */ int16_t WebRtcIsac_Encode(ISACStruct* ISAC_main_inst, const int16_t* speechIn, - int16_t* encoded) { + uint8_t* encoded) { float inFrame[FRAMESAMPLES_10ms]; int16_t speechInLB[FRAMESAMPLES_10ms]; int16_t speechInUB[FRAMESAMPLES_10ms]; @@ -504,7 +504,6 @@ int16_t WebRtcIsac_Encode(ISACStruct* ISAC_main_inst, int16_t streamLenUB = 0; int16_t streamLen = 0; int16_t k = 0; - uint8_t* ptrEncodedUW8 = (uint8_t*)encoded; int garbageLen = 0; int32_t bottleneck = 0; int16_t bottleneckIdx = 0; @@ -643,23 +642,22 @@ int16_t WebRtcIsac_Encode(ISACStruct* ISAC_main_inst, streamLenUB = 0; } - memcpy(ptrEncodedUW8, instLB->ISACencLB_obj.bitstr_obj.stream, streamLenLB); + memcpy(encoded, instLB->ISACencLB_obj.bitstr_obj.stream, streamLenLB); streamLen = streamLenLB; if (streamLenUB > 0) { - ptrEncodedUW8[streamLenLB] = (uint8_t)(streamLenUB + 1 + - LEN_CHECK_SUM_WORD8); - memcpy(&ptrEncodedUW8[streamLenLB + 1], - instUB->ISACencUB_obj.bitstr_obj.stream, streamLenUB); - streamLen += ptrEncodedUW8[streamLenLB]; + encoded[streamLenLB] = streamLenUB + 1 + LEN_CHECK_SUM_WORD8; + memcpy(&encoded[streamLenLB + 1], + instUB->ISACencUB_obj.bitstr_obj.stream, + streamLenUB); + streamLen += encoded[streamLenLB]; } else { - ptrEncodedUW8[streamLenLB] = 0; + encoded[streamLenLB] = 0; } } else { if (streamLenLB == 0) { return 0; } - memcpy(ptrEncodedUW8, instLB->ISACencLB_obj.bitstr_obj.stream, - streamLenLB); + memcpy(encoded, instLB->ISACencLB_obj.bitstr_obj.stream, streamLenLB); streamLenUB = 0; streamLen = streamLenLB; } @@ -697,11 +695,11 @@ int16_t WebRtcIsac_Encode(ISACStruct* ISAC_main_inst, * 255 is the max garbage length we can signal using 8 bits. */ if ((instISAC->bandwidthKHz == isac8kHz) || (streamLenUB == 0)) { - ptrGarbage = &ptrEncodedUW8[streamLenLB]; + ptrGarbage = &encoded[streamLenLB]; limit = streamLen + 255; } else { - ptrGarbage = &ptrEncodedUW8[streamLenLB + 1 + streamLenUB]; - limit = streamLen + (255 - ptrEncodedUW8[streamLenLB]); + ptrGarbage = &encoded[streamLenLB + 1 + streamLenUB]; + limit = streamLen + (255 - encoded[streamLenLB]); } minBytes = (minBytes > limit) ? limit : minBytes; @@ -718,13 +716,12 @@ int16_t WebRtcIsac_Encode(ISACStruct* ISAC_main_inst, * That is the only way to preserve backward compatibility. */ if ((instISAC->bandwidthKHz == isac8kHz) || (streamLenUB == 0)) { - ptrEncodedUW8[streamLenLB] = (uint8_t)garbageLen; + encoded[streamLenLB] = garbageLen; } else { - ptrEncodedUW8[streamLenLB] += (uint8_t)garbageLen; + encoded[streamLenLB] += garbageLen; /* Write the length of the garbage at the end of the upper-band * bit-stream, if exists. This helps for sanity check. */ - ptrEncodedUW8[streamLenLB + 1 + streamLenUB] = - (uint8_t)garbageLen; + encoded[streamLenLB + 1 + streamLenUB] = garbageLen; } streamLen += garbageLen; @@ -741,16 +738,14 @@ int16_t WebRtcIsac_Encode(ISACStruct* ISAC_main_inst, if ((instISAC->bandwidthKHz != isac8kHz) && (streamLenUB > 0)) { uint32_t crc; - WebRtcIsac_GetCrc((int16_t*)(&(ptrEncodedUW8[streamLenLB + 1])), + WebRtcIsac_GetCrc((int16_t*)(&(encoded[streamLenLB + 1])), streamLenUB + garbageLen, &crc); #ifndef WEBRTC_ARCH_BIG_ENDIAN for (k = 0; k < LEN_CHECK_SUM_WORD8; k++) { - ptrEncodedUW8[streamLen - LEN_CHECK_SUM_WORD8 + k] = - (uint8_t)((crc >> (24 - k * 8)) & 0xFF); + encoded[streamLen - LEN_CHECK_SUM_WORD8 + k] = crc >> (24 - k * 8); } #else - memcpy(&ptrEncodedUW8[streamLenLB + streamLenUB + 1], &crc, - LEN_CHECK_SUM_WORD8); + memcpy(&encoded[streamLenLB + streamLenUB + 1], &crc, LEN_CHECK_SUM_WORD8); #endif } return streamLen; diff --git a/modules/audio_coding/codecs/isac/main/source/isac_unittest.cc b/modules/audio_coding/codecs/isac/main/source/isac_unittest.cc index 3c55bd3d..1d653731 100644 --- a/modules/audio_coding/codecs/isac/main/source/isac_unittest.cc +++ b/modules/audio_coding/codecs/isac/main/source/isac_unittest.cc @@ -31,7 +31,7 @@ class IsacTest : public ::testing::Test { int16_t speech_data_[kIsacNumberOfSamples]; int16_t output_data_[kIsacNumberOfSamples]; - int16_t bitstream_[kMaxBytes / 2]; + uint8_t bitstream_[kMaxBytes]; uint8_t bitstream_small_[7]; // Simulate sync packets. }; diff --git a/modules/audio_coding/codecs/isac/main/test/ReleaseTest-API/ReleaseTest-API.cc b/modules/audio_coding/codecs/isac/main/test/ReleaseTest-API/ReleaseTest-API.cc index c5f9561b..8af4e6f9 100644 --- a/modules/audio_coding/codecs/isac/main/test/ReleaseTest-API/ReleaseTest-API.cc +++ b/modules/audio_coding/codecs/isac/main/test/ReleaseTest-API/ReleaseTest-API.cc @@ -662,8 +662,8 @@ int main(int argc, char* argv[]) if(!(testNum == 3 && framecnt == 0)) { stream_len = WebRtcIsac_Encode(ISAC_main_inst, - shortdata, - (int16_t*)streamdata); + shortdata, + (uint8_t*)streamdata); if((payloadSize != 0) && (stream_len > payloadSize)) { if(testNum == 0) diff --git a/modules/audio_coding/codecs/isac/main/test/SwitchingSampRate/SwitchingSampRate.cc b/modules/audio_coding/codecs/isac/main/test/SwitchingSampRate/SwitchingSampRate.cc index fd70eca7..72d3fe86 100644 --- a/modules/audio_coding/codecs/isac/main/test/SwitchingSampRate/SwitchingSampRate.cc +++ b/modules/audio_coding/codecs/isac/main/test/SwitchingSampRate/SwitchingSampRate.cc @@ -283,7 +283,8 @@ int main(int argc, char* argv[]) streamLen = WebRtcIsac_Encode(codecInstance[senderIdx], - audioBuff10ms, (short*)bitStream); + audioBuff10ms, + (uint8_t*)bitStream); int16_t ggg; if (streamLen > 0) { if(( WebRtcIsac_ReadFrameLen(codecInstance[receiverIdx], diff --git a/modules/audio_coding/codecs/isac/main/test/simpleKenny.c b/modules/audio_coding/codecs/isac/main/test/simpleKenny.c index 980465d6..2df5a84e 100644 --- a/modules/audio_coding/codecs/isac/main/test/simpleKenny.c +++ b/modules/audio_coding/codecs/isac/main/test/simpleKenny.c @@ -373,8 +373,10 @@ valid values are 8 and 16.\n", sampFreqKHz); cur_framesmpls += samplesIn10Ms; //-------- iSAC encoding --------- - stream_len = WebRtcIsac_Encode(ISAC_main_inst, shortdata, - (int16_t*)payload); + stream_len = WebRtcIsac_Encode( + ISAC_main_inst, + shortdata, + (uint8_t*)payload); if(stream_len < 0) { diff --git a/modules/audio_coding/main/acm2/acm_g722.h b/modules/audio_coding/main/acm2/acm_g722.h index 6717d2c4..dd3b7669 100644 --- a/modules/audio_coding/main/acm2/acm_g722.h +++ b/modules/audio_coding/main/acm2/acm_g722.h @@ -11,8 +11,8 @@ #ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_G722_H_ #define WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_G722_H_ +#include "webrtc/base/thread_annotations.h" #include "webrtc/modules/audio_coding/main/acm2/acm_generic_codec.h" -#include "webrtc/system_wrappers/interface/thread_annotations.h" typedef struct WebRtcG722EncInst G722EncInst; typedef struct WebRtcG722DecInst G722DecInst; diff --git a/modules/audio_coding/main/acm2/acm_generic_codec.h b/modules/audio_coding/main/acm2/acm_generic_codec.h index adca299c..ad6f412a 100644 --- a/modules/audio_coding/main/acm2/acm_generic_codec.h +++ b/modules/audio_coding/main/acm2/acm_generic_codec.h @@ -11,12 +11,12 @@ #ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_GENERIC_CODEC_H_ #define WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_GENERIC_CODEC_H_ +#include "webrtc/base/thread_annotations.h" #include "webrtc/modules/audio_coding/main/interface/audio_coding_module_typedefs.h" #include "webrtc/modules/audio_coding/main/acm2/acm_common_defs.h" #include "webrtc/modules/audio_coding/neteq/interface/neteq.h" #include "webrtc/modules/audio_coding/neteq/interface/audio_decoder.h" #include "webrtc/system_wrappers/interface/rw_lock_wrapper.h" -#include "webrtc/system_wrappers/interface/thread_annotations.h" #include "webrtc/system_wrappers/interface/trace.h" #define MAX_FRAME_SIZE_10MSEC 6 diff --git a/modules/audio_coding/main/acm2/acm_isac.cc b/modules/audio_coding/main/acm2/acm_isac.cc index 850204f3..f3682f1e 100644 --- a/modules/audio_coding/main/acm2/acm_isac.cc +++ b/modules/audio_coding/main/acm2/acm_isac.cc @@ -347,8 +347,9 @@ int16_t ACMISAC::InternalEncode(uint8_t* bitstream, return -1; } *bitstream_len_byte = ACM_ISAC_ENCODE( - codec_inst_ptr_->inst, &in_audio_[in_audio_ix_read_], - reinterpret_cast<int16_t*>(bitstream)); + codec_inst_ptr_->inst, + &in_audio_[in_audio_ix_read_], + bitstream); // increment the read index this tell the caller that how far // we have gone forward in reading the audio buffer in_audio_ix_read_ += samples_in_10ms_audio_; diff --git a/modules/audio_coding/main/acm2/acm_isac.h b/modules/audio_coding/main/acm2/acm_isac.h index 23548229..486313a0 100644 --- a/modules/audio_coding/main/acm2/acm_isac.h +++ b/modules/audio_coding/main/acm2/acm_isac.h @@ -11,10 +11,10 @@ #ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_ISAC_H_ #define WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_ISAC_H_ +#include "webrtc/base/thread_annotations.h" #include "webrtc/modules/audio_coding/main/acm2/acm_generic_codec.h" #include "webrtc/modules/audio_coding/neteq/interface/audio_decoder.h" #include "webrtc/system_wrappers/interface/scoped_ptr.h" -#include "webrtc/system_wrappers/interface/thread_annotations.h" namespace webrtc { diff --git a/modules/audio_coding/main/acm2/acm_receive_test.cc b/modules/audio_coding/main/acm2/acm_receive_test.cc index 79c9adf0..7e41328e 100644 --- a/modules/audio_coding/main/acm2/acm_receive_test.cc +++ b/modules/audio_coding/main/acm2/acm_receive_test.cc @@ -22,124 +22,67 @@ namespace webrtc { namespace test { -namespace { -// Returns true if the codec should be registered, otherwise false. Changes -// the number of channels for the Opus codec to always be 1. -bool ModifyAndUseThisCodec(CodecInst* codec_param) { - if (STR_CASE_CMP(codec_param->plname, "CN") == 0 && - codec_param->plfreq == 48000) - return false; // Skip 48 kHz comfort noise. - - if (STR_CASE_CMP(codec_param->plname, "telephone-event") == 0) - return false; // Skip DTFM. - - return true; -} - -// Remaps payload types from ACM's default to those used in the resource file -// neteq_universal_new.rtp. Returns true if the codec should be registered, -// otherwise false. The payload types are set as follows (all are mono codecs): -// PCMu = 0; -// PCMa = 8; -// Comfort noise 8 kHz = 13 -// Comfort noise 16 kHz = 98 -// Comfort noise 32 kHz = 99 -// iLBC = 102 -// iSAC wideband = 103 -// iSAC super-wideband = 104 -// iSAC fullband = 124 -// AVT/DTMF = 106 -// RED = 117 -// PCM16b 8 kHz = 93 -// PCM16b 16 kHz = 94 -// PCM16b 32 kHz = 95 -// G.722 = 94 -bool RemapPltypeAndUseThisCodec(const char* plname, - int plfreq, - int channels, - int* pltype) { - if (channels != 1) - return false; // Don't use non-mono codecs. - - // Re-map pltypes to those used in the NetEq test files. - if (STR_CASE_CMP(plname, "PCMU") == 0 && plfreq == 8000) { - *pltype = 0; - } else if (STR_CASE_CMP(plname, "PCMA") == 0 && plfreq == 8000) { - *pltype = 8; - } else if (STR_CASE_CMP(plname, "CN") == 0 && plfreq == 8000) { - *pltype = 13; - } else if (STR_CASE_CMP(plname, "CN") == 0 && plfreq == 16000) { - *pltype = 98; - } else if (STR_CASE_CMP(plname, "CN") == 0 && plfreq == 32000) { - *pltype = 99; - } else if (STR_CASE_CMP(plname, "ILBC") == 0) { - *pltype = 102; - } else if (STR_CASE_CMP(plname, "ISAC") == 0 && plfreq == 16000) { - *pltype = 103; - } else if (STR_CASE_CMP(plname, "ISAC") == 0 && plfreq == 32000) { - *pltype = 104; - } else if (STR_CASE_CMP(plname, "ISAC") == 0 && plfreq == 48000) { - *pltype = 124; - } else if (STR_CASE_CMP(plname, "telephone-event") == 0) { - *pltype = 106; - } else if (STR_CASE_CMP(plname, "red") == 0) { - *pltype = 117; - } else if (STR_CASE_CMP(plname, "L16") == 0 && plfreq == 8000) { - *pltype = 93; - } else if (STR_CASE_CMP(plname, "L16") == 0 && plfreq == 16000) { - *pltype = 94; - } else if (STR_CASE_CMP(plname, "L16") == 0 && plfreq == 32000) { - *pltype = 95; - } else if (STR_CASE_CMP(plname, "G722") == 0) { - *pltype = 9; - } else { - // Don't use any other codecs. - return false; - } - return true; -} -} // namespace - AcmReceiveTest::AcmReceiveTest(PacketSource* packet_source, AudioSink* audio_sink, int output_freq_hz, NumOutputChannels exptected_output_channels) : clock_(0), - acm_(webrtc::AudioCodingModule::Create(0, &clock_)), packet_source_(packet_source), audio_sink_(audio_sink), output_freq_hz_(output_freq_hz), exptected_output_channels_(exptected_output_channels) { + webrtc::AudioCoding::Config config; + config.clock = &clock_; + config.playout_frequency_hz = output_freq_hz_; + acm_.reset(webrtc::AudioCoding::Create(config)); } void AcmReceiveTest::RegisterDefaultCodecs() { - CodecInst my_codec_param; - for (int n = 0; n < acm_->NumberOfCodecs(); n++) { - ASSERT_EQ(0, acm_->Codec(n, &my_codec_param)) << "Failed to get codec."; - if (ModifyAndUseThisCodec(&my_codec_param)) { - ASSERT_EQ(0, acm_->RegisterReceiveCodec(my_codec_param)) - << "Couldn't register receive codec.\n"; - } - } + ASSERT_TRUE(acm_->RegisterReceiveCodec(acm2::ACMCodecDB::kOpus, 120)); + ASSERT_TRUE(acm_->RegisterReceiveCodec(acm2::ACMCodecDB::kISAC, 103)); +#ifndef WEBRTC_ANDROID + ASSERT_TRUE(acm_->RegisterReceiveCodec(acm2::ACMCodecDB::kISACSWB, 104)); + ASSERT_TRUE(acm_->RegisterReceiveCodec(acm2::ACMCodecDB::kISACFB, 105)); +#endif + ASSERT_TRUE(acm_->RegisterReceiveCodec(acm2::ACMCodecDB::kPCM16B, 107)); + ASSERT_TRUE(acm_->RegisterReceiveCodec(acm2::ACMCodecDB::kPCM16Bwb, 108)); + ASSERT_TRUE( + acm_->RegisterReceiveCodec(acm2::ACMCodecDB::kPCM16Bswb32kHz, 109)); + ASSERT_TRUE(acm_->RegisterReceiveCodec(acm2::ACMCodecDB::kPCM16B_2ch, 111)); + ASSERT_TRUE(acm_->RegisterReceiveCodec(acm2::ACMCodecDB::kPCM16Bwb_2ch, 112)); + ASSERT_TRUE( + acm_->RegisterReceiveCodec(acm2::ACMCodecDB::kPCM16Bswb32kHz_2ch, 113)); + ASSERT_TRUE(acm_->RegisterReceiveCodec(acm2::ACMCodecDB::kPCMU, 0)); + ASSERT_TRUE(acm_->RegisterReceiveCodec(acm2::ACMCodecDB::kPCMA, 8)); + ASSERT_TRUE(acm_->RegisterReceiveCodec(acm2::ACMCodecDB::kPCMU_2ch, 110)); + ASSERT_TRUE(acm_->RegisterReceiveCodec(acm2::ACMCodecDB::kPCMA_2ch, 118)); + ASSERT_TRUE(acm_->RegisterReceiveCodec(acm2::ACMCodecDB::kILBC, 102)); + ASSERT_TRUE(acm_->RegisterReceiveCodec(acm2::ACMCodecDB::kG722, 9)); + ASSERT_TRUE(acm_->RegisterReceiveCodec(acm2::ACMCodecDB::kG722_2ch, 119)); + ASSERT_TRUE(acm_->RegisterReceiveCodec(acm2::ACMCodecDB::kCNNB, 13)); + ASSERT_TRUE(acm_->RegisterReceiveCodec(acm2::ACMCodecDB::kCNWB, 98)); + ASSERT_TRUE(acm_->RegisterReceiveCodec(acm2::ACMCodecDB::kCNSWB, 99)); + ASSERT_TRUE(acm_->RegisterReceiveCodec(acm2::ACMCodecDB::kRED, 127)); } void AcmReceiveTest::RegisterNetEqTestCodecs() { - CodecInst my_codec_param; - for (int n = 0; n < acm_->NumberOfCodecs(); n++) { - ASSERT_EQ(0, acm_->Codec(n, &my_codec_param)) << "Failed to get codec."; - if (!ModifyAndUseThisCodec(&my_codec_param)) { - // Skip this codec. - continue; - } - - if (RemapPltypeAndUseThisCodec(my_codec_param.plname, - my_codec_param.plfreq, - my_codec_param.channels, - &my_codec_param.pltype)) { - ASSERT_EQ(0, acm_->RegisterReceiveCodec(my_codec_param)) - << "Couldn't register receive codec.\n"; - } - } + ASSERT_TRUE(acm_->RegisterReceiveCodec(acm2::ACMCodecDB::kISAC, 103)); +#ifndef WEBRTC_ANDROID + ASSERT_TRUE(acm_->RegisterReceiveCodec(acm2::ACMCodecDB::kISACSWB, 104)); + ASSERT_TRUE(acm_->RegisterReceiveCodec(acm2::ACMCodecDB::kISACFB, 124)); +#endif + ASSERT_TRUE(acm_->RegisterReceiveCodec(acm2::ACMCodecDB::kPCM16B, 93)); + ASSERT_TRUE(acm_->RegisterReceiveCodec(acm2::ACMCodecDB::kPCM16Bwb, 94)); + ASSERT_TRUE( + acm_->RegisterReceiveCodec(acm2::ACMCodecDB::kPCM16Bswb32kHz, 95)); + ASSERT_TRUE(acm_->RegisterReceiveCodec(acm2::ACMCodecDB::kPCMU, 0)); + ASSERT_TRUE(acm_->RegisterReceiveCodec(acm2::ACMCodecDB::kPCMA, 8)); + ASSERT_TRUE(acm_->RegisterReceiveCodec(acm2::ACMCodecDB::kILBC, 102)); + ASSERT_TRUE(acm_->RegisterReceiveCodec(acm2::ACMCodecDB::kG722, 9)); + ASSERT_TRUE(acm_->RegisterReceiveCodec(acm2::ACMCodecDB::kCNNB, 13)); + ASSERT_TRUE(acm_->RegisterReceiveCodec(acm2::ACMCodecDB::kCNWB, 98)); + ASSERT_TRUE(acm_->RegisterReceiveCodec(acm2::ACMCodecDB::kCNSWB, 99)); + ASSERT_TRUE(acm_->RegisterReceiveCodec(acm2::ACMCodecDB::kRED, 117)); } void AcmReceiveTest::Run() { @@ -148,7 +91,7 @@ void AcmReceiveTest::Run() { // Pull audio until time to insert packet. while (clock_.TimeInMilliseconds() < packet->time_ms()) { AudioFrame output_frame; - EXPECT_EQ(0, acm_->PlayoutData10Ms(output_freq_hz_, &output_frame)); + EXPECT_TRUE(acm_->Get10MsAudio(&output_frame)); EXPECT_EQ(output_freq_hz_, output_frame.sample_rate_hz_); const int samples_per_block = output_freq_hz_ * 10 / 1000; EXPECT_EQ(samples_per_block, output_frame.samples_per_channel_); @@ -170,11 +113,10 @@ void AcmReceiveTest::Run() { header.header = packet->header(); header.frameType = kAudioFrameSpeech; memset(&header.type.Audio, 0, sizeof(RTPAudioHeader)); - EXPECT_EQ(0, - acm_->IncomingPacket( - packet->payload(), - static_cast<int32_t>(packet->payload_length_bytes()), - header)) + EXPECT_TRUE( + acm_->InsertPacket(packet->payload(), + static_cast<int32_t>(packet->payload_length_bytes()), + header)) << "Failure when inserting packet:" << std::endl << " PT = " << static_cast<int>(header.header.payloadType) << std::endl << " TS = " << header.header.timestamp << std::endl diff --git a/modules/audio_coding/main/acm2/acm_receive_test.h b/modules/audio_coding/main/acm2/acm_receive_test.h index c454020d..19fe4c53 100644 --- a/modules/audio_coding/main/acm2/acm_receive_test.h +++ b/modules/audio_coding/main/acm2/acm_receive_test.h @@ -16,7 +16,7 @@ #include "webrtc/system_wrappers/interface/scoped_ptr.h" namespace webrtc { -class AudioCodingModule; +class AudioCoding; struct CodecInst; namespace test { @@ -50,7 +50,7 @@ class AcmReceiveTest { private: SimulatedClock clock_; - scoped_ptr<AudioCodingModule> acm_; + scoped_ptr<AudioCoding> acm_; PacketSource* packet_source_; AudioSink* audio_sink_; const int output_freq_hz_; diff --git a/modules/audio_coding/main/acm2/acm_receive_test_oldapi.cc b/modules/audio_coding/main/acm2/acm_receive_test_oldapi.cc new file mode 100644 index 00000000..b0c9af13 --- /dev/null +++ b/modules/audio_coding/main/acm2/acm_receive_test_oldapi.cc @@ -0,0 +1,187 @@ +/* + * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#include "webrtc/modules/audio_coding/main/acm2/acm_receive_test_oldapi.h" + +#include <assert.h> +#include <stdio.h> + +#include "testing/gtest/include/gtest/gtest.h" +#include "webrtc/modules/audio_coding/main/interface/audio_coding_module.h" +#include "webrtc/modules/audio_coding/neteq/tools/audio_sink.h" +#include "webrtc/modules/audio_coding/neteq/tools/packet.h" +#include "webrtc/modules/audio_coding/neteq/tools/packet_source.h" + +namespace webrtc { +namespace test { + +namespace { +// Returns true if the codec should be registered, otherwise false. Changes +// the number of channels for the Opus codec to always be 1. +bool ModifyAndUseThisCodec(CodecInst* codec_param) { + if (STR_CASE_CMP(codec_param->plname, "CN") == 0 && + codec_param->plfreq == 48000) + return false; // Skip 48 kHz comfort noise. + + if (STR_CASE_CMP(codec_param->plname, "telephone-event") == 0) + return false; // Skip DTFM. + + return true; +} + +// Remaps payload types from ACM's default to those used in the resource file +// neteq_universal_new.rtp. Returns true if the codec should be registered, +// otherwise false. The payload types are set as follows (all are mono codecs): +// PCMu = 0; +// PCMa = 8; +// Comfort noise 8 kHz = 13 +// Comfort noise 16 kHz = 98 +// Comfort noise 32 kHz = 99 +// iLBC = 102 +// iSAC wideband = 103 +// iSAC super-wideband = 104 +// iSAC fullband = 124 +// AVT/DTMF = 106 +// RED = 117 +// PCM16b 8 kHz = 93 +// PCM16b 16 kHz = 94 +// PCM16b 32 kHz = 95 +// G.722 = 94 +bool RemapPltypeAndUseThisCodec(const char* plname, + int plfreq, + int channels, + int* pltype) { + if (channels != 1) + return false; // Don't use non-mono codecs. + + // Re-map pltypes to those used in the NetEq test files. + if (STR_CASE_CMP(plname, "PCMU") == 0 && plfreq == 8000) { + *pltype = 0; + } else if (STR_CASE_CMP(plname, "PCMA") == 0 && plfreq == 8000) { + *pltype = 8; + } else if (STR_CASE_CMP(plname, "CN") == 0 && plfreq == 8000) { + *pltype = 13; + } else if (STR_CASE_CMP(plname, "CN") == 0 && plfreq == 16000) { + *pltype = 98; + } else if (STR_CASE_CMP(plname, "CN") == 0 && plfreq == 32000) { + *pltype = 99; + } else if (STR_CASE_CMP(plname, "ILBC") == 0) { + *pltype = 102; + } else if (STR_CASE_CMP(plname, "ISAC") == 0 && plfreq == 16000) { + *pltype = 103; + } else if (STR_CASE_CMP(plname, "ISAC") == 0 && plfreq == 32000) { + *pltype = 104; + } else if (STR_CASE_CMP(plname, "ISAC") == 0 && plfreq == 48000) { + *pltype = 124; + } else if (STR_CASE_CMP(plname, "telephone-event") == 0) { + *pltype = 106; + } else if (STR_CASE_CMP(plname, "red") == 0) { + *pltype = 117; + } else if (STR_CASE_CMP(plname, "L16") == 0 && plfreq == 8000) { + *pltype = 93; + } else if (STR_CASE_CMP(plname, "L16") == 0 && plfreq == 16000) { + *pltype = 94; + } else if (STR_CASE_CMP(plname, "L16") == 0 && plfreq == 32000) { + *pltype = 95; + } else if (STR_CASE_CMP(plname, "G722") == 0) { + *pltype = 9; + } else { + // Don't use any other codecs. + return false; + } + return true; +} +} // namespace + +AcmReceiveTestOldApi::AcmReceiveTestOldApi( + PacketSource* packet_source, + AudioSink* audio_sink, + int output_freq_hz, + NumOutputChannels exptected_output_channels) + : clock_(0), + acm_(webrtc::AudioCodingModule::Create(0, &clock_)), + packet_source_(packet_source), + audio_sink_(audio_sink), + output_freq_hz_(output_freq_hz), + exptected_output_channels_(exptected_output_channels) { +} + +void AcmReceiveTestOldApi::RegisterDefaultCodecs() { + CodecInst my_codec_param; + for (int n = 0; n < acm_->NumberOfCodecs(); n++) { + ASSERT_EQ(0, acm_->Codec(n, &my_codec_param)) << "Failed to get codec."; + if (ModifyAndUseThisCodec(&my_codec_param)) { + ASSERT_EQ(0, acm_->RegisterReceiveCodec(my_codec_param)) + << "Couldn't register receive codec.\n"; + } + } +} + +void AcmReceiveTestOldApi::RegisterNetEqTestCodecs() { + CodecInst my_codec_param; + for (int n = 0; n < acm_->NumberOfCodecs(); n++) { + ASSERT_EQ(0, acm_->Codec(n, &my_codec_param)) << "Failed to get codec."; + if (!ModifyAndUseThisCodec(&my_codec_param)) { + // Skip this codec. + continue; + } + + if (RemapPltypeAndUseThisCodec(my_codec_param.plname, + my_codec_param.plfreq, + my_codec_param.channels, + &my_codec_param.pltype)) { + ASSERT_EQ(0, acm_->RegisterReceiveCodec(my_codec_param)) + << "Couldn't register receive codec.\n"; + } + } +} + +void AcmReceiveTestOldApi::Run() { + for (scoped_ptr<Packet> packet(packet_source_->NextPacket()); packet; + packet.reset(packet_source_->NextPacket())) { + // Pull audio until time to insert packet. + while (clock_.TimeInMilliseconds() < packet->time_ms()) { + AudioFrame output_frame; + EXPECT_EQ(0, acm_->PlayoutData10Ms(output_freq_hz_, &output_frame)); + EXPECT_EQ(output_freq_hz_, output_frame.sample_rate_hz_); + const int samples_per_block = output_freq_hz_ * 10 / 1000; + EXPECT_EQ(samples_per_block, output_frame.samples_per_channel_); + if (exptected_output_channels_ != kArbitraryChannels) { + if (output_frame.speech_type_ == webrtc::AudioFrame::kPLC) { + // Don't check number of channels for PLC output, since each test run + // usually starts with a short period of mono PLC before decoding the + // first packet. + } else { + EXPECT_EQ(exptected_output_channels_, output_frame.num_channels_); + } + } + ASSERT_TRUE(audio_sink_->WriteAudioFrame(output_frame)); + clock_.AdvanceTimeMilliseconds(10); + } + + // Insert packet after converting from RTPHeader to WebRtcRTPHeader. + WebRtcRTPHeader header; + header.header = packet->header(); + header.frameType = kAudioFrameSpeech; + memset(&header.type.Audio, 0, sizeof(RTPAudioHeader)); + EXPECT_EQ(0, + acm_->IncomingPacket( + packet->payload(), + static_cast<int32_t>(packet->payload_length_bytes()), + header)) + << "Failure when inserting packet:" << std::endl + << " PT = " << static_cast<int>(header.header.payloadType) << std::endl + << " TS = " << header.header.timestamp << std::endl + << " SN = " << header.header.sequenceNumber; + } +} + +} // namespace test +} // namespace webrtc diff --git a/modules/audio_coding/main/acm2/acm_receive_test_oldapi.h b/modules/audio_coding/main/acm2/acm_receive_test_oldapi.h new file mode 100644 index 00000000..795893c2 --- /dev/null +++ b/modules/audio_coding/main/acm2/acm_receive_test_oldapi.h @@ -0,0 +1,63 @@ +/* + * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_RECEIVE_TEST_H_ +#define WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_RECEIVE_TEST_H_ + +#include "webrtc/base/constructormagic.h" +#include "webrtc/system_wrappers/interface/clock.h" +#include "webrtc/system_wrappers/interface/scoped_ptr.h" + +namespace webrtc { +class AudioCodingModule; +struct CodecInst; + +namespace test { +class AudioSink; +class PacketSource; + +class AcmReceiveTestOldApi { + public: + enum NumOutputChannels { + kArbitraryChannels = 0, + kMonoOutput = 1, + kStereoOutput = 2 + }; + + AcmReceiveTestOldApi(PacketSource* packet_source, + AudioSink* audio_sink, + int output_freq_hz, + NumOutputChannels exptected_output_channels); + virtual ~AcmReceiveTestOldApi() {} + + // Registers the codecs with default parameters from ACM. + void RegisterDefaultCodecs(); + + // Registers codecs with payload types matching the pre-encoded NetEq test + // files. + void RegisterNetEqTestCodecs(); + + // Runs the test and returns true if successful. + void Run(); + + private: + SimulatedClock clock_; + scoped_ptr<AudioCodingModule> acm_; + PacketSource* packet_source_; + AudioSink* audio_sink_; + const int output_freq_hz_; + NumOutputChannels exptected_output_channels_; + + DISALLOW_COPY_AND_ASSIGN(AcmReceiveTestOldApi); +}; + +} // namespace test +} // namespace webrtc +#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_RECEIVE_TEST_H_ diff --git a/modules/audio_coding/main/acm2/acm_receiver.h b/modules/audio_coding/main/acm2/acm_receiver.h index 180b4ba6..94ea5b01 100644 --- a/modules/audio_coding/main/acm2/acm_receiver.h +++ b/modules/audio_coding/main/acm2/acm_receiver.h @@ -13,6 +13,7 @@ #include <vector> +#include "webrtc/base/thread_annotations.h" #include "webrtc/common_audio/vad/include/webrtc_vad.h" #include "webrtc/engine_configurations.h" #include "webrtc/modules/audio_coding/main/interface/audio_coding_module.h" @@ -23,7 +24,6 @@ #include "webrtc/modules/audio_coding/neteq/interface/neteq.h" #include "webrtc/modules/interface/module_common_types.h" #include "webrtc/system_wrappers/interface/scoped_ptr.h" -#include "webrtc/system_wrappers/interface/thread_annotations.h" #include "webrtc/typedefs.h" namespace webrtc { diff --git a/modules/audio_coding/main/acm2/acm_receiver_unittest.cc b/modules/audio_coding/main/acm2/acm_receiver_unittest.cc index 94d51b77..9cfef3a8 100644 --- a/modules/audio_coding/main/acm2/acm_receiver_unittest.cc +++ b/modules/audio_coding/main/acm2/acm_receiver_unittest.cc @@ -47,9 +47,10 @@ class AcmReceiverTest : public AudioPacketizationCallback, packet_sent_(false), last_packet_send_timestamp_(timestamp_), last_frame_type_(kFrameEmpty) { - AudioCodingModule::Config config; - acm_.reset(new AudioCodingModuleImpl(config)); - receiver_.reset(new AcmReceiver(config)); + AudioCoding::Config config; + config.transport = this; + acm_.reset(new AudioCodingImpl(config)); + receiver_.reset(new AcmReceiver(config.ToOldConfig())); } ~AcmReceiverTest() {} @@ -61,10 +62,6 @@ class AcmReceiverTest : public AudioPacketizationCallback, ASSERT_EQ(0, ACMCodecDB::Codec(n, &codecs_[n])); } - acm_->InitializeReceiver(); - acm_->InitializeSender(); - acm_->RegisterTransportCallback(this); - rtp_header_.header.sequenceNumber = 0; rtp_header_.header.timestamp = 0; rtp_header_.header.markerBit = false; @@ -82,12 +79,12 @@ class AcmReceiverTest : public AudioPacketizationCallback, CodecInst codec; ACMCodecDB::Codec(codec_id, &codec); if (timestamp_ == 0) { // This is the first time inserting audio. - ASSERT_EQ(0, acm_->RegisterSendCodec(codec)); + ASSERT_TRUE(acm_->RegisterSendCodec(codec_id, codec.pltype)); } else { - CodecInst current_codec; - ASSERT_EQ(0, acm_->SendCodec(¤t_codec)); - if (!CodecsEqual(codec, current_codec)) - ASSERT_EQ(0, acm_->RegisterSendCodec(codec)); + const CodecInst* current_codec = acm_->GetSenderCodecInst(); + ASSERT_TRUE(current_codec); + if (!CodecsEqual(codec, *current_codec)) + ASSERT_TRUE(acm_->RegisterSendCodec(codec_id, codec.pltype)); } AudioFrame frame; // Frame setup according to the codec. @@ -102,8 +99,7 @@ class AcmReceiverTest : public AudioPacketizationCallback, while (num_bytes == 0) { frame.timestamp_ = timestamp_; timestamp_ += frame.samples_per_channel_; - ASSERT_EQ(0, acm_->Add10MsData(frame)); - num_bytes = acm_->Process(); + num_bytes = acm_->Add10MsAudio(frame); ASSERT_GE(num_bytes, 0); } ASSERT_TRUE(packet_sent_); // Sanity check. @@ -151,7 +147,7 @@ class AcmReceiverTest : public AudioPacketizationCallback, scoped_ptr<AcmReceiver> receiver_; CodecInst codecs_[ACMCodecDB::kMaxNumCodecs]; - scoped_ptr<AudioCodingModule> acm_; + scoped_ptr<AudioCoding> acm_; WebRtcRTPHeader rtp_header_; uint32_t timestamp_; bool packet_sent_; // Set when SendData is called reset when inserting audio. @@ -307,7 +303,7 @@ TEST_F(AcmReceiverTest, DISABLED_ON_ANDROID(LastAudioCodec)) { // Register CNG at sender side. int n = 0; while (kCngId[n] > 0) { - ASSERT_EQ(0, acm_->RegisterSendCodec(codecs_[kCngId[n]])); + ASSERT_TRUE(acm_->RegisterSendCodec(kCngId[n], codecs_[kCngId[n]].pltype)); ++n; } @@ -316,7 +312,7 @@ TEST_F(AcmReceiverTest, DISABLED_ON_ANDROID(LastAudioCodec)) { EXPECT_EQ(-1, receiver_->LastAudioCodec(&codec)); // Start with sending DTX. - ASSERT_EQ(0, acm_->SetVAD(true, true, VADVeryAggr)); + ASSERT_TRUE(acm_->SetVad(true, true, VADVeryAggr)); packet_sent_ = false; InsertOnePacketOfSilence(kCodecId[0]); // Enough to test with one codec. ASSERT_TRUE(packet_sent_); @@ -330,7 +326,7 @@ TEST_F(AcmReceiverTest, DISABLED_ON_ANDROID(LastAudioCodec)) { n = 0; while (kCodecId[n] >= 0) { // Loop over codecs. // Set DTX off to send audio payload. - acm_->SetVAD(false, false, VADAggr); + acm_->SetVad(false, false, VADAggr); packet_sent_ = false; InsertOnePacketOfSilence(kCodecId[n]); @@ -342,7 +338,7 @@ TEST_F(AcmReceiverTest, DISABLED_ON_ANDROID(LastAudioCodec)) { // Set VAD on to send DTX. Then check if the "Last Audio codec" returns // the expected codec. - acm_->SetVAD(true, true, VADAggr); + acm_->SetVad(true, true, VADAggr); // Do as many encoding until a DTX is sent. while (last_frame_type_ != kAudioFrameCN) { diff --git a/modules/audio_coding/main/acm2/acm_receiver_unittest_oldapi.cc b/modules/audio_coding/main/acm2/acm_receiver_unittest_oldapi.cc new file mode 100644 index 00000000..ef890ecb --- /dev/null +++ b/modules/audio_coding/main/acm2/acm_receiver_unittest_oldapi.cc @@ -0,0 +1,364 @@ +/* + * Copyright (c) 2013 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#include "webrtc/modules/audio_coding/main/acm2/acm_receiver.h" + +#include <algorithm> // std::min + +#include "testing/gtest/include/gtest/gtest.h" +#include "webrtc/modules/audio_coding/main/interface/audio_coding_module.h" +#include "webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.h" +#include "webrtc/modules/audio_coding/main/acm2/acm_codec_database.h" +#include "webrtc/modules/audio_coding/neteq/tools/rtp_generator.h" +#include "webrtc/system_wrappers/interface/clock.h" +#include "webrtc/system_wrappers/interface/scoped_ptr.h" +#include "webrtc/test/test_suite.h" +#include "webrtc/test/testsupport/fileutils.h" +#include "webrtc/test/testsupport/gtest_disable.h" + +namespace webrtc { + +namespace acm2 { +namespace { + +bool CodecsEqual(const CodecInst& codec_a, const CodecInst& codec_b) { + if (strcmp(codec_a.plname, codec_b.plname) != 0 || + codec_a.plfreq != codec_b.plfreq || + codec_a.pltype != codec_b.pltype || + codec_b.channels != codec_a.channels) + return false; + return true; +} + +} // namespace + +class AcmReceiverTestOldApi : public AudioPacketizationCallback, + public ::testing::Test { + protected: + AcmReceiverTestOldApi() + : timestamp_(0), + packet_sent_(false), + last_packet_send_timestamp_(timestamp_), + last_frame_type_(kFrameEmpty) { + AudioCodingModule::Config config; + acm_.reset(new AudioCodingModuleImpl(config)); + receiver_.reset(new AcmReceiver(config)); + } + + ~AcmReceiverTestOldApi() {} + + virtual void SetUp() OVERRIDE { + ASSERT_TRUE(receiver_.get() != NULL); + ASSERT_TRUE(acm_.get() != NULL); + for (int n = 0; n < ACMCodecDB::kNumCodecs; n++) { + ASSERT_EQ(0, ACMCodecDB::Codec(n, &codecs_[n])); + } + + acm_->InitializeReceiver(); + acm_->InitializeSender(); + acm_->RegisterTransportCallback(this); + + rtp_header_.header.sequenceNumber = 0; + rtp_header_.header.timestamp = 0; + rtp_header_.header.markerBit = false; + rtp_header_.header.ssrc = 0x12345678; // Arbitrary. + rtp_header_.header.numCSRCs = 0; + rtp_header_.header.payloadType = 0; + rtp_header_.frameType = kAudioFrameSpeech; + rtp_header_.type.Audio.isCNG = false; + } + + virtual void TearDown() OVERRIDE { + } + + void InsertOnePacketOfSilence(int codec_id) { + CodecInst codec; + ACMCodecDB::Codec(codec_id, &codec); + if (timestamp_ == 0) { // This is the first time inserting audio. + ASSERT_EQ(0, acm_->RegisterSendCodec(codec)); + } else { + CodecInst current_codec; + ASSERT_EQ(0, acm_->SendCodec(¤t_codec)); + if (!CodecsEqual(codec, current_codec)) + ASSERT_EQ(0, acm_->RegisterSendCodec(codec)); + } + AudioFrame frame; + // Frame setup according to the codec. + frame.sample_rate_hz_ = codec.plfreq; + frame.samples_per_channel_ = codec.plfreq / 100; // 10 ms. + frame.num_channels_ = codec.channels; + memset(frame.data_, 0, frame.samples_per_channel_ * frame.num_channels_ * + sizeof(int16_t)); + int num_bytes = 0; + packet_sent_ = false; + last_packet_send_timestamp_ = timestamp_; + while (num_bytes == 0) { + frame.timestamp_ = timestamp_; + timestamp_ += frame.samples_per_channel_; + ASSERT_EQ(0, acm_->Add10MsData(frame)); + num_bytes = acm_->Process(); + ASSERT_GE(num_bytes, 0); + } + ASSERT_TRUE(packet_sent_); // Sanity check. + } + + // Last element of id should be negative. + void AddSetOfCodecs(const int* id) { + int n = 0; + while (id[n] >= 0) { + ASSERT_EQ(0, receiver_->AddCodec(id[n], codecs_[id[n]].pltype, + codecs_[id[n]].channels, NULL)); + ++n; + } + } + + virtual int SendData( + FrameType frame_type, + uint8_t payload_type, + uint32_t timestamp, + const uint8_t* payload_data, + uint16_t payload_len_bytes, + const RTPFragmentationHeader* fragmentation) OVERRIDE { + if (frame_type == kFrameEmpty) + return 0; + + rtp_header_.header.payloadType = payload_type; + rtp_header_.frameType = frame_type; + if (frame_type == kAudioFrameSpeech) + rtp_header_.type.Audio.isCNG = false; + else + rtp_header_.type.Audio.isCNG = true; + rtp_header_.header.timestamp = timestamp; + + int ret_val = receiver_->InsertPacket(rtp_header_, payload_data, + payload_len_bytes); + if (ret_val < 0) { + assert(false); + return -1; + } + rtp_header_.header.sequenceNumber++; + packet_sent_ = true; + last_frame_type_ = frame_type; + return 0; + } + + scoped_ptr<AcmReceiver> receiver_; + CodecInst codecs_[ACMCodecDB::kMaxNumCodecs]; + scoped_ptr<AudioCodingModule> acm_; + WebRtcRTPHeader rtp_header_; + uint32_t timestamp_; + bool packet_sent_; // Set when SendData is called reset when inserting audio. + uint32_t last_packet_send_timestamp_; + FrameType last_frame_type_; +}; + +TEST_F(AcmReceiverTestOldApi, DISABLED_ON_ANDROID(AddCodecGetCodec)) { + // Add codec. + for (int n = 0; n < ACMCodecDB::kNumCodecs; ++n) { + if (n & 0x1) // Just add codecs with odd index. + EXPECT_EQ(0, receiver_->AddCodec(n, codecs_[n].pltype, + codecs_[n].channels, NULL)); + } + // Get codec and compare. + for (int n = 0; n < ACMCodecDB::kNumCodecs; ++n) { + CodecInst my_codec; + if (n & 0x1) { + // Codecs with odd index should match the reference. + EXPECT_EQ(0, receiver_->DecoderByPayloadType(codecs_[n].pltype, + &my_codec)); + EXPECT_TRUE(CodecsEqual(codecs_[n], my_codec)); + } else { + // Codecs with even index are not registered. + EXPECT_EQ(-1, receiver_->DecoderByPayloadType(codecs_[n].pltype, + &my_codec)); + } + } +} + +TEST_F(AcmReceiverTestOldApi, DISABLED_ON_ANDROID(AddCodecChangePayloadType)) { + CodecInst ref_codec; + const int codec_id = ACMCodecDB::kPCMA; + EXPECT_EQ(0, ACMCodecDB::Codec(codec_id, &ref_codec)); + const int payload_type = ref_codec.pltype; + EXPECT_EQ(0, receiver_->AddCodec(codec_id, ref_codec.pltype, + ref_codec.channels, NULL)); + CodecInst test_codec; + EXPECT_EQ(0, receiver_->DecoderByPayloadType(payload_type, &test_codec)); + EXPECT_EQ(true, CodecsEqual(ref_codec, test_codec)); + + // Re-register the same codec with different payload. + ref_codec.pltype = payload_type + 1; + EXPECT_EQ(0, receiver_->AddCodec(codec_id, ref_codec.pltype, + ref_codec.channels, NULL)); + + // Payload type |payload_type| should not exist. + EXPECT_EQ(-1, receiver_->DecoderByPayloadType(payload_type, &test_codec)); + + // Payload type |payload_type + 1| should exist. + EXPECT_EQ(0, receiver_->DecoderByPayloadType(payload_type + 1, &test_codec)); + EXPECT_TRUE(CodecsEqual(test_codec, ref_codec)); +} + +TEST_F(AcmReceiverTestOldApi, DISABLED_ON_ANDROID(AddCodecRemoveCodec)) { + CodecInst codec; + const int codec_id = ACMCodecDB::kPCMA; + EXPECT_EQ(0, ACMCodecDB::Codec(codec_id, &codec)); + const int payload_type = codec.pltype; + EXPECT_EQ(0, receiver_->AddCodec(codec_id, codec.pltype, + codec.channels, NULL)); + + // Remove non-existing codec should not fail. ACM1 legacy. + EXPECT_EQ(0, receiver_->RemoveCodec(payload_type + 1)); + + // Remove an existing codec. + EXPECT_EQ(0, receiver_->RemoveCodec(payload_type)); + + // Ask for the removed codec, must fail. + EXPECT_EQ(-1, receiver_->DecoderByPayloadType(payload_type, &codec)); +} + +TEST_F(AcmReceiverTestOldApi, DISABLED_ON_ANDROID(SampleRate)) { + const int kCodecId[] = { + ACMCodecDB::kISAC, ACMCodecDB::kISACSWB, ACMCodecDB::kISACFB, + -1 // Terminator. + }; + AddSetOfCodecs(kCodecId); + + AudioFrame frame; + const int kOutSampleRateHz = 8000; // Different than codec sample rate. + int n = 0; + while (kCodecId[n] >= 0) { + const int num_10ms_frames = codecs_[kCodecId[n]].pacsize / + (codecs_[kCodecId[n]].plfreq / 100); + InsertOnePacketOfSilence(kCodecId[n]); + for (int k = 0; k < num_10ms_frames; ++k) { + EXPECT_EQ(0, receiver_->GetAudio(kOutSampleRateHz, &frame)); + } + EXPECT_EQ(std::min(32000, codecs_[kCodecId[n]].plfreq), + receiver_->current_sample_rate_hz()); + ++n; + } +} + +// Verify that the playout mode is set correctly. +TEST_F(AcmReceiverTestOldApi, DISABLED_ON_ANDROID(PlayoutMode)) { + receiver_->SetPlayoutMode(voice); + EXPECT_EQ(voice, receiver_->PlayoutMode()); + + receiver_->SetPlayoutMode(streaming); + EXPECT_EQ(streaming, receiver_->PlayoutMode()); + + receiver_->SetPlayoutMode(fax); + EXPECT_EQ(fax, receiver_->PlayoutMode()); + + receiver_->SetPlayoutMode(off); + EXPECT_EQ(off, receiver_->PlayoutMode()); +} + +TEST_F(AcmReceiverTestOldApi, DISABLED_ON_ANDROID(PostdecodingVad)) { + receiver_->EnableVad(); + EXPECT_TRUE(receiver_->vad_enabled()); + + const int id = ACMCodecDB::kPCM16Bwb; + ASSERT_EQ(0, receiver_->AddCodec(id, codecs_[id].pltype, codecs_[id].channels, + NULL)); + const int kNumPackets = 5; + const int num_10ms_frames = codecs_[id].pacsize / (codecs_[id].plfreq / 100); + AudioFrame frame; + for (int n = 0; n < kNumPackets; ++n) { + InsertOnePacketOfSilence(id); + for (int k = 0; k < num_10ms_frames; ++k) + ASSERT_EQ(0, receiver_->GetAudio(codecs_[id].plfreq, &frame)); + } + EXPECT_EQ(AudioFrame::kVadPassive, frame.vad_activity_); + + receiver_->DisableVad(); + EXPECT_FALSE(receiver_->vad_enabled()); + + for (int n = 0; n < kNumPackets; ++n) { + InsertOnePacketOfSilence(id); + for (int k = 0; k < num_10ms_frames; ++k) + ASSERT_EQ(0, receiver_->GetAudio(codecs_[id].plfreq, &frame)); + } + EXPECT_EQ(AudioFrame::kVadUnknown, frame.vad_activity_); +} + +TEST_F(AcmReceiverTestOldApi, DISABLED_ON_ANDROID(LastAudioCodec)) { + const int kCodecId[] = { + ACMCodecDB::kISAC, ACMCodecDB::kPCMA, ACMCodecDB::kISACSWB, + ACMCodecDB::kPCM16Bswb32kHz, ACMCodecDB::kG722_1C_48, + -1 // Terminator. + }; + AddSetOfCodecs(kCodecId); + + const int kCngId[] = { // Not including full-band. + ACMCodecDB::kCNNB, ACMCodecDB::kCNWB, ACMCodecDB::kCNSWB, + -1 // Terminator. + }; + AddSetOfCodecs(kCngId); + + // Register CNG at sender side. + int n = 0; + while (kCngId[n] > 0) { + ASSERT_EQ(0, acm_->RegisterSendCodec(codecs_[kCngId[n]])); + ++n; + } + + CodecInst codec; + // No audio payload is received. + EXPECT_EQ(-1, receiver_->LastAudioCodec(&codec)); + + // Start with sending DTX. + ASSERT_EQ(0, acm_->SetVAD(true, true, VADVeryAggr)); + packet_sent_ = false; + InsertOnePacketOfSilence(kCodecId[0]); // Enough to test with one codec. + ASSERT_TRUE(packet_sent_); + EXPECT_EQ(kAudioFrameCN, last_frame_type_); + + // Has received, only, DTX. Last Audio codec is undefined. + EXPECT_EQ(-1, receiver_->LastAudioCodec(&codec)); + EXPECT_EQ(-1, receiver_->last_audio_codec_id()); + EXPECT_EQ(-1, receiver_->last_audio_payload_type()); + + n = 0; + while (kCodecId[n] >= 0) { // Loop over codecs. + // Set DTX off to send audio payload. + acm_->SetVAD(false, false, VADAggr); + packet_sent_ = false; + InsertOnePacketOfSilence(kCodecId[n]); + + // Sanity check if Actually an audio payload received, and it should be + // of type "speech." + ASSERT_TRUE(packet_sent_); + ASSERT_EQ(kAudioFrameSpeech, last_frame_type_); + EXPECT_EQ(kCodecId[n], receiver_->last_audio_codec_id()); + + // Set VAD on to send DTX. Then check if the "Last Audio codec" returns + // the expected codec. + acm_->SetVAD(true, true, VADAggr); + + // Do as many encoding until a DTX is sent. + while (last_frame_type_ != kAudioFrameCN) { + packet_sent_ = false; + InsertOnePacketOfSilence(kCodecId[n]); + ASSERT_TRUE(packet_sent_); + } + EXPECT_EQ(kCodecId[n], receiver_->last_audio_codec_id()); + EXPECT_EQ(codecs_[kCodecId[n]].pltype, + receiver_->last_audio_payload_type()); + EXPECT_EQ(0, receiver_->LastAudioCodec(&codec)); + EXPECT_TRUE(CodecsEqual(codecs_[kCodecId[n]], codec)); + ++n; + } +} + +} // namespace acm2 + +} // namespace webrtc diff --git a/modules/audio_coding/main/acm2/acm_send_test.cc b/modules/audio_coding/main/acm2/acm_send_test.cc index 30bf2fd3..ec3c2540 100644 --- a/modules/audio_coding/main/acm2/acm_send_test.cc +++ b/modules/audio_coding/main/acm2/acm_send_test.cc @@ -27,7 +27,6 @@ AcmSendTest::AcmSendTest(InputAudioFile* audio_source, int source_rate_hz, int test_duration_ms) : clock_(0), - acm_(webrtc::AudioCodingModule::Create(0, &clock_)), audio_source_(audio_source), source_rate_hz_(source_rate_hz), input_block_size_samples_(source_rate_hz_ * kBlockSizeMs / 1000), @@ -37,24 +36,23 @@ AcmSendTest::AcmSendTest(InputAudioFile* audio_source, payload_type_(0), timestamp_(0), sequence_number_(0) { + webrtc::AudioCoding::Config config; + config.clock = &clock_; + config.transport = this; + acm_.reset(webrtc::AudioCoding::Create(config)); input_frame_.sample_rate_hz_ = source_rate_hz_; input_frame_.num_channels_ = 1; input_frame_.samples_per_channel_ = input_block_size_samples_; assert(input_block_size_samples_ * input_frame_.num_channels_ <= AudioFrame::kMaxDataSizeSamples); - acm_->RegisterTransportCallback(this); } -bool AcmSendTest::RegisterCodec(const char* payload_name, - int sampling_freq_hz, +bool AcmSendTest::RegisterCodec(int codec_type, int channels, int payload_type, int frame_size_samples) { - CHECK_EQ(0, AudioCodingModule::Codec(payload_name, &codec_, sampling_freq_hz, - channels)); - codec_.pltype = payload_type; - codec_.pacsize = frame_size_samples; - codec_registered_ = (acm_->RegisterSendCodec(codec_) == 0); + codec_registered_ = + acm_->RegisterSendCodec(codec_type, payload_type, frame_size_samples); input_frame_.num_channels_ = channels; assert(input_block_size_samples_ * input_frame_.num_channels_ <= AudioFrame::kMaxDataSizeSamples); @@ -79,9 +77,9 @@ Packet* AcmSendTest::NextPacket() { input_frame_.num_channels_, input_frame_.data_); } - CHECK_EQ(0, acm_->Add10MsData(input_frame_)); + int32_t encoded_bytes = acm_->Add10MsAudio(input_frame_); + EXPECT_GE(encoded_bytes, 0); input_frame_.timestamp_ += input_block_size_samples_; - int32_t encoded_bytes = acm_->Process(); if (encoded_bytes > 0) { // Encoded packet received. return CreatePacket(); diff --git a/modules/audio_coding/main/acm2/acm_send_test.h b/modules/audio_coding/main/acm2/acm_send_test.h index db5d9e55..8bc0cde0 100644 --- a/modules/audio_coding/main/acm2/acm_send_test.h +++ b/modules/audio_coding/main/acm2/acm_send_test.h @@ -33,8 +33,7 @@ class AcmSendTest : public AudioPacketizationCallback, public PacketSource { virtual ~AcmSendTest() {} // Registers the send codec. Returns true on success, false otherwise. - bool RegisterCodec(const char* payload_name, - int sampling_freq_hz, + bool RegisterCodec(int codec_type, int channels, int payload_type, int frame_size_samples); @@ -62,12 +61,11 @@ class AcmSendTest : public AudioPacketizationCallback, public PacketSource { Packet* CreatePacket(); SimulatedClock clock_; - scoped_ptr<AudioCodingModule> acm_; + scoped_ptr<AudioCoding> acm_; InputAudioFile* audio_source_; int source_rate_hz_; const int input_block_size_samples_; AudioFrame input_frame_; - CodecInst codec_; bool codec_registered_; int test_duration_ms_; // The following member variables are set whenever SendData() is called. diff --git a/modules/audio_coding/main/acm2/acm_send_test_oldapi.cc b/modules/audio_coding/main/acm2/acm_send_test_oldapi.cc new file mode 100644 index 00000000..2f5178e0 --- /dev/null +++ b/modules/audio_coding/main/acm2/acm_send_test_oldapi.cc @@ -0,0 +1,145 @@ +/* + * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#include "webrtc/modules/audio_coding/main/acm2/acm_send_test_oldapi.h" + +#include <assert.h> +#include <stdio.h> +#include <string.h> + +#include "testing/gtest/include/gtest/gtest.h" +#include "webrtc/base/checks.h" +#include "webrtc/modules/audio_coding/main/interface/audio_coding_module.h" +#include "webrtc/modules/audio_coding/neteq/tools/input_audio_file.h" +#include "webrtc/modules/audio_coding/neteq/tools/packet.h" + +namespace webrtc { +namespace test { + +AcmSendTestOldApi::AcmSendTestOldApi(InputAudioFile* audio_source, + int source_rate_hz, + int test_duration_ms) + : clock_(0), + acm_(webrtc::AudioCodingModule::Create(0, &clock_)), + audio_source_(audio_source), + source_rate_hz_(source_rate_hz), + input_block_size_samples_(source_rate_hz_ * kBlockSizeMs / 1000), + codec_registered_(false), + test_duration_ms_(test_duration_ms), + frame_type_(kAudioFrameSpeech), + payload_type_(0), + timestamp_(0), + sequence_number_(0) { + input_frame_.sample_rate_hz_ = source_rate_hz_; + input_frame_.num_channels_ = 1; + input_frame_.samples_per_channel_ = input_block_size_samples_; + assert(input_block_size_samples_ * input_frame_.num_channels_ <= + AudioFrame::kMaxDataSizeSamples); + acm_->RegisterTransportCallback(this); +} + +bool AcmSendTestOldApi::RegisterCodec(const char* payload_name, + int sampling_freq_hz, + int channels, + int payload_type, + int frame_size_samples) { + CHECK_EQ(0, + AudioCodingModule::Codec( + payload_name, &codec_, sampling_freq_hz, channels)); + codec_.pltype = payload_type; + codec_.pacsize = frame_size_samples; + codec_registered_ = (acm_->RegisterSendCodec(codec_) == 0); + input_frame_.num_channels_ = channels; + assert(input_block_size_samples_ * input_frame_.num_channels_ <= + AudioFrame::kMaxDataSizeSamples); + return codec_registered_; +} + +Packet* AcmSendTestOldApi::NextPacket() { + assert(codec_registered_); + if (filter_.test(payload_type_)) { + // This payload type should be filtered out. Since the payload type is the + // same throughout the whole test run, no packet at all will be delivered. + // We can just as well signal that the test is over by returning NULL. + return NULL; + } + // Insert audio and process until one packet is produced. + while (clock_.TimeInMilliseconds() < test_duration_ms_) { + clock_.AdvanceTimeMilliseconds(kBlockSizeMs); + CHECK(audio_source_->Read(input_block_size_samples_, input_frame_.data_)); + if (input_frame_.num_channels_ > 1) { + InputAudioFile::DuplicateInterleaved(input_frame_.data_, + input_block_size_samples_, + input_frame_.num_channels_, + input_frame_.data_); + } + CHECK_EQ(0, acm_->Add10MsData(input_frame_)); + input_frame_.timestamp_ += input_block_size_samples_; + int32_t encoded_bytes = acm_->Process(); + if (encoded_bytes > 0) { + // Encoded packet received. + return CreatePacket(); + } + } + // Test ended. + return NULL; +} + +// This method receives the callback from ACM when a new packet is produced. +int32_t AcmSendTestOldApi::SendData( + FrameType frame_type, + uint8_t payload_type, + uint32_t timestamp, + const uint8_t* payload_data, + uint16_t payload_len_bytes, + const RTPFragmentationHeader* fragmentation) { + // Store the packet locally. + frame_type_ = frame_type; + payload_type_ = payload_type; + timestamp_ = timestamp; + last_payload_vec_.assign(payload_data, payload_data + payload_len_bytes); + assert(last_payload_vec_.size() == payload_len_bytes); + return 0; +} + +Packet* AcmSendTestOldApi::CreatePacket() { + const size_t kRtpHeaderSize = 12; + size_t allocated_bytes = last_payload_vec_.size() + kRtpHeaderSize; + uint8_t* packet_memory = new uint8_t[allocated_bytes]; + // Populate the header bytes. + packet_memory[0] = 0x80; + packet_memory[1] = payload_type_; + packet_memory[2] = (sequence_number_ >> 8) & 0xFF; + packet_memory[3] = (sequence_number_) & 0xFF; + packet_memory[4] = (timestamp_ >> 24) & 0xFF; + packet_memory[5] = (timestamp_ >> 16) & 0xFF; + packet_memory[6] = (timestamp_ >> 8) & 0xFF; + packet_memory[7] = timestamp_ & 0xFF; + // Set SSRC to 0x12345678. + packet_memory[8] = 0x12; + packet_memory[9] = 0x34; + packet_memory[10] = 0x56; + packet_memory[11] = 0x78; + + ++sequence_number_; + + // Copy the payload data. + memcpy(packet_memory + kRtpHeaderSize, + &last_payload_vec_[0], + last_payload_vec_.size()); + Packet* packet = + new Packet(packet_memory, allocated_bytes, clock_.TimeInMilliseconds()); + assert(packet); + assert(packet->valid_header()); + return packet; +} + +} // namespace test +} // namespace webrtc diff --git a/modules/audio_coding/main/acm2/acm_send_test_oldapi.h b/modules/audio_coding/main/acm2/acm_send_test_oldapi.h new file mode 100644 index 00000000..ff229a0a --- /dev/null +++ b/modules/audio_coding/main/acm2/acm_send_test_oldapi.h @@ -0,0 +1,86 @@ +/* + * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_SEND_TEST_H_ +#define WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_SEND_TEST_H_ + +#include <vector> + +#include "webrtc/base/constructormagic.h" +#include "webrtc/modules/audio_coding/main/interface/audio_coding_module.h" +#include "webrtc/modules/audio_coding/neteq/tools/packet_source.h" +#include "webrtc/system_wrappers/interface/clock.h" +#include "webrtc/system_wrappers/interface/scoped_ptr.h" + +namespace webrtc { + +namespace test { +class InputAudioFile; +class Packet; + +class AcmSendTestOldApi : public AudioPacketizationCallback, + public PacketSource { + public: + AcmSendTestOldApi(InputAudioFile* audio_source, + int source_rate_hz, + int test_duration_ms); + virtual ~AcmSendTestOldApi() {} + + // Registers the send codec. Returns true on success, false otherwise. + bool RegisterCodec(const char* payload_name, + int sampling_freq_hz, + int channels, + int payload_type, + int frame_size_samples); + + // Returns the next encoded packet. Returns NULL if the test duration was + // exceeded. Ownership of the packet is handed over to the caller. + // Inherited from PacketSource. + Packet* NextPacket(); + + // Inherited from AudioPacketizationCallback. + virtual int32_t SendData( + FrameType frame_type, + uint8_t payload_type, + uint32_t timestamp, + const uint8_t* payload_data, + uint16_t payload_len_bytes, + const RTPFragmentationHeader* fragmentation) OVERRIDE; + + private: + static const int kBlockSizeMs = 10; + + // Creates a Packet object from the last packet produced by ACM (and received + // through the SendData method as a callback). Ownership of the new Packet + // object is transferred to the caller. + Packet* CreatePacket(); + + SimulatedClock clock_; + scoped_ptr<AudioCodingModule> acm_; + InputAudioFile* audio_source_; + int source_rate_hz_; + const int input_block_size_samples_; + AudioFrame input_frame_; + CodecInst codec_; + bool codec_registered_; + int test_duration_ms_; + // The following member variables are set whenever SendData() is called. + FrameType frame_type_; + int payload_type_; + uint32_t timestamp_; + uint16_t sequence_number_; + std::vector<uint8_t> last_payload_vec_; + + DISALLOW_COPY_AND_ASSIGN(AcmSendTestOldApi); +}; + +} // namespace test +} // namespace webrtc +#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_SEND_TEST_H_ diff --git a/modules/audio_coding/main/acm2/audio_coding_module.gypi b/modules/audio_coding/main/acm2/audio_coding_module.gypi index f88dbd37..d746a80b 100644 --- a/modules/audio_coding/main/acm2/audio_coding_module.gypi +++ b/modules/audio_coding/main/acm2/audio_coding_module.gypi @@ -119,7 +119,11 @@ { 'target_name': 'acm_receive_test', 'type': 'static_library', + 'defines': [ + '<@(audio_coding_defines)', + ], 'dependencies': [ + '<@(audio_coding_dependencies)', 'audio_coding_module', 'neteq_unittest_tools', '<(DEPTH)/testing/gtest.gyp:gtest', @@ -127,12 +131,18 @@ 'sources': [ 'acm_receive_test.cc', 'acm_receive_test.h', + 'acm_receive_test_oldapi.cc', + 'acm_receive_test_oldapi.h', ], }, # acm_receive_test { 'target_name': 'acm_send_test', 'type': 'static_library', + 'defines': [ + '<@(audio_coding_defines)', + ], 'dependencies': [ + '<@(audio_coding_dependencies)', 'audio_coding_module', 'neteq_unittest_tools', '<(DEPTH)/testing/gtest.gyp:gtest', @@ -140,6 +150,8 @@ 'sources': [ 'acm_send_test.cc', 'acm_send_test.h', + 'acm_send_test_oldapi.cc', + 'acm_send_test_oldapi.h', ], }, # acm_send_test { diff --git a/modules/audio_coding/main/acm2/audio_coding_module_impl.cc b/modules/audio_coding/main/acm2/audio_coding_module_impl.cc index 2212f83c..687c5b81 100644 --- a/modules/audio_coding/main/acm2/audio_coding_module_impl.cc +++ b/modules/audio_coding/main/acm2/audio_coding_module_impl.cc @@ -2073,6 +2073,13 @@ const AudioEncoder* AudioCodingImpl::GetSenderInfo() const { FATAL() << "Not implemented yet."; } +const CodecInst* AudioCodingImpl::GetSenderCodecInst() { + if (acm_old_->SendCodec(¤t_send_codec_) != 0) { + return NULL; + } + return ¤t_send_codec_; +} + int AudioCodingImpl::Add10MsAudio(const AudioFrame& audio_frame) { if (acm_old_->Add10MsData(audio_frame) != 0) { return -1; @@ -2151,6 +2158,12 @@ void AudioCodingImpl::DisableNack() { FATAL() << "Not implemented yet."; } +bool AudioCodingImpl::SetVad(bool enable_dtx, + bool enable_vad, + ACMVADMode vad_mode) { + return acm_old_->SetVAD(enable_dtx, enable_vad, vad_mode) == 0; +} + std::vector<uint16_t> AudioCodingImpl::GetNackList( int round_trip_time_ms) const { return acm_old_->GetNackList(round_trip_time_ms); diff --git a/modules/audio_coding/main/acm2/audio_coding_module_impl.h b/modules/audio_coding/main/acm2/audio_coding_module_impl.h index 93fd96be..b8d128fe 100644 --- a/modules/audio_coding/main/acm2/audio_coding_module_impl.h +++ b/modules/audio_coding/main/acm2/audio_coding_module_impl.h @@ -13,13 +13,13 @@ #include <vector> +#include "webrtc/base/thread_annotations.h" #include "webrtc/common_types.h" #include "webrtc/engine_configurations.h" #include "webrtc/modules/audio_coding/main/acm2/acm_codec_database.h" #include "webrtc/modules/audio_coding/main/acm2/acm_receiver.h" #include "webrtc/modules/audio_coding/main/acm2/acm_resampler.h" #include "webrtc/system_wrappers/interface/scoped_ptr.h" -#include "webrtc/system_wrappers/interface/thread_annotations.h" namespace webrtc { @@ -390,10 +390,7 @@ class AudioCodingModuleImpl : public AudioCodingModule { class AudioCodingImpl : public AudioCoding { public: AudioCodingImpl(const Config& config) { - AudioCodingModule::Config config_old; - config_old.id = 0; - config_old.neteq_config = config.neteq_config; - config_old.clock = config.clock; + AudioCodingModule::Config config_old = config.ToOldConfig(); acm_old_.reset(new acm2::AudioCodingModuleImpl(config_old)); acm_old_->RegisterTransportCallback(config.transport); acm_old_->RegisterVADCallback(config.vad_callback); @@ -414,6 +411,8 @@ class AudioCodingImpl : public AudioCoding { virtual const AudioEncoder* GetSenderInfo() const OVERRIDE; + virtual const CodecInst* GetSenderCodecInst() OVERRIDE; + virtual int Add10MsAudio(const AudioFrame& audio_frame) OVERRIDE; virtual const ReceiverInfo* GetReceiverInfo() const OVERRIDE; @@ -449,6 +448,10 @@ class AudioCodingImpl : public AudioCoding { virtual void DisableNack() OVERRIDE; + virtual bool SetVad(bool enable_dtx, + bool enable_vad, + ACMVADMode vad_mode) OVERRIDE; + virtual std::vector<uint16_t> GetNackList( int round_trip_time_ms) const OVERRIDE; @@ -465,8 +468,11 @@ class AudioCodingImpl : public AudioCoding { int* sample_rate_hz, int* channels); - scoped_ptr<acm2::AudioCodingModuleImpl> acm_old_; int playout_frequency_hz_; + // TODO(henrik.lundin): All members below this line are temporary and should + // be removed after refactoring is completed. + scoped_ptr<acm2::AudioCodingModuleImpl> acm_old_; + CodecInst current_send_codec_; }; } // namespace webrtc diff --git a/modules/audio_coding/main/acm2/audio_coding_module_unittest.cc b/modules/audio_coding/main/acm2/audio_coding_module_unittest.cc index 009218d1..828b772e 100644 --- a/modules/audio_coding/main/acm2/audio_coding_module_unittest.cc +++ b/modules/audio_coding/main/acm2/audio_coding_module_unittest.cc @@ -12,7 +12,9 @@ #include <vector> #include "testing/gtest/include/gtest/gtest.h" +#include "webrtc/base/checks.h" #include "webrtc/base/md5digest.h" +#include "webrtc/base/thread_annotations.h" #include "webrtc/modules/audio_coding/main/acm2/acm_receive_test.h" #include "webrtc/modules/audio_coding/main/acm2/acm_send_test.h" #include "webrtc/modules/audio_coding/main/interface/audio_coding_module.h" @@ -30,7 +32,6 @@ #include "webrtc/system_wrappers/interface/event_wrapper.h" #include "webrtc/system_wrappers/interface/scoped_ptr.h" #include "webrtc/system_wrappers/interface/sleep.h" -#include "webrtc/system_wrappers/interface/thread_annotations.h" #include "webrtc/system_wrappers/interface/thread_wrapper.h" #include "webrtc/test/testsupport/fileutils.h" #include "webrtc/test/testsupport/gtest_disable.h" @@ -118,19 +119,15 @@ class PacketizationCallbackStub : public AudioPacketizationCallback { class AudioCodingModuleTest : public ::testing::Test { protected: AudioCodingModuleTest() - : id_(1), - rtp_utility_(new RtpUtility(kFrameSizeSamples, kPayloadType)), - clock_(Clock::GetRealTimeClock()) {} + : rtp_utility_(new RtpUtility(kFrameSizeSamples, kPayloadType)) { + config_.transport = &packet_cb_; + } ~AudioCodingModuleTest() {} void TearDown() OVERRIDE {} void SetUp() OVERRIDE { - acm_.reset(AudioCodingModule::Create(id_, clock_)); - - RegisterCodec(); - rtp_utility_->Populate(&rtp_header_); input_frame_.sample_rate_hz_ = kSampleRateHz; @@ -141,17 +138,32 @@ class AudioCodingModuleTest : public ::testing::Test { memset(input_frame_.data_, 0, input_frame_.samples_per_channel_ * sizeof(input_frame_.data_[0])); + } - ASSERT_EQ(0, acm_->RegisterTransportCallback(&packet_cb_)); + void CreateAcm() { + acm_.reset(AudioCoding::Create(config_)); + ASSERT_TRUE(acm_.get() != NULL); + RegisterCodec(); } virtual void RegisterCodec() { - AudioCodingModule::Codec("L16", &codec_, kSampleRateHz, 1); - codec_.pltype = kPayloadType; - // Register L16 codec in ACM. - ASSERT_EQ(0, acm_->RegisterReceiveCodec(codec_)); - ASSERT_EQ(0, acm_->RegisterSendCodec(codec_)); + int codec_type = acm2::ACMCodecDB::kNone; + switch (kSampleRateHz) { + case 8000: + codec_type = acm2::ACMCodecDB::kPCM16B; + break; + case 16000: + codec_type = acm2::ACMCodecDB::kPCM16Bwb; + break; + case 32000: + codec_type = acm2::ACMCodecDB::kPCM16Bswb32kHz; + break; + default: + FATAL() << "Sample rate not supported in this test."; + } + ASSERT_TRUE(acm_->RegisterSendCodec(codec_type, kPayloadType)); + ASSERT_TRUE(acm_->RegisterReceiveCodec(codec_type, kPayloadType)); } virtual void InsertPacketAndPullAudio() { @@ -161,41 +173,33 @@ class AudioCodingModuleTest : public ::testing::Test { virtual void InsertPacket() { const uint8_t kPayload[kPayloadSizeBytes] = {0}; - ASSERT_EQ(0, - acm_->IncomingPacket(kPayload, kPayloadSizeBytes, rtp_header_)); + ASSERT_TRUE(acm_->InsertPacket(kPayload, kPayloadSizeBytes, rtp_header_)); rtp_utility_->Forward(&rtp_header_); } virtual void PullAudio() { AudioFrame audio_frame; - ASSERT_EQ(0, acm_->PlayoutData10Ms(-1, &audio_frame)); + ASSERT_TRUE(acm_->Get10MsAudio(&audio_frame)); } virtual void InsertAudio() { - ASSERT_EQ(0, acm_->Add10MsData(input_frame_)); + int encoded_bytes = acm_->Add10MsAudio(input_frame_); + ASSERT_GE(encoded_bytes, 0); input_frame_.timestamp_ += kNumSamples10ms; } - virtual void Encode() { - int32_t encoded_bytes = acm_->Process(); - // Expect to get one packet with two bytes per sample, or no packet at all, - // depending on how many 10 ms blocks go into |codec_.pacsize|. - EXPECT_TRUE(encoded_bytes == 2 * codec_.pacsize || encoded_bytes == 0); - } - - const int id_; + AudioCoding::Config config_; scoped_ptr<RtpUtility> rtp_utility_; - scoped_ptr<AudioCodingModule> acm_; + scoped_ptr<AudioCoding> acm_; PacketizationCallbackStub packet_cb_; WebRtcRTPHeader rtp_header_; AudioFrame input_frame_; - CodecInst codec_; - Clock* clock_; }; // Check if the statistics are initialized correctly. Before any call to ACM // all fields have to be zero. TEST_F(AudioCodingModuleTest, DISABLED_ON_ANDROID(InitializedToZero)) { + CreateAcm(); AudioDecodingCallStats stats; acm_->GetDecodingCallStatistics(&stats); EXPECT_EQ(0, stats.calls_to_neteq); @@ -209,10 +213,10 @@ TEST_F(AudioCodingModuleTest, DISABLED_ON_ANDROID(InitializedToZero)) { // Apply an initial playout delay. Calls to AudioCodingModule::PlayoutData10ms() // should result in generating silence, check the associated field. TEST_F(AudioCodingModuleTest, DISABLED_ON_ANDROID(SilenceGeneratorCalled)) { - AudioDecodingCallStats stats; const int kInitialDelay = 100; - - acm_->SetInitialPlayoutDelay(kInitialDelay); + config_.initial_playout_delay_ms = kInitialDelay; + CreateAcm(); + AudioDecodingCallStats stats; int num_calls = 0; for (int time_ms = 0; time_ms < kInitialDelay; @@ -232,6 +236,7 @@ TEST_F(AudioCodingModuleTest, DISABLED_ON_ANDROID(SilenceGeneratorCalled)) { // simulate packet loss and check if PLC and PLC-to-CNG statistics are // correctly updated. TEST_F(AudioCodingModuleTest, DISABLED_ON_ANDROID(NetEqCalls)) { + CreateAcm(); AudioDecodingCallStats stats; const int kNumNormalCalls = 10; @@ -263,21 +268,16 @@ TEST_F(AudioCodingModuleTest, DISABLED_ON_ANDROID(NetEqCalls)) { } TEST_F(AudioCodingModuleTest, VerifyOutputFrame) { + CreateAcm(); AudioFrame audio_frame; const int kSampleRateHz = 32000; - EXPECT_EQ(0, acm_->PlayoutData10Ms(kSampleRateHz, &audio_frame)); - EXPECT_EQ(id_, audio_frame.id_); + EXPECT_TRUE(acm_->Get10MsAudio(&audio_frame)); EXPECT_EQ(0u, audio_frame.timestamp_); EXPECT_GT(audio_frame.num_channels_, 0); EXPECT_EQ(kSampleRateHz / 100, audio_frame.samples_per_channel_); EXPECT_EQ(kSampleRateHz, audio_frame.sample_rate_hz_); } -TEST_F(AudioCodingModuleTest, FailOnZeroDesiredFrequency) { - AudioFrame audio_frame; - EXPECT_EQ(-1, acm_->PlayoutData10Ms(0, &audio_frame)); -} - // A multi-threaded test for ACM. This base class is using the PCM16b 16 kHz // codec, while the derive class AcmIsacMtTest is using iSAC. class AudioCodingModuleMtTest : public AudioCodingModuleTest { @@ -306,11 +306,12 @@ class AudioCodingModuleMtTest : public AudioCodingModuleTest { crit_sect_(CriticalSectionWrapper::CreateCriticalSection()), next_insert_packet_time_ms_(0), fake_clock_(new SimulatedClock(0)) { - clock_ = fake_clock_.get(); + config_.clock = fake_clock_.get(); } virtual void SetUp() OVERRIDE { AudioCodingModuleTest::SetUp(); + CreateAcm(); StartThreads(); } @@ -357,7 +358,6 @@ class AudioCodingModuleMtTest : public AudioCodingModuleTest { } ++send_count_; InsertAudio(); - Encode(); if (TestDone()) { test_complete_->Set(); } @@ -373,7 +373,7 @@ class AudioCodingModuleMtTest : public AudioCodingModuleTest { SleepMs(1); { CriticalSectionScoped lock(crit_sect_.get()); - if (clock_->TimeInMilliseconds() < next_insert_packet_time_ms_) { + if (fake_clock_->TimeInMilliseconds() < next_insert_packet_time_ms_) { return true; } next_insert_packet_time_ms_ += 10; @@ -394,7 +394,7 @@ class AudioCodingModuleMtTest : public AudioCodingModuleTest { { CriticalSectionScoped lock(crit_sect_.get()); // Don't let the insert thread fall behind. - if (next_insert_packet_time_ms_ < clock_->TimeInMilliseconds()) { + if (next_insert_packet_time_ms_ < fake_clock_->TimeInMilliseconds()) { return true; } ++pull_audio_count_; @@ -439,6 +439,7 @@ class AcmIsacMtTest : public AudioCodingModuleMtTest { virtual void SetUp() OVERRIDE { AudioCodingModuleTest::SetUp(); + CreateAcm(); // Set up input audio source to read from specified file, loop after 5 // seconds, and deliver blocks of 10 ms. @@ -450,7 +451,6 @@ class AcmIsacMtTest : public AudioCodingModuleMtTest { int loop_counter = 0; while (packet_cb_.last_payload_len_bytes() == 0) { InsertAudio(); - Encode(); ASSERT_LT(loop_counter++, 10); } // Set |last_packet_number_| to one less that |num_calls| so that the packet @@ -462,13 +462,12 @@ class AcmIsacMtTest : public AudioCodingModuleMtTest { virtual void RegisterCodec() OVERRIDE { COMPILE_ASSERT(kSampleRateHz == 16000, test_designed_for_isac_16khz); - AudioCodingModule::Codec("ISAC", &codec_, kSampleRateHz, 1); - codec_.pltype = kPayloadType; // Register iSAC codec in ACM, effectively unregistering the PCM16B codec // registered in AudioCodingModuleTest::SetUp(); - ASSERT_EQ(0, acm_->RegisterReceiveCodec(codec_)); - ASSERT_EQ(0, acm_->RegisterSendCodec(codec_)); + ASSERT_TRUE(acm_->RegisterSendCodec(acm2::ACMCodecDB::kISAC, kPayloadType)); + ASSERT_TRUE( + acm_->RegisterReceiveCodec(acm2::ACMCodecDB::kISAC, kPayloadType)); } virtual void InsertPacket() OVERRIDE { @@ -484,10 +483,8 @@ class AcmIsacMtTest : public AudioCodingModuleMtTest { last_packet_number_ = num_calls; } ASSERT_GT(last_payload_vec_.size(), 0u); - ASSERT_EQ( - 0, - acm_->IncomingPacket( - &last_payload_vec_[0], last_payload_vec_.size(), rtp_header_)); + ASSERT_TRUE(acm_->InsertPacket( + &last_payload_vec_[0], last_payload_vec_.size(), rtp_header_)); } virtual void InsertAudio() OVERRIDE { @@ -495,8 +492,6 @@ class AcmIsacMtTest : public AudioCodingModuleMtTest { AudioCodingModuleTest::InsertAudio(); } - virtual void Encode() OVERRIDE { ASSERT_GE(acm_->Process(), 0); } - // This method is the same as AudioCodingModuleMtTest::TestDone(), but here // it is using the constants defined in this class (i.e., shorter test run). virtual bool TestDone() OVERRIDE { @@ -634,19 +629,15 @@ class AcmSenderBitExactness : public ::testing::Test, // Registers a send codec in the test::AcmSendTest object. Returns true on // success, false on failure. - bool RegisterSendCodec(const char* payload_name, - int sampling_freq_hz, + bool RegisterSendCodec(int codec_type, int channels, int payload_type, int frame_size_samples, int frame_size_rtp_timestamps) { payload_type_ = payload_type; frame_size_rtp_timestamps_ = frame_size_rtp_timestamps; - return send_test_->RegisterCodec(payload_name, - sampling_freq_hz, - channels, - payload_type, - frame_size_samples); + return send_test_->RegisterCodec( + codec_type, channels, payload_type, frame_size_samples); } // Runs the test. SetUpSender() and RegisterSendCodec() must have been called @@ -728,15 +719,13 @@ class AcmSenderBitExactness : public ::testing::Test, payload_checksum_.Update(packet->payload(), packet->payload_length_bytes()); } - void SetUpTest(const char* codec_name, - int codec_sample_rate_hz, + void SetUpTest(int codec_type, int channels, int payload_type, int codec_frame_size_samples, int codec_frame_size_rtp_timestamps) { ASSERT_TRUE(SetUpSender()); - ASSERT_TRUE(RegisterSendCodec(codec_name, - codec_sample_rate_hz, + ASSERT_TRUE(RegisterSendCodec(codec_type, channels, payload_type, codec_frame_size_samples, @@ -754,7 +743,7 @@ class AcmSenderBitExactness : public ::testing::Test, }; TEST_F(AcmSenderBitExactness, IsacWb30ms) { - ASSERT_NO_FATAL_FAILURE(SetUpTest("ISAC", 16000, 1, 103, 480, 480)); + ASSERT_NO_FATAL_FAILURE(SetUpTest(acm2::ACMCodecDB::kISAC, 1, 103, 480, 480)); Run(AcmReceiverBitExactness::PlatformChecksum( "c7e5bdadfa2871df95639fcc297cf23d", "0499ca260390769b3172136faad925b9", @@ -768,7 +757,7 @@ TEST_F(AcmSenderBitExactness, IsacWb30ms) { } TEST_F(AcmSenderBitExactness, IsacWb60ms) { - ASSERT_NO_FATAL_FAILURE(SetUpTest("ISAC", 16000, 1, 103, 960, 960)); + ASSERT_NO_FATAL_FAILURE(SetUpTest(acm2::ACMCodecDB::kISAC, 1, 103, 960, 960)); Run(AcmReceiverBitExactness::PlatformChecksum( "14d63c5f08127d280e722e3191b73bdd", "8da003e16c5371af2dc2be79a50f9076", @@ -782,7 +771,8 @@ TEST_F(AcmSenderBitExactness, IsacWb60ms) { } TEST_F(AcmSenderBitExactness, DISABLED_ON_ANDROID(IsacSwb30ms)) { - ASSERT_NO_FATAL_FAILURE(SetUpTest("ISAC", 32000, 1, 104, 960, 960)); + ASSERT_NO_FATAL_FAILURE( + SetUpTest(acm2::ACMCodecDB::kISACSWB, 1, 104, 960, 960)); Run(AcmReceiverBitExactness::PlatformChecksum( "98d960600eb4ddb3fcbe11f5057ddfd7", "", @@ -796,7 +786,7 @@ TEST_F(AcmSenderBitExactness, DISABLED_ON_ANDROID(IsacSwb30ms)) { } TEST_F(AcmSenderBitExactness, Pcm16_8000khz_10ms) { - ASSERT_NO_FATAL_FAILURE(SetUpTest("L16", 8000, 1, 107, 80, 80)); + ASSERT_NO_FATAL_FAILURE(SetUpTest(acm2::ACMCodecDB::kPCM16B, 1, 107, 80, 80)); Run("de4a98e1406f8b798d99cd0704e862e2", "c1edd36339ce0326cc4550041ad719a0", 100, @@ -804,7 +794,8 @@ TEST_F(AcmSenderBitExactness, Pcm16_8000khz_10ms) { } TEST_F(AcmSenderBitExactness, Pcm16_16000khz_10ms) { - ASSERT_NO_FATAL_FAILURE(SetUpTest("L16", 16000, 1, 108, 160, 160)); + ASSERT_NO_FATAL_FAILURE( + SetUpTest(acm2::ACMCodecDB::kPCM16Bwb, 1, 108, 160, 160)); Run("ae646d7b68384a1269cc080dd4501916", "ad786526383178b08d80d6eee06e9bad", 100, @@ -812,7 +803,8 @@ TEST_F(AcmSenderBitExactness, Pcm16_16000khz_10ms) { } TEST_F(AcmSenderBitExactness, Pcm16_32000khz_10ms) { - ASSERT_NO_FATAL_FAILURE(SetUpTest("L16", 32000, 1, 109, 320, 320)); + ASSERT_NO_FATAL_FAILURE( + SetUpTest(acm2::ACMCodecDB::kPCM16Bswb32kHz, 1, 109, 320, 320)); Run("7fe325e8fbaf755e3c5df0b11a4774fb", "5ef82ea885e922263606c6fdbc49f651", 100, @@ -820,7 +812,8 @@ TEST_F(AcmSenderBitExactness, Pcm16_32000khz_10ms) { } TEST_F(AcmSenderBitExactness, Pcm16_stereo_8000khz_10ms) { - ASSERT_NO_FATAL_FAILURE(SetUpTest("L16", 8000, 2, 111, 80, 80)); + ASSERT_NO_FATAL_FAILURE( + SetUpTest(acm2::ACMCodecDB::kPCM16B_2ch, 2, 111, 80, 80)); Run("fb263b74e7ac3de915474d77e4744ceb", "62ce5adb0d4965d0a52ec98ae7f98974", 100, @@ -828,7 +821,8 @@ TEST_F(AcmSenderBitExactness, Pcm16_stereo_8000khz_10ms) { } TEST_F(AcmSenderBitExactness, Pcm16_stereo_16000khz_10ms) { - ASSERT_NO_FATAL_FAILURE(SetUpTest("L16", 16000, 2, 112, 160, 160)); + ASSERT_NO_FATAL_FAILURE( + SetUpTest(acm2::ACMCodecDB::kPCM16Bwb_2ch, 2, 112, 160, 160)); Run("d09e9239553649d7ac93e19d304281fd", "41ca8edac4b8c71cd54fd9f25ec14870", 100, @@ -836,7 +830,8 @@ TEST_F(AcmSenderBitExactness, Pcm16_stereo_16000khz_10ms) { } TEST_F(AcmSenderBitExactness, Pcm16_stereo_32000khz_10ms) { - ASSERT_NO_FATAL_FAILURE(SetUpTest("L16", 32000, 2, 113, 320, 320)); + ASSERT_NO_FATAL_FAILURE( + SetUpTest(acm2::ACMCodecDB::kPCM16Bswb32kHz_2ch, 2, 113, 320, 320)); Run("5f025d4f390982cc26b3d92fe02e3044", "50e58502fb04421bf5b857dda4c96879", 100, @@ -844,7 +839,7 @@ TEST_F(AcmSenderBitExactness, Pcm16_stereo_32000khz_10ms) { } TEST_F(AcmSenderBitExactness, Pcmu_20ms) { - ASSERT_NO_FATAL_FAILURE(SetUpTest("PCMU", 8000, 1, 0, 160, 160)); + ASSERT_NO_FATAL_FAILURE(SetUpTest(acm2::ACMCodecDB::kPCMU, 1, 0, 160, 160)); Run("81a9d4c0bb72e9becc43aef124c981e9", "8f9b8750bd80fe26b6cbf6659b89f0f9", 50, @@ -852,7 +847,7 @@ TEST_F(AcmSenderBitExactness, Pcmu_20ms) { } TEST_F(AcmSenderBitExactness, Pcma_20ms) { - ASSERT_NO_FATAL_FAILURE(SetUpTest("PCMA", 8000, 1, 8, 160, 160)); + ASSERT_NO_FATAL_FAILURE(SetUpTest(acm2::ACMCodecDB::kPCMA, 1, 8, 160, 160)); Run("39611f798969053925a49dc06d08de29", "6ad745e55aa48981bfc790d0eeef2dd1", 50, @@ -860,7 +855,8 @@ TEST_F(AcmSenderBitExactness, Pcma_20ms) { } TEST_F(AcmSenderBitExactness, Pcmu_stereo_20ms) { - ASSERT_NO_FATAL_FAILURE(SetUpTest("PCMU", 8000, 2, 110, 160, 160)); + ASSERT_NO_FATAL_FAILURE( + SetUpTest(acm2::ACMCodecDB::kPCMU_2ch, 2, 110, 160, 160)); Run("437bec032fdc5cbaa0d5175430af7b18", "60b6f25e8d1e74cb679cfe756dd9bca5", 50, @@ -868,7 +864,8 @@ TEST_F(AcmSenderBitExactness, Pcmu_stereo_20ms) { } TEST_F(AcmSenderBitExactness, Pcma_stereo_20ms) { - ASSERT_NO_FATAL_FAILURE(SetUpTest("PCMA", 8000, 2, 118, 160, 160)); + ASSERT_NO_FATAL_FAILURE( + SetUpTest(acm2::ACMCodecDB::kPCMA_2ch, 2, 118, 160, 160)); Run("a5c6d83c5b7cedbeff734238220a4b0c", "92b282c83efd20e7eeef52ba40842cf7", 50, @@ -876,7 +873,7 @@ TEST_F(AcmSenderBitExactness, Pcma_stereo_20ms) { } TEST_F(AcmSenderBitExactness, DISABLED_ON_ANDROID(Ilbc_30ms)) { - ASSERT_NO_FATAL_FAILURE(SetUpTest("ILBC", 8000, 1, 102, 240, 240)); + ASSERT_NO_FATAL_FAILURE(SetUpTest(acm2::ACMCodecDB::kILBC, 1, 102, 240, 240)); Run(AcmReceiverBitExactness::PlatformChecksum( "7b6ec10910debd9af08011d3ed5249f7", "android_audio", @@ -890,7 +887,7 @@ TEST_F(AcmSenderBitExactness, DISABLED_ON_ANDROID(Ilbc_30ms)) { } TEST_F(AcmSenderBitExactness, DISABLED_ON_ANDROID(G722_20ms)) { - ASSERT_NO_FATAL_FAILURE(SetUpTest("G722", 16000, 1, 9, 320, 160)); + ASSERT_NO_FATAL_FAILURE(SetUpTest(acm2::ACMCodecDB::kG722, 1, 9, 320, 160)); Run(AcmReceiverBitExactness::PlatformChecksum( "7d759436f2533582950d148b5161a36c", "android_audio", @@ -904,7 +901,8 @@ TEST_F(AcmSenderBitExactness, DISABLED_ON_ANDROID(G722_20ms)) { } TEST_F(AcmSenderBitExactness, DISABLED_ON_ANDROID(G722_stereo_20ms)) { - ASSERT_NO_FATAL_FAILURE(SetUpTest("G722", 16000, 2, 119, 320, 160)); + ASSERT_NO_FATAL_FAILURE( + SetUpTest(acm2::ACMCodecDB::kG722_2ch, 2, 119, 320, 160)); Run(AcmReceiverBitExactness::PlatformChecksum( "7190ee718ab3d80eca181e5f7140c210", "android_audio", @@ -918,7 +916,7 @@ TEST_F(AcmSenderBitExactness, DISABLED_ON_ANDROID(G722_stereo_20ms)) { } TEST_F(AcmSenderBitExactness, Opus_stereo_20ms) { - ASSERT_NO_FATAL_FAILURE(SetUpTest("opus", 48000, 2, 120, 960, 960)); + ASSERT_NO_FATAL_FAILURE(SetUpTest(acm2::ACMCodecDB::kOpus, 2, 120, 960, 960)); Run(AcmReceiverBitExactness::PlatformChecksum( "855041f2490b887302bce9d544731849", "1e1a0fce893fef2d66886a7f09e2ebce", diff --git a/modules/audio_coding/main/acm2/audio_coding_module_unittest_oldapi.cc b/modules/audio_coding/main/acm2/audio_coding_module_unittest_oldapi.cc new file mode 100644 index 00000000..0e9d2736 --- /dev/null +++ b/modules/audio_coding/main/acm2/audio_coding_module_unittest_oldapi.cc @@ -0,0 +1,938 @@ +/* + * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#include <string.h> +#include <vector> + +#include "testing/gtest/include/gtest/gtest.h" +#include "webrtc/base/md5digest.h" +#include "webrtc/base/thread_annotations.h" +#include "webrtc/modules/audio_coding/main/acm2/acm_receive_test_oldapi.h" +#include "webrtc/modules/audio_coding/main/acm2/acm_send_test_oldapi.h" +#include "webrtc/modules/audio_coding/main/interface/audio_coding_module.h" +#include "webrtc/modules/audio_coding/main/interface/audio_coding_module_typedefs.h" +#include "webrtc/modules/audio_coding/neteq/tools/audio_checksum.h" +#include "webrtc/modules/audio_coding/neteq/tools/audio_loop.h" +#include "webrtc/modules/audio_coding/neteq/tools/input_audio_file.h" +#include "webrtc/modules/audio_coding/neteq/tools/output_audio_file.h" +#include "webrtc/modules/audio_coding/neteq/tools/packet.h" +#include "webrtc/modules/audio_coding/neteq/tools/rtp_file_source.h" +#include "webrtc/modules/interface/module_common_types.h" +#include "webrtc/system_wrappers/interface/clock.h" +#include "webrtc/system_wrappers/interface/compile_assert.h" +#include "webrtc/system_wrappers/interface/critical_section_wrapper.h" +#include "webrtc/system_wrappers/interface/event_wrapper.h" +#include "webrtc/system_wrappers/interface/scoped_ptr.h" +#include "webrtc/system_wrappers/interface/sleep.h" +#include "webrtc/system_wrappers/interface/thread_wrapper.h" +#include "webrtc/test/testsupport/fileutils.h" +#include "webrtc/test/testsupport/gtest_disable.h" + +namespace webrtc { + +const int kSampleRateHz = 16000; +const int kNumSamples10ms = kSampleRateHz / 100; +const int kFrameSizeMs = 10; // Multiple of 10. +const int kFrameSizeSamples = kFrameSizeMs / 10 * kNumSamples10ms; +const int kPayloadSizeBytes = kFrameSizeSamples * sizeof(int16_t); +const uint8_t kPayloadType = 111; + +class RtpUtility { + public: + RtpUtility(int samples_per_packet, uint8_t payload_type) + : samples_per_packet_(samples_per_packet), payload_type_(payload_type) {} + + virtual ~RtpUtility() {} + + void Populate(WebRtcRTPHeader* rtp_header) { + rtp_header->header.sequenceNumber = 0xABCD; + rtp_header->header.timestamp = 0xABCDEF01; + rtp_header->header.payloadType = payload_type_; + rtp_header->header.markerBit = false; + rtp_header->header.ssrc = 0x1234; + rtp_header->header.numCSRCs = 0; + rtp_header->frameType = kAudioFrameSpeech; + + rtp_header->header.payload_type_frequency = kSampleRateHz; + rtp_header->type.Audio.channel = 1; + rtp_header->type.Audio.isCNG = false; + } + + void Forward(WebRtcRTPHeader* rtp_header) { + ++rtp_header->header.sequenceNumber; + rtp_header->header.timestamp += samples_per_packet_; + } + + private: + int samples_per_packet_; + uint8_t payload_type_; +}; + +class PacketizationCallbackStub : public AudioPacketizationCallback { + public: + PacketizationCallbackStub() + : num_calls_(0), + crit_sect_(CriticalSectionWrapper::CreateCriticalSection()) {} + + virtual int32_t SendData( + FrameType frame_type, + uint8_t payload_type, + uint32_t timestamp, + const uint8_t* payload_data, + uint16_t payload_len_bytes, + const RTPFragmentationHeader* fragmentation) OVERRIDE { + CriticalSectionScoped lock(crit_sect_.get()); + ++num_calls_; + last_payload_vec_.assign(payload_data, payload_data + payload_len_bytes); + return 0; + } + + int num_calls() const { + CriticalSectionScoped lock(crit_sect_.get()); + return num_calls_; + } + + int last_payload_len_bytes() const { + CriticalSectionScoped lock(crit_sect_.get()); + return last_payload_vec_.size(); + } + + void SwapBuffers(std::vector<uint8_t>* payload) { + CriticalSectionScoped lock(crit_sect_.get()); + last_payload_vec_.swap(*payload); + } + + private: + int num_calls_ GUARDED_BY(crit_sect_); + std::vector<uint8_t> last_payload_vec_ GUARDED_BY(crit_sect_); + const scoped_ptr<CriticalSectionWrapper> crit_sect_; +}; + +class AudioCodingModuleTestOldApi : public ::testing::Test { + protected: + AudioCodingModuleTestOldApi() + : id_(1), + rtp_utility_(new RtpUtility(kFrameSizeSamples, kPayloadType)), + clock_(Clock::GetRealTimeClock()) {} + + ~AudioCodingModuleTestOldApi() {} + + void TearDown() {} + + void SetUp() { + acm_.reset(AudioCodingModule::Create(id_, clock_)); + + RegisterCodec(); + + rtp_utility_->Populate(&rtp_header_); + + input_frame_.sample_rate_hz_ = kSampleRateHz; + input_frame_.num_channels_ = 1; + input_frame_.samples_per_channel_ = kSampleRateHz * 10 / 1000; // 10 ms. + COMPILE_ASSERT(kSampleRateHz * 10 / 1000 <= AudioFrame::kMaxDataSizeSamples, + audio_frame_too_small); + memset(input_frame_.data_, + 0, + input_frame_.samples_per_channel_ * sizeof(input_frame_.data_[0])); + + ASSERT_EQ(0, acm_->RegisterTransportCallback(&packet_cb_)); + } + + virtual void RegisterCodec() { + AudioCodingModule::Codec("L16", &codec_, kSampleRateHz, 1); + codec_.pltype = kPayloadType; + + // Register L16 codec in ACM. + ASSERT_EQ(0, acm_->RegisterReceiveCodec(codec_)); + ASSERT_EQ(0, acm_->RegisterSendCodec(codec_)); + } + + virtual void InsertPacketAndPullAudio() { + InsertPacket(); + PullAudio(); + } + + virtual void InsertPacket() { + const uint8_t kPayload[kPayloadSizeBytes] = {0}; + ASSERT_EQ(0, + acm_->IncomingPacket(kPayload, kPayloadSizeBytes, rtp_header_)); + rtp_utility_->Forward(&rtp_header_); + } + + virtual void PullAudio() { + AudioFrame audio_frame; + ASSERT_EQ(0, acm_->PlayoutData10Ms(-1, &audio_frame)); + } + + virtual void InsertAudio() { + ASSERT_EQ(0, acm_->Add10MsData(input_frame_)); + input_frame_.timestamp_ += kNumSamples10ms; + } + + virtual void Encode() { + int32_t encoded_bytes = acm_->Process(); + // Expect to get one packet with two bytes per sample, or no packet at all, + // depending on how many 10 ms blocks go into |codec_.pacsize|. + EXPECT_TRUE(encoded_bytes == 2 * codec_.pacsize || encoded_bytes == 0); + } + + const int id_; + scoped_ptr<RtpUtility> rtp_utility_; + scoped_ptr<AudioCodingModule> acm_; + PacketizationCallbackStub packet_cb_; + WebRtcRTPHeader rtp_header_; + AudioFrame input_frame_; + CodecInst codec_; + Clock* clock_; +}; + +// Check if the statistics are initialized correctly. Before any call to ACM +// all fields have to be zero. +TEST_F(AudioCodingModuleTestOldApi, DISABLED_ON_ANDROID(InitializedToZero)) { + AudioDecodingCallStats stats; + acm_->GetDecodingCallStatistics(&stats); + EXPECT_EQ(0, stats.calls_to_neteq); + EXPECT_EQ(0, stats.calls_to_silence_generator); + EXPECT_EQ(0, stats.decoded_normal); + EXPECT_EQ(0, stats.decoded_cng); + EXPECT_EQ(0, stats.decoded_plc); + EXPECT_EQ(0, stats.decoded_plc_cng); +} + +// Apply an initial playout delay. Calls to AudioCodingModule::PlayoutData10ms() +// should result in generating silence, check the associated field. +TEST_F(AudioCodingModuleTestOldApi, + DISABLED_ON_ANDROID(SilenceGeneratorCalled)) { + AudioDecodingCallStats stats; + const int kInitialDelay = 100; + + acm_->SetInitialPlayoutDelay(kInitialDelay); + + int num_calls = 0; + for (int time_ms = 0; time_ms < kInitialDelay; + time_ms += kFrameSizeMs, ++num_calls) { + InsertPacketAndPullAudio(); + } + acm_->GetDecodingCallStatistics(&stats); + EXPECT_EQ(0, stats.calls_to_neteq); + EXPECT_EQ(num_calls, stats.calls_to_silence_generator); + EXPECT_EQ(0, stats.decoded_normal); + EXPECT_EQ(0, stats.decoded_cng); + EXPECT_EQ(0, stats.decoded_plc); + EXPECT_EQ(0, stats.decoded_plc_cng); +} + +// Insert some packets and pull audio. Check statistics are valid. Then, +// simulate packet loss and check if PLC and PLC-to-CNG statistics are +// correctly updated. +TEST_F(AudioCodingModuleTestOldApi, DISABLED_ON_ANDROID(NetEqCalls)) { + AudioDecodingCallStats stats; + const int kNumNormalCalls = 10; + + for (int num_calls = 0; num_calls < kNumNormalCalls; ++num_calls) { + InsertPacketAndPullAudio(); + } + acm_->GetDecodingCallStatistics(&stats); + EXPECT_EQ(kNumNormalCalls, stats.calls_to_neteq); + EXPECT_EQ(0, stats.calls_to_silence_generator); + EXPECT_EQ(kNumNormalCalls, stats.decoded_normal); + EXPECT_EQ(0, stats.decoded_cng); + EXPECT_EQ(0, stats.decoded_plc); + EXPECT_EQ(0, stats.decoded_plc_cng); + + const int kNumPlc = 3; + const int kNumPlcCng = 5; + + // Simulate packet-loss. NetEq first performs PLC then PLC fades to CNG. + for (int n = 0; n < kNumPlc + kNumPlcCng; ++n) { + PullAudio(); + } + acm_->GetDecodingCallStatistics(&stats); + EXPECT_EQ(kNumNormalCalls + kNumPlc + kNumPlcCng, stats.calls_to_neteq); + EXPECT_EQ(0, stats.calls_to_silence_generator); + EXPECT_EQ(kNumNormalCalls, stats.decoded_normal); + EXPECT_EQ(0, stats.decoded_cng); + EXPECT_EQ(kNumPlc, stats.decoded_plc); + EXPECT_EQ(kNumPlcCng, stats.decoded_plc_cng); +} + +TEST_F(AudioCodingModuleTestOldApi, VerifyOutputFrame) { + AudioFrame audio_frame; + const int kSampleRateHz = 32000; + EXPECT_EQ(0, acm_->PlayoutData10Ms(kSampleRateHz, &audio_frame)); + EXPECT_EQ(id_, audio_frame.id_); + EXPECT_EQ(0u, audio_frame.timestamp_); + EXPECT_GT(audio_frame.num_channels_, 0); + EXPECT_EQ(kSampleRateHz / 100, audio_frame.samples_per_channel_); + EXPECT_EQ(kSampleRateHz, audio_frame.sample_rate_hz_); +} + +TEST_F(AudioCodingModuleTestOldApi, FailOnZeroDesiredFrequency) { + AudioFrame audio_frame; + EXPECT_EQ(-1, acm_->PlayoutData10Ms(0, &audio_frame)); +} + +// A multi-threaded test for ACM. This base class is using the PCM16b 16 kHz +// codec, while the derive class AcmIsacMtTest is using iSAC. +class AudioCodingModuleMtTestOldApi : public AudioCodingModuleTestOldApi { + protected: + static const int kNumPackets = 500; + static const int kNumPullCalls = 500; + + AudioCodingModuleMtTestOldApi() + : AudioCodingModuleTestOldApi(), + send_thread_(ThreadWrapper::CreateThread(CbSendThread, + this, + kRealtimePriority, + "send")), + insert_packet_thread_(ThreadWrapper::CreateThread(CbInsertPacketThread, + this, + kRealtimePriority, + "insert_packet")), + pull_audio_thread_(ThreadWrapper::CreateThread(CbPullAudioThread, + this, + kRealtimePriority, + "pull_audio")), + test_complete_(EventWrapper::Create()), + send_count_(0), + insert_packet_count_(0), + pull_audio_count_(0), + crit_sect_(CriticalSectionWrapper::CreateCriticalSection()), + next_insert_packet_time_ms_(0), + fake_clock_(new SimulatedClock(0)) { + clock_ = fake_clock_.get(); + } + + void SetUp() { + AudioCodingModuleTestOldApi::SetUp(); + StartThreads(); + } + + void StartThreads() { + unsigned int thread_id = 0; + ASSERT_TRUE(send_thread_->Start(thread_id)); + ASSERT_TRUE(insert_packet_thread_->Start(thread_id)); + ASSERT_TRUE(pull_audio_thread_->Start(thread_id)); + } + + void TearDown() { + AudioCodingModuleTestOldApi::TearDown(); + pull_audio_thread_->Stop(); + send_thread_->Stop(); + insert_packet_thread_->Stop(); + } + + EventTypeWrapper RunTest() { + return test_complete_->Wait(10 * 60 * 1000); // 10 minutes' timeout. + } + + virtual bool TestDone() { + if (packet_cb_.num_calls() > kNumPackets) { + CriticalSectionScoped lock(crit_sect_.get()); + if (pull_audio_count_ > kNumPullCalls) { + // Both conditions for completion are met. End the test. + return true; + } + } + return false; + } + + static bool CbSendThread(void* context) { + return reinterpret_cast<AudioCodingModuleMtTestOldApi*>(context) + ->CbSendImpl(); + } + + // The send thread doesn't have to care about the current simulated time, + // since only the AcmReceiver is using the clock. + bool CbSendImpl() { + SleepMs(1); + if (HasFatalFailure()) { + // End the test early if a fatal failure (ASSERT_*) has occurred. + test_complete_->Set(); + } + ++send_count_; + InsertAudio(); + Encode(); + if (TestDone()) { + test_complete_->Set(); + } + return true; + } + + static bool CbInsertPacketThread(void* context) { + return reinterpret_cast<AudioCodingModuleMtTestOldApi*>(context) + ->CbInsertPacketImpl(); + } + + bool CbInsertPacketImpl() { + SleepMs(1); + { + CriticalSectionScoped lock(crit_sect_.get()); + if (clock_->TimeInMilliseconds() < next_insert_packet_time_ms_) { + return true; + } + next_insert_packet_time_ms_ += 10; + } + // Now we're not holding the crit sect when calling ACM. + ++insert_packet_count_; + InsertPacket(); + return true; + } + + static bool CbPullAudioThread(void* context) { + return reinterpret_cast<AudioCodingModuleMtTestOldApi*>(context) + ->CbPullAudioImpl(); + } + + bool CbPullAudioImpl() { + SleepMs(1); + { + CriticalSectionScoped lock(crit_sect_.get()); + // Don't let the insert thread fall behind. + if (next_insert_packet_time_ms_ < clock_->TimeInMilliseconds()) { + return true; + } + ++pull_audio_count_; + } + // Now we're not holding the crit sect when calling ACM. + PullAudio(); + fake_clock_->AdvanceTimeMilliseconds(10); + return true; + } + + scoped_ptr<ThreadWrapper> send_thread_; + scoped_ptr<ThreadWrapper> insert_packet_thread_; + scoped_ptr<ThreadWrapper> pull_audio_thread_; + const scoped_ptr<EventWrapper> test_complete_; + int send_count_; + int insert_packet_count_; + int pull_audio_count_ GUARDED_BY(crit_sect_); + const scoped_ptr<CriticalSectionWrapper> crit_sect_; + int64_t next_insert_packet_time_ms_ GUARDED_BY(crit_sect_); + scoped_ptr<SimulatedClock> fake_clock_; +}; + +TEST_F(AudioCodingModuleMtTestOldApi, DoTest) { + EXPECT_EQ(kEventSignaled, RunTest()); +} + +// This is a multi-threaded ACM test using iSAC. The test encodes audio +// from a PCM file. The most recent encoded frame is used as input to the +// receiving part. Depending on timing, it may happen that the same RTP packet +// is inserted into the receiver multiple times, but this is a valid use-case, +// and simplifies the test code a lot. +class AcmIsacMtTestOldApi : public AudioCodingModuleMtTestOldApi { + protected: + static const int kNumPackets = 500; + static const int kNumPullCalls = 500; + + AcmIsacMtTestOldApi() + : AudioCodingModuleMtTestOldApi(), last_packet_number_(0) {} + + ~AcmIsacMtTestOldApi() {} + + void SetUp() { + AudioCodingModuleTestOldApi::SetUp(); + + // Set up input audio source to read from specified file, loop after 5 + // seconds, and deliver blocks of 10 ms. + const std::string input_file_name = + webrtc::test::ResourcePath("audio_coding/speech_mono_16kHz", "pcm"); + audio_loop_.Init(input_file_name, 5 * kSampleRateHz, kNumSamples10ms); + + // Generate one packet to have something to insert. + int loop_counter = 0; + while (packet_cb_.last_payload_len_bytes() == 0) { + InsertAudio(); + Encode(); + ASSERT_LT(loop_counter++, 10); + } + // Set |last_packet_number_| to one less that |num_calls| so that the packet + // will be fetched in the next InsertPacket() call. + last_packet_number_ = packet_cb_.num_calls() - 1; + + StartThreads(); + } + + virtual void RegisterCodec() { + COMPILE_ASSERT(kSampleRateHz == 16000, test_designed_for_isac_16khz); + AudioCodingModule::Codec("ISAC", &codec_, kSampleRateHz, 1); + codec_.pltype = kPayloadType; + + // Register iSAC codec in ACM, effectively unregistering the PCM16B codec + // registered in AudioCodingModuleTestOldApi::SetUp(); + ASSERT_EQ(0, acm_->RegisterReceiveCodec(codec_)); + ASSERT_EQ(0, acm_->RegisterSendCodec(codec_)); + } + + void InsertPacket() { + int num_calls = packet_cb_.num_calls(); // Store locally for thread safety. + if (num_calls > last_packet_number_) { + // Get the new payload out from the callback handler. + // Note that since we swap buffers here instead of directly inserting + // a pointer to the data in |packet_cb_|, we avoid locking the callback + // for the duration of the IncomingPacket() call. + packet_cb_.SwapBuffers(&last_payload_vec_); + ASSERT_GT(last_payload_vec_.size(), 0u); + rtp_utility_->Forward(&rtp_header_); + last_packet_number_ = num_calls; + } + ASSERT_GT(last_payload_vec_.size(), 0u); + ASSERT_EQ( + 0, + acm_->IncomingPacket( + &last_payload_vec_[0], last_payload_vec_.size(), rtp_header_)); + } + + void InsertAudio() { + memcpy(input_frame_.data_, audio_loop_.GetNextBlock(), kNumSamples10ms); + AudioCodingModuleTestOldApi::InsertAudio(); + } + + void Encode() { ASSERT_GE(acm_->Process(), 0); } + + // This method is the same as AudioCodingModuleMtTestOldApi::TestDone(), but + // here it is using the constants defined in this class (i.e., shorter test + // run). + virtual bool TestDone() { + if (packet_cb_.num_calls() > kNumPackets) { + CriticalSectionScoped lock(crit_sect_.get()); + if (pull_audio_count_ > kNumPullCalls) { + // Both conditions for completion are met. End the test. + return true; + } + } + return false; + } + + int last_packet_number_; + std::vector<uint8_t> last_payload_vec_; + test::AudioLoop audio_loop_; +}; + +TEST_F(AcmIsacMtTestOldApi, DoTest) { + EXPECT_EQ(kEventSignaled, RunTest()); +} + +class AcmReceiverBitExactnessOldApi : public ::testing::Test { + public: + static std::string PlatformChecksum(std::string win64, + std::string android, + std::string others) { +#if defined(_WIN32) && defined(WEBRTC_ARCH_64_BITS) + return win64; +#elif defined(WEBRTC_ANDROID) + return android; +#else + return others; +#endif + } + + protected: + void Run(int output_freq_hz, const std::string& checksum_ref) { + const std::string input_file_name = + webrtc::test::ResourcePath("audio_coding/neteq_universal_new", "rtp"); + scoped_ptr<test::RtpFileSource> packet_source( + test::RtpFileSource::Create(input_file_name)); +#ifdef WEBRTC_ANDROID + // Filter out iLBC and iSAC-swb since they are not supported on Android. + packet_source->FilterOutPayloadType(102); // iLBC. + packet_source->FilterOutPayloadType(104); // iSAC-swb. +#endif + + test::AudioChecksum checksum; + const std::string output_file_name = + webrtc::test::OutputPath() + + ::testing::UnitTest::GetInstance() + ->current_test_info() + ->test_case_name() + + "_" + ::testing::UnitTest::GetInstance()->current_test_info()->name() + + "_output.pcm"; + test::OutputAudioFile output_file(output_file_name); + test::AudioSinkFork output(&checksum, &output_file); + + test::AcmReceiveTestOldApi test( + packet_source.get(), + &output, + output_freq_hz, + test::AcmReceiveTestOldApi::kArbitraryChannels); + ASSERT_NO_FATAL_FAILURE(test.RegisterNetEqTestCodecs()); + test.Run(); + + std::string checksum_string = checksum.Finish(); + EXPECT_EQ(checksum_ref, checksum_string); + } +}; + +TEST_F(AcmReceiverBitExactnessOldApi, 8kHzOutput) { + Run(8000, + PlatformChecksum("bd6f8d9602cd82444ea2539e674df747", + "6ac89c7145072c26bfeba602cd661afb", + "8a8440f5511eb729221b9aac25cda3a0")); +} + +TEST_F(AcmReceiverBitExactnessOldApi, 16kHzOutput) { + Run(16000, + PlatformChecksum("a39bc6ee0c4eb15f3ad2f43cebcc571d", + "3e888eb04f57db2c6ef952fe64f17fe6", + "7be583092c5adbcb0f6cd66eca20ea63")); +} + +TEST_F(AcmReceiverBitExactnessOldApi, 32kHzOutput) { + Run(32000, + PlatformChecksum("80964572aaa2dc92f9e34896dd3802b3", + "aeca37e963310f5b6552b7edea23c2f1", + "3a84188abe9fca25fedd6034760f3e22")); +} + +TEST_F(AcmReceiverBitExactnessOldApi, 48kHzOutput) { + Run(48000, + PlatformChecksum("8aacde91f390e0d5a9c2ed571a25fd37", + "76b9e99e0a3998aa28355e7a2bd836f7", + "89b4b19bdb4de40f1d88302ef8cb9f9b")); +} + +// This test verifies bit exactness for the send-side of ACM. The test setup is +// a chain of three different test classes: +// +// test::AcmSendTest -> AcmSenderBitExactness -> test::AcmReceiveTest +// +// The receiver side is driving the test by requesting new packets from +// AcmSenderBitExactness::NextPacket(). This method, in turn, asks for the +// packet from test::AcmSendTest::NextPacket, which inserts audio from the +// input file until one packet is produced. (The input file loops indefinitely.) +// Before passing the packet to the receiver, this test class verifies the +// packet header and updates a payload checksum with the new payload. The +// decoded output from the receiver is also verified with a (separate) checksum. +class AcmSenderBitExactnessOldApi : public ::testing::Test, + public test::PacketSource { + protected: + static const int kTestDurationMs = 1000; + + AcmSenderBitExactnessOldApi() + : frame_size_rtp_timestamps_(0), + packet_count_(0), + payload_type_(0), + last_sequence_number_(0), + last_timestamp_(0) {} + + // Sets up the test::AcmSendTest object. Returns true on success, otherwise + // false. + bool SetUpSender() { + const std::string input_file_name = + webrtc::test::ResourcePath("audio_coding/testfile32kHz", "pcm"); + // Note that |audio_source_| will loop forever. The test duration is set + // explicitly by |kTestDurationMs|. + audio_source_.reset(new test::InputAudioFile(input_file_name)); + static const int kSourceRateHz = 32000; + send_test_.reset(new test::AcmSendTestOldApi( + audio_source_.get(), kSourceRateHz, kTestDurationMs)); + return send_test_.get() != NULL; + } + + // Registers a send codec in the test::AcmSendTest object. Returns true on + // success, false on failure. + bool RegisterSendCodec(const char* payload_name, + int sampling_freq_hz, + int channels, + int payload_type, + int frame_size_samples, + int frame_size_rtp_timestamps) { + payload_type_ = payload_type; + frame_size_rtp_timestamps_ = frame_size_rtp_timestamps; + return send_test_->RegisterCodec(payload_name, + sampling_freq_hz, + channels, + payload_type, + frame_size_samples); + } + + // Runs the test. SetUpSender() and RegisterSendCodec() must have been called + // before calling this method. + void Run(const std::string& audio_checksum_ref, + const std::string& payload_checksum_ref, + int expected_packets, + test::AcmReceiveTestOldApi::NumOutputChannels expected_channels) { + // Set up the receiver used to decode the packets and verify the decoded + // output. + test::AudioChecksum audio_checksum; + const std::string output_file_name = + webrtc::test::OutputPath() + + ::testing::UnitTest::GetInstance() + ->current_test_info() + ->test_case_name() + + "_" + ::testing::UnitTest::GetInstance()->current_test_info()->name() + + "_output.pcm"; + test::OutputAudioFile output_file(output_file_name); + // Have the output audio sent both to file and to the checksum calculator. + test::AudioSinkFork output(&audio_checksum, &output_file); + const int kOutputFreqHz = 8000; + test::AcmReceiveTestOldApi receive_test( + this, &output, kOutputFreqHz, expected_channels); + ASSERT_NO_FATAL_FAILURE(receive_test.RegisterDefaultCodecs()); + + // This is where the actual test is executed. + receive_test.Run(); + + // Extract and verify the audio checksum. + std::string checksum_string = audio_checksum.Finish(); + EXPECT_EQ(audio_checksum_ref, checksum_string); + + // Extract and verify the payload checksum. + char checksum_result[rtc::Md5Digest::kSize]; + payload_checksum_.Finish(checksum_result, rtc::Md5Digest::kSize); + checksum_string = rtc::hex_encode(checksum_result, rtc::Md5Digest::kSize); + EXPECT_EQ(payload_checksum_ref, checksum_string); + + // Verify number of packets produced. + EXPECT_EQ(expected_packets, packet_count_); + } + + // Returns a pointer to the next packet. Returns NULL if the source is + // depleted (i.e., the test duration is exceeded), or if an error occurred. + // Inherited from test::PacketSource. + test::Packet* NextPacket() OVERRIDE { + // Get the next packet from AcmSendTest. Ownership of |packet| is + // transferred to this method. + test::Packet* packet = send_test_->NextPacket(); + if (!packet) + return NULL; + + VerifyPacket(packet); + // TODO(henrik.lundin) Save the packet to file as well. + + // Pass it on to the caller. The caller becomes the owner of |packet|. + return packet; + } + + // Verifies the packet. + void VerifyPacket(const test::Packet* packet) { + EXPECT_TRUE(packet->valid_header()); + // (We can check the header fields even if valid_header() is false.) + EXPECT_EQ(payload_type_, packet->header().payloadType); + if (packet_count_ > 0) { + // This is not the first packet. + uint16_t sequence_number_diff = + packet->header().sequenceNumber - last_sequence_number_; + EXPECT_EQ(1, sequence_number_diff); + uint32_t timestamp_diff = packet->header().timestamp - last_timestamp_; + EXPECT_EQ(frame_size_rtp_timestamps_, timestamp_diff); + } + ++packet_count_; + last_sequence_number_ = packet->header().sequenceNumber; + last_timestamp_ = packet->header().timestamp; + // Update the checksum. + payload_checksum_.Update(packet->payload(), packet->payload_length_bytes()); + } + + void SetUpTest(const char* codec_name, + int codec_sample_rate_hz, + int channels, + int payload_type, + int codec_frame_size_samples, + int codec_frame_size_rtp_timestamps) { + ASSERT_TRUE(SetUpSender()); + ASSERT_TRUE(RegisterSendCodec(codec_name, + codec_sample_rate_hz, + channels, + payload_type, + codec_frame_size_samples, + codec_frame_size_rtp_timestamps)); + } + + scoped_ptr<test::AcmSendTestOldApi> send_test_; + scoped_ptr<test::InputAudioFile> audio_source_; + uint32_t frame_size_rtp_timestamps_; + int packet_count_; + uint8_t payload_type_; + uint16_t last_sequence_number_; + uint32_t last_timestamp_; + rtc::Md5Digest payload_checksum_; +}; + +TEST_F(AcmSenderBitExactnessOldApi, IsacWb30ms) { + ASSERT_NO_FATAL_FAILURE(SetUpTest("ISAC", 16000, 1, 103, 480, 480)); + Run(AcmReceiverBitExactnessOldApi::PlatformChecksum( + "c7e5bdadfa2871df95639fcc297cf23d", + "0499ca260390769b3172136faad925b9", + "0b58f9eeee43d5891f5f6c75e77984a3"), + AcmReceiverBitExactnessOldApi::PlatformChecksum( + "d42cb5195463da26c8129bbfe73a22e6", + "83de248aea9c3c2bd680b6952401b4ca", + "3c79f16f34218271f3dca4e2b1dfe1bb"), + 33, + test::AcmReceiveTestOldApi::kMonoOutput); +} + +TEST_F(AcmSenderBitExactnessOldApi, IsacWb60ms) { + ASSERT_NO_FATAL_FAILURE(SetUpTest("ISAC", 16000, 1, 103, 960, 960)); + Run(AcmReceiverBitExactnessOldApi::PlatformChecksum( + "14d63c5f08127d280e722e3191b73bdd", + "8da003e16c5371af2dc2be79a50f9076", + "1ad29139a04782a33daad8c2b9b35875"), + AcmReceiverBitExactnessOldApi::PlatformChecksum( + "ebe04a819d3a9d83a83a17f271e1139a", + "97aeef98553b5a4b5a68f8b716e8eaf0", + "9e0a0ab743ad987b55b8e14802769c56"), + 16, + test::AcmReceiveTestOldApi::kMonoOutput); +} + +TEST_F(AcmSenderBitExactnessOldApi, DISABLED_ON_ANDROID(IsacSwb30ms)) { + ASSERT_NO_FATAL_FAILURE(SetUpTest("ISAC", 32000, 1, 104, 960, 960)); + Run(AcmReceiverBitExactnessOldApi::PlatformChecksum( + "98d960600eb4ddb3fcbe11f5057ddfd7", + "", + "2f6dfe142f735f1d96f6bd86d2526f42"), + AcmReceiverBitExactnessOldApi::PlatformChecksum( + "cc9d2d86a71d6f99f97680a5c27e2762", + "", + "7b214fc3a5e33d68bf30e77969371f31"), + 33, + test::AcmReceiveTestOldApi::kMonoOutput); +} + +TEST_F(AcmSenderBitExactnessOldApi, Pcm16_8000khz_10ms) { + ASSERT_NO_FATAL_FAILURE(SetUpTest("L16", 8000, 1, 107, 80, 80)); + Run("de4a98e1406f8b798d99cd0704e862e2", + "c1edd36339ce0326cc4550041ad719a0", + 100, + test::AcmReceiveTestOldApi::kMonoOutput); +} + +TEST_F(AcmSenderBitExactnessOldApi, Pcm16_16000khz_10ms) { + ASSERT_NO_FATAL_FAILURE(SetUpTest("L16", 16000, 1, 108, 160, 160)); + Run("ae646d7b68384a1269cc080dd4501916", + "ad786526383178b08d80d6eee06e9bad", + 100, + test::AcmReceiveTestOldApi::kMonoOutput); +} + +TEST_F(AcmSenderBitExactnessOldApi, Pcm16_32000khz_10ms) { + ASSERT_NO_FATAL_FAILURE(SetUpTest("L16", 32000, 1, 109, 320, 320)); + Run("7fe325e8fbaf755e3c5df0b11a4774fb", + "5ef82ea885e922263606c6fdbc49f651", + 100, + test::AcmReceiveTestOldApi::kMonoOutput); +} + +TEST_F(AcmSenderBitExactnessOldApi, Pcm16_stereo_8000khz_10ms) { + ASSERT_NO_FATAL_FAILURE(SetUpTest("L16", 8000, 2, 111, 80, 80)); + Run("fb263b74e7ac3de915474d77e4744ceb", + "62ce5adb0d4965d0a52ec98ae7f98974", + 100, + test::AcmReceiveTestOldApi::kStereoOutput); +} + +TEST_F(AcmSenderBitExactnessOldApi, Pcm16_stereo_16000khz_10ms) { + ASSERT_NO_FATAL_FAILURE(SetUpTest("L16", 16000, 2, 112, 160, 160)); + Run("d09e9239553649d7ac93e19d304281fd", + "41ca8edac4b8c71cd54fd9f25ec14870", + 100, + test::AcmReceiveTestOldApi::kStereoOutput); +} + +TEST_F(AcmSenderBitExactnessOldApi, Pcm16_stereo_32000khz_10ms) { + ASSERT_NO_FATAL_FAILURE(SetUpTest("L16", 32000, 2, 113, 320, 320)); + Run("5f025d4f390982cc26b3d92fe02e3044", + "50e58502fb04421bf5b857dda4c96879", + 100, + test::AcmReceiveTestOldApi::kStereoOutput); +} + +TEST_F(AcmSenderBitExactnessOldApi, Pcmu_20ms) { + ASSERT_NO_FATAL_FAILURE(SetUpTest("PCMU", 8000, 1, 0, 160, 160)); + Run("81a9d4c0bb72e9becc43aef124c981e9", + "8f9b8750bd80fe26b6cbf6659b89f0f9", + 50, + test::AcmReceiveTestOldApi::kMonoOutput); +} + +TEST_F(AcmSenderBitExactnessOldApi, Pcma_20ms) { + ASSERT_NO_FATAL_FAILURE(SetUpTest("PCMA", 8000, 1, 8, 160, 160)); + Run("39611f798969053925a49dc06d08de29", + "6ad745e55aa48981bfc790d0eeef2dd1", + 50, + test::AcmReceiveTestOldApi::kMonoOutput); +} + +TEST_F(AcmSenderBitExactnessOldApi, Pcmu_stereo_20ms) { + ASSERT_NO_FATAL_FAILURE(SetUpTest("PCMU", 8000, 2, 110, 160, 160)); + Run("437bec032fdc5cbaa0d5175430af7b18", + "60b6f25e8d1e74cb679cfe756dd9bca5", + 50, + test::AcmReceiveTestOldApi::kStereoOutput); +} + +TEST_F(AcmSenderBitExactnessOldApi, Pcma_stereo_20ms) { + ASSERT_NO_FATAL_FAILURE(SetUpTest("PCMA", 8000, 2, 118, 160, 160)); + Run("a5c6d83c5b7cedbeff734238220a4b0c", + "92b282c83efd20e7eeef52ba40842cf7", + 50, + test::AcmReceiveTestOldApi::kStereoOutput); +} + +TEST_F(AcmSenderBitExactnessOldApi, DISABLED_ON_ANDROID(Ilbc_30ms)) { + ASSERT_NO_FATAL_FAILURE(SetUpTest("ILBC", 8000, 1, 102, 240, 240)); + Run(AcmReceiverBitExactnessOldApi::PlatformChecksum( + "7b6ec10910debd9af08011d3ed5249f7", + "android_audio", + "7b6ec10910debd9af08011d3ed5249f7"), + AcmReceiverBitExactnessOldApi::PlatformChecksum( + "cfae2e9f6aba96e145f2bcdd5050ce78", + "android_payload", + "cfae2e9f6aba96e145f2bcdd5050ce78"), + 33, + test::AcmReceiveTestOldApi::kMonoOutput); +} + +TEST_F(AcmSenderBitExactnessOldApi, DISABLED_ON_ANDROID(G722_20ms)) { + ASSERT_NO_FATAL_FAILURE(SetUpTest("G722", 16000, 1, 9, 320, 160)); + Run(AcmReceiverBitExactnessOldApi::PlatformChecksum( + "7d759436f2533582950d148b5161a36c", + "android_audio", + "7d759436f2533582950d148b5161a36c"), + AcmReceiverBitExactnessOldApi::PlatformChecksum( + "fc68a87e1380614e658087cb35d5ca10", + "android_payload", + "fc68a87e1380614e658087cb35d5ca10"), + 50, + test::AcmReceiveTestOldApi::kMonoOutput); +} + +TEST_F(AcmSenderBitExactnessOldApi, DISABLED_ON_ANDROID(G722_stereo_20ms)) { + ASSERT_NO_FATAL_FAILURE(SetUpTest("G722", 16000, 2, 119, 320, 160)); + Run(AcmReceiverBitExactnessOldApi::PlatformChecksum( + "7190ee718ab3d80eca181e5f7140c210", + "android_audio", + "7190ee718ab3d80eca181e5f7140c210"), + AcmReceiverBitExactnessOldApi::PlatformChecksum( + "66516152eeaa1e650ad94ff85f668dac", + "android_payload", + "66516152eeaa1e650ad94ff85f668dac"), + 50, + test::AcmReceiveTestOldApi::kStereoOutput); +} + +TEST_F(AcmSenderBitExactnessOldApi, Opus_stereo_20ms) { + ASSERT_NO_FATAL_FAILURE(SetUpTest("opus", 48000, 2, 120, 960, 960)); + Run(AcmReceiverBitExactnessOldApi::PlatformChecksum( + "855041f2490b887302bce9d544731849", + "1e1a0fce893fef2d66886a7f09e2ebce", + "855041f2490b887302bce9d544731849"), + AcmReceiverBitExactnessOldApi::PlatformChecksum( + "d781cce1ab986b618d0da87226cdde30", + "1a1fe04dd12e755949987c8d729fb3e0", + "d781cce1ab986b618d0da87226cdde30"), + 50, + test::AcmReceiveTestOldApi::kStereoOutput); +} + +} // namespace webrtc diff --git a/modules/audio_coding/main/interface/audio_coding_module.h b/modules/audio_coding/main/interface/audio_coding_module.h index 389b93fe..8d73285a 100644 --- a/modules/audio_coding/main/interface/audio_coding_module.h +++ b/modules/audio_coding/main/interface/audio_coding_module.h @@ -1015,6 +1015,14 @@ class AudioCoding { playout_channels(1), playout_frequency_hz(32000) {} + AudioCodingModule::Config ToOldConfig() const { + AudioCodingModule::Config old_config; + old_config.id = 0; + old_config.neteq_config = neteq_config; + old_config.clock = clock; + return old_config; + } + NetEq::Config neteq_config; Clock* clock; AudioPacketizationCallback* transport; @@ -1046,6 +1054,9 @@ class AudioCoding { // codec that was registered in the latest call to RegisterSendCodec(). virtual const AudioEncoder* GetSenderInfo() const = 0; + // Temporary solution to be used during refactoring. + virtual const CodecInst* GetSenderCodecInst() = 0; + // Adds 10 ms of raw (PCM) audio data to the encoder. If the sampling // frequency of the audio does not match the sampling frequency of the // current encoder, ACM will resample the audio. @@ -1139,6 +1150,22 @@ class AudioCoding { // Disables NACK. virtual void DisableNack() = 0; + + // Temporary solution to be used during refactoring. + // If DTX is enabled and the codec does not have internal DTX/VAD + // WebRtc VAD will be automatically enabled and |enable_vad| is ignored. + // + // If DTX is disabled but VAD is enabled no DTX packets are sent, + // regardless of whether the codec has internal DTX/VAD or not. In this + // case, WebRtc VAD is running to label frames as active/in-active. + // + // NOTE! VAD/DTX is not supported when sending stereo. + // + // Return true if successful, false otherwise. + virtual bool SetVad(bool enable_dtx, + bool enable_vad, + ACMVADMode vad_mode) = 0; + // Returns a list of packets to request retransmission of. // |round_trip_time_ms| is an estimate of the round-trip-time (in // milliseconds). Missing packets which will be decoded sooner than the diff --git a/modules/audio_coding/neteq/audio_decoder_unittest.cc b/modules/audio_coding/neteq/audio_decoder_unittest.cc index d33c8f23..624e6a4d 100644 --- a/modules/audio_coding/neteq/audio_decoder_unittest.cc +++ b/modules/audio_coding/neteq/audio_decoder_unittest.cc @@ -348,14 +348,11 @@ class AudioDecoderIsacFloatTest : public AudioDecoderTest { virtual int EncodeFrame(const int16_t* input, size_t input_len_samples, uint8_t* output) { // Insert 3 * 10 ms. Expect non-zero output on third call. - EXPECT_EQ(0, WebRtcIsac_Encode(encoder_, input, - reinterpret_cast<int16_t*>(output))); + EXPECT_EQ(0, WebRtcIsac_Encode(encoder_, input, output)); input += input_size_; - EXPECT_EQ(0, WebRtcIsac_Encode(encoder_, input, - reinterpret_cast<int16_t*>(output))); + EXPECT_EQ(0, WebRtcIsac_Encode(encoder_, input, output)); input += input_size_; - int enc_len_bytes = - WebRtcIsac_Encode(encoder_, input, reinterpret_cast<int16_t*>(output)); + int enc_len_bytes = WebRtcIsac_Encode(encoder_, input, output); EXPECT_GT(enc_len_bytes, 0); return enc_len_bytes; } @@ -388,14 +385,11 @@ class AudioDecoderIsacSwbTest : public AudioDecoderTest { virtual int EncodeFrame(const int16_t* input, size_t input_len_samples, uint8_t* output) { // Insert 3 * 10 ms. Expect non-zero output on third call. - EXPECT_EQ(0, WebRtcIsac_Encode(encoder_, input, - reinterpret_cast<int16_t*>(output))); + EXPECT_EQ(0, WebRtcIsac_Encode(encoder_, input, output)); input += input_size_; - EXPECT_EQ(0, WebRtcIsac_Encode(encoder_, input, - reinterpret_cast<int16_t*>(output))); + EXPECT_EQ(0, WebRtcIsac_Encode(encoder_, input, output)); input += input_size_; - int enc_len_bytes = - WebRtcIsac_Encode(encoder_, input, reinterpret_cast<int16_t*>(output)); + int enc_len_bytes = WebRtcIsac_Encode(encoder_, input, output); EXPECT_GT(enc_len_bytes, 0); return enc_len_bytes; } @@ -441,14 +435,11 @@ class AudioDecoderIsacFixTest : public AudioDecoderTest { virtual int EncodeFrame(const int16_t* input, size_t input_len_samples, uint8_t* output) { // Insert 3 * 10 ms. Expect non-zero output on third call. - EXPECT_EQ(0, WebRtcIsacfix_Encode(encoder_, input, - reinterpret_cast<int16_t*>(output))); + EXPECT_EQ(0, WebRtcIsacfix_Encode(encoder_, input, output)); input += input_size_; - EXPECT_EQ(0, WebRtcIsacfix_Encode(encoder_, input, - reinterpret_cast<int16_t*>(output))); + EXPECT_EQ(0, WebRtcIsacfix_Encode(encoder_, input, output)); input += input_size_; - int enc_len_bytes = WebRtcIsacfix_Encode( - encoder_, input, reinterpret_cast<int16_t*>(output)); + int enc_len_bytes = WebRtcIsacfix_Encode(encoder_, input, output); EXPECT_GT(enc_len_bytes, 0); return enc_len_bytes; } diff --git a/modules/audio_coding/neteq/decision_logic_normal.cc b/modules/audio_coding/neteq/decision_logic_normal.cc index 97a8843a..9e422041 100644 --- a/modules/audio_coding/neteq/decision_logic_normal.cc +++ b/modules/audio_coding/neteq/decision_logic_normal.cc @@ -76,8 +76,10 @@ Operations DecisionLogicNormal::GetDecisionSpecialized( available_timestamp, play_dtmf); } else { // This implies that available_timestamp < target_timestamp, which can - // happen when a new stream or codec is received. Signal for a reset. - return kUndefined; + // happen when a new stream or codec is received. Do Expand instead, and + // wait for a newer packet to arrive, or for the buffer to flush (resulting + // in a master reset). + return kExpand; } } diff --git a/modules/audio_coding/neteq/neteq_external_decoder_unittest.cc b/modules/audio_coding/neteq/neteq_external_decoder_unittest.cc index 77615257..6a8eafa1 100644 --- a/modules/audio_coding/neteq/neteq_external_decoder_unittest.cc +++ b/modules/audio_coding/neteq/neteq_external_decoder_unittest.cc @@ -19,6 +19,7 @@ #include "webrtc/modules/audio_coding/neteq/mock/mock_external_decoder_pcm16b.h" #include "webrtc/modules/audio_coding/neteq/tools/input_audio_file.h" #include "webrtc/modules/audio_coding/neteq/tools/rtp_generator.h" +#include "webrtc/system_wrappers/interface/compile_assert.h" #include "webrtc/system_wrappers/interface/scoped_ptr.h" #include "webrtc/test/testsupport/fileutils.h" #include "webrtc/test/testsupport/gtest_disable.h" @@ -26,6 +27,7 @@ namespace webrtc { using ::testing::_; +using ::testing::Return; // This test encodes a few packets of PCM16b 32 kHz data and inserts it into two // different NetEq instances. The first instance uses the internal version of @@ -50,10 +52,9 @@ class NetEqExternalDecoderTest : public ::testing::Test { payload_size_bytes_(0), last_send_time_(0), last_arrival_time_(0) { - NetEq::Config config; - config.sample_rate_hz = sample_rate_hz_; - neteq_external_ = NetEq::Create(config); - neteq_ = NetEq::Create(config); + config_.sample_rate_hz = sample_rate_hz_; + neteq_external_ = NetEq::Create(config_); + neteq_ = NetEq::Create(config_); input_ = new int16_t[frame_size_samples_]; encoded_ = new uint8_t[2 * frame_size_samples_]; } @@ -158,6 +159,8 @@ class NetEqExternalDecoderTest : public ::testing::Test { EXPECT_EQ(output_size_samples_, samples_per_channel); } + virtual int NumExpectedDecodeCalls(int num_loops) const { return num_loops; } + void RunTest(int num_loops) { // Get next input packets (mono and multi-channel). int next_send_time; @@ -169,7 +172,7 @@ class NetEqExternalDecoderTest : public ::testing::Test { } while (Lost()); // If lost, immediately read the next packet. EXPECT_CALL(*external_decoder_, Decode(_, payload_size_bytes_, _, _)) - .Times(num_loops); + .Times(NumExpectedDecodeCalls(num_loops)); int time_now = 0; for (int k = 0; k < num_loops; ++k) { @@ -196,11 +199,12 @@ class NetEqExternalDecoderTest : public ::testing::Test { } } - const int sample_rate_hz_; - const int samples_per_ms_; + NetEq::Config config_; + int sample_rate_hz_; + int samples_per_ms_; const int frame_size_ms_; - const int frame_size_samples_; - const int output_size_samples_; + int frame_size_samples_; + int output_size_samples_; NetEq* neteq_external_; NetEq* neteq_; scoped_ptr<MockExternalPcm16B> external_decoder_; @@ -220,4 +224,256 @@ TEST_F(NetEqExternalDecoderTest, RunTest) { RunTest(100); // Run 100 laps @ 10 ms each in the test loop. } +class LargeTimestampJumpTest : public NetEqExternalDecoderTest { + protected: + enum TestStates { + kInitialPhase, + kNormalPhase, + kExpandPhase, + kFadedExpandPhase, + kRecovered + }; + + LargeTimestampJumpTest() + : NetEqExternalDecoderTest(), test_state_(kInitialPhase) { + sample_rate_hz_ = 8000; + samples_per_ms_ = sample_rate_hz_ / 1000; + frame_size_samples_ = frame_size_ms_ * samples_per_ms_; + output_size_samples_ = frame_size_ms_ * samples_per_ms_; + EXPECT_CALL(*external_decoder_, Die()).Times(1); + external_decoder_.reset(new MockExternalPcm16B(kDecoderPCM16B)); + } + + void SetUp() OVERRIDE { + const std::string file_name = + webrtc::test::ResourcePath("audio_coding/testfile32kHz", "pcm"); + input_file_.reset(new test::InputAudioFile(file_name)); + assert(sample_rate_hz_ == 8000); + NetEqDecoder decoder = kDecoderPCM16B; + EXPECT_CALL(*external_decoder_, Init()); + EXPECT_CALL(*external_decoder_, HasDecodePlc()) + .WillRepeatedly(Return(false)); + // NetEq is not allowed to delete the external decoder (hence Times(0)). + EXPECT_CALL(*external_decoder_, Die()).Times(0); + ASSERT_EQ(NetEq::kOK, + neteq_external_->RegisterExternalDecoder( + external_decoder_.get(), decoder, kPayloadType)); + ASSERT_EQ(NetEq::kOK, neteq_->RegisterPayloadType(decoder, kPayloadType)); + } + + void InsertPackets(int next_arrival_time) OVERRIDE { + // Insert packet in external decoder instance. + EXPECT_CALL(*external_decoder_, + IncomingPacket(_, + payload_size_bytes_, + rtp_header_.header.sequenceNumber, + rtp_header_.header.timestamp, + next_arrival_time)); + ASSERT_EQ( + NetEq::kOK, + neteq_external_->InsertPacket( + rtp_header_, encoded_, payload_size_bytes_, next_arrival_time)); + } + + void GetOutputAudio() OVERRIDE { + NetEqOutputType output_type; + int samples_per_channel; + int num_channels; + // Get audio from external decoder instance. + ASSERT_EQ(NetEq::kOK, + neteq_external_->GetAudio(kMaxBlockSize, + output_external_, + &samples_per_channel, + &num_channels, + &output_type)); + EXPECT_EQ(1, num_channels); + EXPECT_EQ(output_size_samples_, samples_per_channel); + UpdateState(output_type); + } + + virtual void UpdateState(NetEqOutputType output_type) { + switch (test_state_) { + case kInitialPhase: { + if (output_type == kOutputNormal) { + test_state_ = kNormalPhase; + } + break; + } + case kNormalPhase: { + if (output_type == kOutputPLC) { + test_state_ = kExpandPhase; + } + break; + } + case kExpandPhase: { + if (output_type == kOutputPLCtoCNG) { + test_state_ = kFadedExpandPhase; + } + break; + } + case kFadedExpandPhase: { + if (output_type == kOutputNormal) { + test_state_ = kRecovered; + } + break; + } + case kRecovered: { + break; + } + } + } + + void VerifyOutput(size_t num_samples) const OVERRIDE { + if (test_state_ == kExpandPhase || test_state_ == kFadedExpandPhase) { + // Don't verify the output in this phase of the test. + return; + } + for (size_t i = 0; i < num_samples; ++i) { + if (output_external_[i] != 0) + return; + } + EXPECT_TRUE(false) + << "Expected at least one non-zero sample in each output block."; + } + + int NumExpectedDecodeCalls(int num_loops) const OVERRIDE { + // Some packets won't be decoded because of the buffer being flushed after + // the timestamp jump. + return num_loops - (config_.max_packets_in_buffer + 1); + } + + TestStates test_state_; +}; + +TEST_F(LargeTimestampJumpTest, JumpLongerThanHalfRange) { + // Set the timestamp series to start at 2880, increase to 7200, then jump to + // 2869342376. The sequence numbers start at 42076 and increase by 1 for each + // packet, also when the timestamp jumps. + static const uint16_t kStartSeqeunceNumber = 42076; + static const uint32_t kStartTimestamp = 2880; + static const uint32_t kJumpFromTimestamp = 7200; + static const uint32_t kJumpToTimestamp = 2869342376; + COMPILE_ASSERT(kJumpFromTimestamp < kJumpToTimestamp, + timestamp_jump_should_not_result_in_wrap); + COMPILE_ASSERT( + static_cast<uint32_t>(kJumpToTimestamp - kJumpFromTimestamp) > 0x7FFFFFFF, + jump_should_be_larger_than_half_range); + // Replace the default RTP generator with one that jumps in timestamp. + rtp_generator_.reset(new test::TimestampJumpRtpGenerator(samples_per_ms_, + kStartSeqeunceNumber, + kStartTimestamp, + kJumpFromTimestamp, + kJumpToTimestamp)); + + RunTest(130); // Run 130 laps @ 10 ms each in the test loop. + EXPECT_EQ(kRecovered, test_state_); +} + +TEST_F(LargeTimestampJumpTest, JumpLongerThanHalfRangeAndWrap) { + // Make a jump larger than half the 32-bit timestamp range. Set the start + // timestamp such that the jump will result in a wrap around. + static const uint16_t kStartSeqeunceNumber = 42076; + // Set the jump length slightly larger than 2^31. + static const uint32_t kStartTimestamp = 3221223116; + static const uint32_t kJumpFromTimestamp = 3221223216; + static const uint32_t kJumpToTimestamp = 1073744278; + COMPILE_ASSERT(kJumpToTimestamp < kJumpFromTimestamp, + timestamp_jump_should_result_in_wrap); + COMPILE_ASSERT( + static_cast<uint32_t>(kJumpToTimestamp - kJumpFromTimestamp) > 0x7FFFFFFF, + jump_should_be_larger_than_half_range); + // Replace the default RTP generator with one that jumps in timestamp. + rtp_generator_.reset(new test::TimestampJumpRtpGenerator(samples_per_ms_, + kStartSeqeunceNumber, + kStartTimestamp, + kJumpFromTimestamp, + kJumpToTimestamp)); + + RunTest(130); // Run 130 laps @ 10 ms each in the test loop. + EXPECT_EQ(kRecovered, test_state_); +} + +class ShortTimestampJumpTest : public LargeTimestampJumpTest { + protected: + void UpdateState(NetEqOutputType output_type) OVERRIDE { + switch (test_state_) { + case kInitialPhase: { + if (output_type == kOutputNormal) { + test_state_ = kNormalPhase; + } + break; + } + case kNormalPhase: { + if (output_type == kOutputPLC) { + test_state_ = kExpandPhase; + } + break; + } + case kExpandPhase: { + if (output_type == kOutputNormal) { + test_state_ = kRecovered; + } + break; + } + case kRecovered: { + break; + } + default: { FAIL(); } + } + } + + int NumExpectedDecodeCalls(int num_loops) const OVERRIDE { + // Some packets won't be decoded because of the timestamp jump. + return num_loops - 2; + } +}; + +TEST_F(ShortTimestampJumpTest, JumpShorterThanHalfRange) { + // Make a jump shorter than half the 32-bit timestamp range. Set the start + // timestamp such that the jump will not result in a wrap around. + static const uint16_t kStartSeqeunceNumber = 42076; + // Set the jump length slightly smaller than 2^31. + static const uint32_t kStartTimestamp = 4711; + static const uint32_t kJumpFromTimestamp = 4811; + static const uint32_t kJumpToTimestamp = 2147483747; + COMPILE_ASSERT(kJumpFromTimestamp < kJumpToTimestamp, + timestamp_jump_should_not_result_in_wrap); + COMPILE_ASSERT( + static_cast<uint32_t>(kJumpToTimestamp - kJumpFromTimestamp) < 0x7FFFFFFF, + jump_should_be_smaller_than_half_range); + // Replace the default RTP generator with one that jumps in timestamp. + rtp_generator_.reset(new test::TimestampJumpRtpGenerator(samples_per_ms_, + kStartSeqeunceNumber, + kStartTimestamp, + kJumpFromTimestamp, + kJumpToTimestamp)); + + RunTest(130); // Run 130 laps @ 10 ms each in the test loop. + EXPECT_EQ(kRecovered, test_state_); +} + +TEST_F(ShortTimestampJumpTest, JumpShorterThanHalfRangeAndWrap) { + // Make a jump shorter than half the 32-bit timestamp range. Set the start + // timestamp such that the jump will result in a wrap around. + static const uint16_t kStartSeqeunceNumber = 42076; + // Set the jump length slightly smaller than 2^31. + static const uint32_t kStartTimestamp = 3221227827; + static const uint32_t kJumpFromTimestamp = 3221227927; + static const uint32_t kJumpToTimestamp = 1073739567; + COMPILE_ASSERT(kJumpToTimestamp < kJumpFromTimestamp, + timestamp_jump_should_result_in_wrap); + COMPILE_ASSERT( + static_cast<uint32_t>(kJumpToTimestamp - kJumpFromTimestamp) < 0x7FFFFFFF, + jump_should_be_smaller_than_half_range); + // Replace the default RTP generator with one that jumps in timestamp. + rtp_generator_.reset(new test::TimestampJumpRtpGenerator(samples_per_ms_, + kStartSeqeunceNumber, + kStartTimestamp, + kJumpFromTimestamp, + kJumpToTimestamp)); + + RunTest(130); // Run 130 laps @ 10 ms each in the test loop. + EXPECT_EQ(kRecovered, test_state_); +} + } // namespace webrtc diff --git a/modules/audio_coding/neteq/neteq_impl.cc b/modules/audio_coding/neteq/neteq_impl.cc index a184fc33..d7147335 100644 --- a/modules/audio_coding/neteq/neteq_impl.cc +++ b/modules/audio_coding/neteq/neteq_impl.cc @@ -864,9 +864,6 @@ int NetEqImpl::GetDecision(Operations* operation, assert(sync_buffer_.get()); uint32_t end_timestamp = sync_buffer_->end_timestamp(); - if (!new_codec_) { - packet_buffer_->DiscardOldPackets(end_timestamp); - } const RTPHeader* header = packet_buffer_->NextRtpHeader(); if (decision_logic_->CngRfc3389On() || last_mode_ == kModeRfc3389Cng) { @@ -1817,6 +1814,14 @@ int NetEqImpl::ExtractPackets(int required_samples, PacketList* packet_list) { } } while (extracted_samples < required_samples && next_packet_available); + if (extracted_samples > 0) { + // Delete old packets only when we are going to decode something. Otherwise, + // we could end up in the situation where we never decode anything, since + // all incoming packets are considered too old but the buffer will also + // never be flooded and flushed. + packet_buffer_->DiscardOldPackets(timestamp_); + } + return extracted_samples; } diff --git a/modules/audio_coding/neteq/neteq_impl.h b/modules/audio_coding/neteq/neteq_impl.h index d3dff185..cfe6a402 100644 --- a/modules/audio_coding/neteq/neteq_impl.h +++ b/modules/audio_coding/neteq/neteq_impl.h @@ -14,6 +14,7 @@ #include <vector> #include "webrtc/base/constructormagic.h" +#include "webrtc/base/thread_annotations.h" #include "webrtc/modules/audio_coding/neteq/audio_multi_vector.h" #include "webrtc/modules/audio_coding/neteq/defines.h" #include "webrtc/modules/audio_coding/neteq/interface/neteq.h" @@ -22,7 +23,6 @@ #include "webrtc/modules/audio_coding/neteq/rtcp.h" #include "webrtc/modules/audio_coding/neteq/statistics_calculator.h" #include "webrtc/system_wrappers/interface/scoped_ptr.h" -#include "webrtc/system_wrappers/interface/thread_annotations.h" #include "webrtc/typedefs.h" namespace webrtc { diff --git a/modules/audio_coding/neteq/test/RTPencode.cc b/modules/audio_coding/neteq/test/RTPencode.cc index 93b366b9..92bccee2 100644 --- a/modules/audio_coding/neteq/test/RTPencode.cc +++ b/modules/audio_coding/neteq/test/RTPencode.cc @@ -1632,9 +1632,13 @@ int NetEQTest_encode(int coder, int16_t *indata, int frameLen, unsigned char * e cdlen=0; while (cdlen<=0) { #ifdef CODEC_ISAC /* floating point */ - cdlen=WebRtcIsac_Encode(ISAC_inst[k],&indata[noOfCalls*160],(int16_t*)encoded); + cdlen = WebRtcIsac_Encode(ISAC_inst[k], + &indata[noOfCalls * 160], + encoded); #else /* fixed point */ - cdlen=WebRtcIsacfix_Encode(ISAC_inst[k],&indata[noOfCalls*160],(int16_t*)encoded); + cdlen = WebRtcIsacfix_Encode(ISAC_inst[k], + &indata[noOfCalls * 160], + encoded); #endif noOfCalls++; } @@ -1645,7 +1649,9 @@ int NetEQTest_encode(int coder, int16_t *indata, int frameLen, unsigned char * e int noOfCalls=0; cdlen=0; while (cdlen<=0) { - cdlen=WebRtcIsac_Encode(ISACSWB_inst[k],&indata[noOfCalls*320],(int16_t*)encoded); + cdlen = WebRtcIsac_Encode(ISACSWB_inst[k], + &indata[noOfCalls * 320], + encoded); noOfCalls++; } } diff --git a/modules/audio_coding/neteq/test/neteq_isac_quality_test.cc b/modules/audio_coding/neteq/test/neteq_isac_quality_test.cc index 6b0f4828..b672a0c2 100644 --- a/modules/audio_coding/neteq/test/neteq_isac_quality_test.cc +++ b/modules/audio_coding/neteq/test/neteq_isac_quality_test.cc @@ -138,8 +138,7 @@ int NetEqIsacQualityTest::EncodeBlock(int16_t* in_data, // The Isac encoder does not perform encoding (and returns 0) until it // receives a sequence of sub-blocks that amount to the frame duration. EXPECT_EQ(0, value); - value = WebRtcIsacfix_Encode(isac_encoder_, &in_data[pointer], - reinterpret_cast<int16_t*>(payload)); + value = WebRtcIsacfix_Encode(isac_encoder_, &in_data[pointer], payload); } EXPECT_GT(value, 0); return value; diff --git a/modules/audio_coding/neteq/tools/rtp_generator.cc b/modules/audio_coding/neteq/tools/rtp_generator.cc index 17ac209f..db9988d9 100644 --- a/modules/audio_coding/neteq/tools/rtp_generator.cc +++ b/modules/audio_coding/neteq/tools/rtp_generator.cc @@ -44,5 +44,19 @@ void RtpGenerator::set_drift_factor(double factor) { } } +uint32_t TimestampJumpRtpGenerator::GetRtpHeader(uint8_t payload_type, + size_t payload_length_samples, + WebRtcRTPHeader* rtp_header) { + uint32_t ret = RtpGenerator::GetRtpHeader( + payload_type, payload_length_samples, rtp_header); + if (timestamp_ - static_cast<uint32_t>(payload_length_samples) <= + jump_from_timestamp_ && + timestamp_ > jump_from_timestamp_) { + // We just moved across the |jump_from_timestamp_| timestamp. Do the jump. + timestamp_ = jump_to_timestamp_; + } + return ret; +} + } // namespace test } // namespace webrtc diff --git a/modules/audio_coding/neteq/tools/rtp_generator.h b/modules/audio_coding/neteq/tools/rtp_generator.h index d3824c8d..22804361 100644 --- a/modules/audio_coding/neteq/tools/rtp_generator.h +++ b/modules/audio_coding/neteq/tools/rtp_generator.h @@ -34,24 +34,50 @@ class RtpGenerator { drift_factor_(0.0) { } + virtual ~RtpGenerator() {} + // Writes the next RTP header to |rtp_header|, which will be of type // |payload_type|. Returns the send time for this packet (in ms). The value of // |payload_length_samples| determines the send time for the next packet. - uint32_t GetRtpHeader(uint8_t payload_type, size_t payload_length_samples, - WebRtcRTPHeader* rtp_header); + virtual uint32_t GetRtpHeader(uint8_t payload_type, + size_t payload_length_samples, + WebRtcRTPHeader* rtp_header); void set_drift_factor(double factor); - private: + protected: uint16_t seq_number_; uint32_t timestamp_; uint32_t next_send_time_ms_; const uint32_t ssrc_; const int samples_per_ms_; double drift_factor_; + + private: DISALLOW_COPY_AND_ASSIGN(RtpGenerator); }; +class TimestampJumpRtpGenerator : public RtpGenerator { + public: + TimestampJumpRtpGenerator(int samples_per_ms, + uint16_t start_seq_number, + uint32_t start_timestamp, + uint32_t jump_from_timestamp, + uint32_t jump_to_timestamp) + : RtpGenerator(samples_per_ms, start_seq_number, start_timestamp), + jump_from_timestamp_(jump_from_timestamp), + jump_to_timestamp_(jump_to_timestamp) {} + + uint32_t GetRtpHeader(uint8_t payload_type, + size_t payload_length_samples, + WebRtcRTPHeader* rtp_header) OVERRIDE; + + private: + uint32_t jump_from_timestamp_; + uint32_t jump_to_timestamp_; + DISALLOW_COPY_AND_ASSIGN(TimestampJumpRtpGenerator); +}; + } // namespace test } // namespace webrtc #endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_RTP_GENERATOR_H_ diff --git a/modules/audio_processing/agc/analog_agc.c b/modules/audio_processing/agc/analog_agc.c index 0376dae5..32929dda 100644 --- a/modules/audio_processing/agc/analog_agc.c +++ b/modules/audio_processing/agc/analog_agc.c @@ -19,7 +19,7 @@ #include <assert.h> #include <stdlib.h> -#ifdef AGC_DEBUG //test log +#ifdef WEBRTC_AGC_DEBUG_DUMP #include <stdio.h> #endif #include "webrtc/modules/audio_processing/agc/analog_agc.h" @@ -139,10 +139,10 @@ int WebRtcAgc_AddMic(void *state, int16_t *in_mic, int16_t *in_mic_H, L = 8; } else { -#ifdef AGC_DEBUG //test log +#ifdef WEBRTC_AGC_DEBUG_DUMP fprintf(stt->fpt, "AGC->add_mic, frame %d: Invalid number of samples\n\n", - (stt->fcount + 1)); + stt->fcount + 1); #endif return -1; } @@ -160,10 +160,10 @@ int WebRtcAgc_AddMic(void *state, int16_t *in_mic, int16_t *in_mic_H, L = 16; } else { -#ifdef AGC_DEBUG //test log +#ifdef WEBRTC_AGC_DEBUG_DUMP fprintf(stt->fpt, "AGC->add_mic, frame %d: Invalid number of samples\n\n", - (stt->fcount + 1)); + stt->fcount + 1); #endif return -1; } @@ -177,10 +177,10 @@ int WebRtcAgc_AddMic(void *state, int16_t *in_mic, int16_t *in_mic_H, L = 16; } else { -#ifdef AGC_DEBUG +#ifdef WEBRTC_AGC_DEBUG_DUMP fprintf(stt->fpt, "AGC->add_mic, frame %d: Invalid sample rate\n\n", - (stt->fcount + 1)); + stt->fcount + 1); #endif return -1; } @@ -343,7 +343,7 @@ int WebRtcAgc_AddFarend(void *state, const int16_t *in_far, int16_t samples) { if ((samples != 80) && (samples != 160)) { -#ifdef AGC_DEBUG //test log +#ifdef WEBRTC_AGC_DEBUG_DUMP fprintf(stt->fpt, "AGC->add_far_end, frame %d: Invalid number of samples\n\n", stt->fcount); @@ -355,7 +355,7 @@ int WebRtcAgc_AddFarend(void *state, const int16_t *in_far, int16_t samples) { if ((samples != 160) && (samples != 320)) { -#ifdef AGC_DEBUG //test log +#ifdef WEBRTC_AGC_DEBUG_DUMP fprintf(stt->fpt, "AGC->add_far_end, frame %d: Invalid number of samples\n\n", stt->fcount); @@ -367,7 +367,7 @@ int WebRtcAgc_AddFarend(void *state, const int16_t *in_far, int16_t samples) { if ((samples != 160) && (samples != 320)) { -#ifdef AGC_DEBUG //test log +#ifdef WEBRTC_AGC_DEBUG_DUMP fprintf(stt->fpt, "AGC->add_far_end, frame %d: Invalid number of samples\n\n", stt->fcount); @@ -377,7 +377,7 @@ int WebRtcAgc_AddFarend(void *state, const int16_t *in_far, int16_t samples) subFrames = 160; } else { -#ifdef AGC_DEBUG //test log +#ifdef WEBRTC_AGC_DEBUG_DUMP fprintf(stt->fpt, "AGC->add_far_end, frame %d: Invalid sample rate\n\n", stt->fcount + 1); @@ -657,10 +657,12 @@ void WebRtcAgc_ZeroCtrl(Agc_t *stt, int32_t *inMicLevel, int32_t *env) stt->micVol = *inMicLevel; } -#ifdef AGC_DEBUG //test log +#ifdef WEBRTC_AGC_DEBUG_DUMP fprintf(stt->fpt, - "\t\tAGC->zeroCntrl, frame %d: 500 ms under threshold, micVol:\n", - stt->fcount, stt->micVol); + "\t\tAGC->zeroCntrl, frame %d: 500 ms under threshold," + " micVol: %d\n", + stt->fcount, + stt->micVol); #endif stt->activeSpeech = 0; @@ -771,14 +773,18 @@ int32_t WebRtcAgc_ProcessAnalog(void *state, int32_t inMicLevel, if (inMicLevelTmp > stt->maxAnalog) { -#ifdef AGC_DEBUG //test log - fprintf(stt->fpt, "\tAGC->ProcessAnalog, frame %d: micLvl > maxAnalog\n", stt->fcount); +#ifdef WEBRTC_AGC_DEBUG_DUMP + fprintf(stt->fpt, + "\tAGC->ProcessAnalog, frame %d: micLvl > maxAnalog\n", + stt->fcount); #endif return -1; } else if (inMicLevelTmp < stt->minLevel) { -#ifdef AGC_DEBUG //test log - fprintf(stt->fpt, "\tAGC->ProcessAnalog, frame %d: micLvl < minLevel\n", stt->fcount); +#ifdef WEBRTC_AGC_DEBUG_DUMP + fprintf(stt->fpt, + "\tAGC->ProcessAnalog, frame %d: micLvl < minLevel\n", + stt->fcount); #endif return -1; } @@ -813,9 +819,10 @@ int32_t WebRtcAgc_ProcessAnalog(void *state, int32_t inMicLevel, #ifdef MIC_LEVEL_FEEDBACK //stt->numBlocksMicLvlSat = 0; #endif -#ifdef AGC_DEBUG //test log +#ifdef WEBRTC_AGC_DEBUG_DUMP fprintf(stt->fpt, - "\tAGC->ProcessAnalog, frame %d: micLvl < minLevel by manual decrease, raise vol\n", + "\tAGC->ProcessAnalog, frame %d: micLvl < minLevel by manual" + " decrease, raise vol\n", stt->fcount); #endif } @@ -871,10 +878,11 @@ int32_t WebRtcAgc_ProcessAnalog(void *state, int32_t inMicLevel, } inMicLevelTmp = stt->micVol; -#ifdef AGC_DEBUG //test log +#ifdef WEBRTC_AGC_DEBUG_DUMP fprintf(stt->fpt, "\tAGC->ProcessAnalog, frame %d: saturated, micVol = %d\n", - stt->fcount, stt->micVol); + stt->fcount, + stt->micVol); #endif if (stt->micVol < stt->minOutput) @@ -1011,10 +1019,13 @@ int32_t WebRtcAgc_ProcessAnalog(void *state, int32_t inMicLevel, #ifdef MIC_LEVEL_FEEDBACK //stt->numBlocksMicLvlSat = 0; #endif -#ifdef AGC_DEBUG //test log +#ifdef WEBRTC_AGC_DEBUG_DUMP fprintf(stt->fpt, - "\tAGC->ProcessAnalog, frame %d: measure > 2ndUpperLim, micVol = %d, maxLevel = %d\n", - stt->fcount, stt->micVol, stt->maxLevel); + "\tAGC->ProcessAnalog, frame %d: measure >" + " 2ndUpperLim, micVol = %d, maxLevel = %d\n", + stt->fcount, + stt->micVol, + stt->maxLevel); #endif } } else if (stt->Rxx160_LPw32 > stt->upperLimit) @@ -1054,10 +1065,13 @@ int32_t WebRtcAgc_ProcessAnalog(void *state, int32_t inMicLevel, #ifdef MIC_LEVEL_FEEDBACK //stt->numBlocksMicLvlSat = 0; #endif -#ifdef AGC_DEBUG //test log +#ifdef WEBRTC_AGC_DEBUG_DUMP fprintf(stt->fpt, - "\tAGC->ProcessAnalog, frame %d: measure > UpperLim, micVol = %d, maxLevel = %d\n", - stt->fcount, stt->micVol, stt->maxLevel); + "\tAGC->ProcessAnalog, frame %d: measure >" + " UpperLim, micVol = %d, maxLevel = %d\n", + stt->fcount, + stt->micVol, + stt->maxLevel); #endif } } else if (stt->Rxx160_LPw32 < stt->lowerSecondaryLimit) @@ -1113,10 +1127,12 @@ int32_t WebRtcAgc_ProcessAnalog(void *state, int32_t inMicLevel, fprintf(stderr, "Sat mic Level: %d\n", stt->numBlocksMicLvlSat); } #endif -#ifdef AGC_DEBUG //test log +#ifdef WEBRTC_AGC_DEBUG_DUMP fprintf(stt->fpt, - "\tAGC->ProcessAnalog, frame %d: measure < 2ndLowerLim, micVol = %d\n", - stt->fcount, stt->micVol); + "\tAGC->ProcessAnalog, frame %d: measure <" + " 2ndLowerLim, micVol = %d\n", + stt->fcount, + stt->micVol); #endif } } else if (stt->Rxx160_LPw32 < stt->lowerLimit) @@ -1172,10 +1188,11 @@ int32_t WebRtcAgc_ProcessAnalog(void *state, int32_t inMicLevel, fprintf(stderr, "Sat mic Level: %d\n", stt->numBlocksMicLvlSat); } #endif -#ifdef AGC_DEBUG //test log +#ifdef WEBRTC_AGC_DEBUG_DUMP fprintf(stt->fpt, "\tAGC->ProcessAnalog, frame %d: measure < LowerLim, micVol = %d\n", - stt->fcount, stt->micVol); + stt->fcount, + stt->micVol); #endif } @@ -1272,9 +1289,10 @@ int WebRtcAgc_Process(void *agcInst, const int16_t *in_near, { if ((samples != 80) && (samples != 160)) { -#ifdef AGC_DEBUG //test log +#ifdef WEBRTC_AGC_DEBUG_DUMP fprintf(stt->fpt, - "AGC->Process, frame %d: Invalid number of samples\n\n", stt->fcount); + "AGC->Process, frame %d: Invalid number of samples\n\n", + stt->fcount); #endif return -1; } @@ -1283,9 +1301,10 @@ int WebRtcAgc_Process(void *agcInst, const int16_t *in_near, { if ((samples != 160) && (samples != 320)) { -#ifdef AGC_DEBUG //test log +#ifdef WEBRTC_AGC_DEBUG_DUMP fprintf(stt->fpt, - "AGC->Process, frame %d: Invalid number of samples\n\n", stt->fcount); + "AGC->Process, frame %d: Invalid number of samples\n\n", + stt->fcount); #endif return -1; } @@ -1294,18 +1313,20 @@ int WebRtcAgc_Process(void *agcInst, const int16_t *in_near, { if ((samples != 160) && (samples != 320)) { -#ifdef AGC_DEBUG //test log +#ifdef WEBRTC_AGC_DEBUG_DUMP fprintf(stt->fpt, - "AGC->Process, frame %d: Invalid number of samples\n\n", stt->fcount); + "AGC->Process, frame %d: Invalid number of samples\n\n", + stt->fcount); #endif return -1; } subFrames = 160; } else { -#ifdef AGC_DEBUG// test log +#ifdef WEBRTC_AGC_DEBUG_DUMP fprintf(stt->fpt, - "AGC->Process, frame %d: Invalid sample rate\n\n", stt->fcount); + "AGC->Process, frame %d: Invalid sample rate\n\n", + stt->fcount); #endif return -1; } @@ -1341,7 +1362,7 @@ int WebRtcAgc_Process(void *agcInst, const int16_t *in_near, } } -#ifdef AGC_DEBUG//test log +#ifdef WEBRTC_AGC_DEBUG_DUMP stt->fcount++; #endif @@ -1350,8 +1371,10 @@ int WebRtcAgc_Process(void *agcInst, const int16_t *in_near, if (WebRtcAgc_ProcessDigital(&stt->digitalAgc, &in_near[i], &in_near_H[i], &out[i], &out_H[i], stt->fs, stt->lowLevelSignal) == -1) { -#ifdef AGC_DEBUG//test log - fprintf(stt->fpt, "AGC->Process, frame %d: Error from DigAGC\n\n", stt->fcount); +#ifdef WEBRTC_AGC_DEBUG_DUMP + fprintf(stt->fpt, + "AGC->Process, frame %d: Error from DigAGC\n\n", + stt->fcount); #endif return -1; } @@ -1364,8 +1387,14 @@ int WebRtcAgc_Process(void *agcInst, const int16_t *in_near, return -1; } } -#ifdef AGC_DEBUG//test log - fprintf(stt->agcLog, "%5d\t%d\t%d\t%d\n", stt->fcount, inMicLevelTmp, *outMicLevel, stt->maxLevel, stt->micVol); +#ifdef WEBRTC_AGC_DEBUG_DUMP + fprintf(stt->agcLog, + "%5d\t%d\t%d\t%d\t%d\n", + stt->fcount, + inMicLevelTmp, + *outMicLevel, + stt->maxLevel, + stt->micVol); #endif /* update queue */ @@ -1441,8 +1470,10 @@ int WebRtcAgc_set_config(void *agcInst, WebRtcAgc_config_t agcConfig) if (WebRtcAgc_CalculateGainTable(&(stt->digitalAgc.gainTable[0]), stt->compressionGaindB, stt->targetLevelDbfs, stt->limiterEnable, stt->analogTarget) == -1) { -#ifdef AGC_DEBUG//test log - fprintf(stt->fpt, "AGC->set_config, frame %d: Error from calcGainTable\n\n", stt->fcount); +#ifdef WEBRTC_AGC_DEBUG_DUMP + fprintf(stt->fpt, + "AGC->set_config, frame %d: Error from calcGainTable\n\n", + stt->fcount); #endif return -1; } @@ -1498,7 +1529,7 @@ int WebRtcAgc_Create(void **agcInst) return -1; } -#ifdef AGC_DEBUG +#ifdef WEBRTC_AGC_DEBUG_DUMP stt->fpt = fopen("./agc_test_log.txt", "wt"); stt->agcLog = fopen("./agc_debug_log.txt", "wt"); stt->digitalAgc.logFile = fopen("./agc_log.txt", "wt"); @@ -1515,7 +1546,7 @@ int WebRtcAgc_Free(void *state) Agc_t *stt; stt = (Agc_t *)state; -#ifdef AGC_DEBUG +#ifdef WEBRTC_AGC_DEBUG_DUMP fclose(stt->fpt); fclose(stt->agcLog); fclose(stt->digitalAgc.logFile); @@ -1553,13 +1584,13 @@ int WebRtcAgc_Init(void *agcInst, int32_t minLevel, int32_t maxLevel, * 2 - Digital Automatic Gain Control [-targetLevelDbfs (default -3 dBOv)] * 3 - Fixed Digital Gain [compressionGaindB (default 8 dB)] */ -#ifdef AGC_DEBUG//test log +#ifdef WEBRTC_AGC_DEBUG_DUMP stt->fcount = 0; fprintf(stt->fpt, "AGC->Init\n"); #endif if (agcMode < kAgcModeUnchanged || agcMode > kAgcModeFixedDigital) { -#ifdef AGC_DEBUG//test log +#ifdef WEBRTC_AGC_DEBUG_DUMP fprintf(stt->fpt, "AGC->Init: error, incorrect mode\n\n"); #endif return -1; @@ -1616,10 +1647,12 @@ int WebRtcAgc_Init(void *agcInst, int32_t minLevel, int32_t maxLevel, stt->numBlocksMicLvlSat = 0; stt->micLvlSat = 0; #endif -#ifdef AGC_DEBUG//test log +#ifdef WEBRTC_AGC_DEBUG_DUMP fprintf(stt->fpt, "AGC->Init: minLevel = %d, maxAnalog = %d, maxLevel = %d\n", - stt->minLevel, stt->maxAnalog, stt->maxLevel); + stt->minLevel, + stt->maxAnalog, + stt->maxLevel); #endif /* Minimum output volume is 4% higher than the available lowest volume level */ @@ -1687,13 +1720,13 @@ int WebRtcAgc_Init(void *agcInst, int32_t minLevel, int32_t maxLevel, /* Only positive values are allowed that are not too large */ if ((minLevel >= maxLevel) || (maxLevel & 0xFC000000)) { -#ifdef AGC_DEBUG//test log +#ifdef WEBRTC_AGC_DEBUG_DUMP fprintf(stt->fpt, "minLevel, maxLevel value(s) are invalid\n\n"); #endif return -1; } else { -#ifdef AGC_DEBUG//test log +#ifdef WEBRTC_AGC_DEBUG_DUMP fprintf(stt->fpt, "\n"); #endif return 0; diff --git a/modules/audio_processing/agc/analog_agc.h b/modules/audio_processing/agc/analog_agc.h index b036f449..19b324f1 100644 --- a/modules/audio_processing/agc/analog_agc.h +++ b/modules/audio_processing/agc/analog_agc.h @@ -15,9 +15,8 @@ #include "webrtc/modules/audio_processing/agc/include/gain_control.h" #include "webrtc/typedefs.h" -//#define AGC_DEBUG //#define MIC_LEVEL_FEEDBACK -#ifdef AGC_DEBUG +#ifdef WEBRTC_AGC_DEBUG_DUMP #include <stdio.h> #endif @@ -122,10 +121,10 @@ typedef struct AgcVad_t vadMic; DigitalAgc_t digitalAgc; -#ifdef AGC_DEBUG - FILE* fpt; - FILE* agcLog; - int32_t fcount; +#ifdef WEBRTC_AGC_DEBUG_DUMP + FILE* fpt; + FILE* agcLog; + int32_t fcount; #endif int16_t lowLevelSignal; diff --git a/modules/audio_processing/agc/digital_agc.c b/modules/audio_processing/agc/digital_agc.c index e74bb4c0..da087ca3 100644 --- a/modules/audio_processing/agc/digital_agc.c +++ b/modules/audio_processing/agc/digital_agc.c @@ -16,7 +16,7 @@ #include <assert.h> #include <string.h> -#ifdef AGC_DEBUG +#ifdef WEBRTC_AGC_DEBUG_DUMP #include <stdio.h> #endif @@ -274,7 +274,7 @@ int32_t WebRtcAgc_InitDigital(DigitalAgc_t *stt, int16_t agcMode) stt->gain = 65536; stt->gatePrevious = 0; stt->agcMode = agcMode; -#ifdef AGC_DEBUG +#ifdef WEBRTC_AGC_DEBUG_DUMP stt->frameCounter = 0; #endif @@ -397,9 +397,14 @@ int32_t WebRtcAgc_ProcessDigital(DigitalAgc_t *stt, const int16_t *in_near, decay = 0; } } -#ifdef AGC_DEBUG +#ifdef WEBRTC_AGC_DEBUG_DUMP stt->frameCounter++; - fprintf(stt->logFile, "%5.2f\t%d\t%d\t%d\t", (float)(stt->frameCounter) / 100, logratio, decay, stt->vadNearend.stdLongTerm); + fprintf(stt->logFile, + "%5.2f\t%d\t%d\t%d\t", + (float)(stt->frameCounter) / 100, + logratio, + decay, + stt->vadNearend.stdLongTerm); #endif // Find max amplitude per sub frame // iterate over sub frames @@ -461,10 +466,15 @@ int32_t WebRtcAgc_ProcessDigital(DigitalAgc_t *stt, const int16_t *in_near, frac = (int16_t)WEBRTC_SPL_RSHIFT_W32(tmp32, 19); // Q12 tmp32 = WEBRTC_SPL_MUL((stt->gainTable[zeros-1] - stt->gainTable[zeros]), frac); gains[k + 1] = stt->gainTable[zeros] + WEBRTC_SPL_RSHIFT_W32(tmp32, 12); -#ifdef AGC_DEBUG - if (k == 0) - { - fprintf(stt->logFile, "%d\t%d\t%d\t%d\t%d\n", env[0], cur_level, stt->capacitorFast, stt->capacitorSlow, zeros); +#ifdef WEBRTC_AGC_DEBUG_DUMP + if (k == 0) { + fprintf(stt->logFile, + "%d\t%d\t%d\t%d\t%d\n", + env[0], + cur_level, + stt->capacitorFast, + stt->capacitorSlow, + zeros); } #endif } diff --git a/modules/audio_processing/agc/digital_agc.h b/modules/audio_processing/agc/digital_agc.h index 6bd086fc..3dcd9375 100644 --- a/modules/audio_processing/agc/digital_agc.h +++ b/modules/audio_processing/agc/digital_agc.h @@ -11,7 +11,7 @@ #ifndef WEBRTC_MODULES_AUDIO_PROCESSING_AGC_MAIN_SOURCE_DIGITAL_AGC_H_ #define WEBRTC_MODULES_AUDIO_PROCESSING_AGC_MAIN_SOURCE_DIGITAL_AGC_H_ -#ifdef AGC_DEBUG +#ifdef WEBRTC_AGC_DEBUG_DUMP #include <stdio.h> #endif #include "webrtc/common_audio/signal_processing/include/signal_processing_library.h" @@ -46,9 +46,9 @@ typedef struct int16_t agcMode; AgcVad_t vadNearend; AgcVad_t vadFarend; -#ifdef AGC_DEBUG - FILE* logFile; - int frameCounter; +#ifdef WEBRTC_AGC_DEBUG_DUMP + FILE* logFile; + int frameCounter; #endif } DigitalAgc_t; diff --git a/modules/audio_processing/audio_processing.gypi b/modules/audio_processing/audio_processing.gypi index 9298f118..9bbcfae7 100644 --- a/modules/audio_processing/audio_processing.gypi +++ b/modules/audio_processing/audio_processing.gypi @@ -21,6 +21,7 @@ 'variables': { # Outputs some low-level debug files. 'aec_debug_dump%': 0, + 'agc_debug_dump%': 0, # Disables the usual mode where we trust the reported system delay # values the AEC receives. The corresponding define is set appropriately @@ -93,6 +94,9 @@ ['aec_untrusted_delay_for_testing==1', { 'defines': ['WEBRTC_UNTRUSTED_DELAY',], }], + ['agc_debug_dump==1', { + 'defines': ['WEBRTC_AGC_DEBUG_DUMP',], + }], ['enable_protobuf==1', { 'dependencies': ['audioproc_debug_proto'], 'defines': ['WEBRTC_AUDIOPROC_DEBUG_DUMP'], diff --git a/modules/audio_processing/ns/ns_core.c b/modules/audio_processing/ns/ns_core.c index 0d60aafc..285e4048 100644 --- a/modules/audio_processing/ns/ns_core.c +++ b/modules/audio_processing/ns/ns_core.c @@ -10,8 +10,8 @@ #include <math.h> #include <string.h> -//#include <stdio.h> #include <stdlib.h> + #include "webrtc/common_audio/signal_processing/include/signal_processing_library.h" #include "webrtc/modules/audio_processing/ns/include/noise_suppression.h" #include "webrtc/modules/audio_processing/ns/ns_core.h" @@ -20,37 +20,40 @@ // Set Feature Extraction Parameters void WebRtcNs_set_feature_extraction_parameters(NSinst_t* inst) { - //bin size of histogram - inst->featureExtractionParams.binSizeLrt = (float)0.1; + // bin size of histogram + inst->featureExtractionParams.binSizeLrt = (float)0.1; inst->featureExtractionParams.binSizeSpecFlat = (float)0.05; inst->featureExtractionParams.binSizeSpecDiff = (float)0.1; - //range of histogram over which lrt threshold is computed + // range of histogram over which lrt threshold is computed inst->featureExtractionParams.rangeAvgHistLrt = (float)1.0; - //scale parameters: multiply dominant peaks of the histograms by scale factor to obtain - // thresholds for prior model - inst->featureExtractionParams.factor1ModelPars = (float)1.20; //for lrt and spectral diff - inst->featureExtractionParams.factor2ModelPars = (float)0.9; //for spectral_flatness: + // scale parameters: multiply dominant peaks of the histograms by scale factor + // to obtain thresholds for prior model + inst->featureExtractionParams.factor1ModelPars = + (float)1.20; // for lrt and spectral diff + inst->featureExtractionParams.factor2ModelPars = + (float)0.9; // for spectral_flatness: // used when noise is flatter than speech - //peak limit for spectral flatness (varies between 0 and 1) + // peak limit for spectral flatness (varies between 0 and 1) inst->featureExtractionParams.thresPosSpecFlat = (float)0.6; - //limit on spacing of two highest peaks in histogram: spacing determined by bin size + // limit on spacing of two highest peaks in histogram: spacing determined by + // bin size inst->featureExtractionParams.limitPeakSpacingSpecFlat = 2 * inst->featureExtractionParams.binSizeSpecFlat; inst->featureExtractionParams.limitPeakSpacingSpecDiff = 2 * inst->featureExtractionParams.binSizeSpecDiff; - //limit on relevance of second peak: + // limit on relevance of second peak: inst->featureExtractionParams.limitPeakWeightsSpecFlat = (float)0.5; inst->featureExtractionParams.limitPeakWeightsSpecDiff = (float)0.5; // fluctuation limit of lrt feature inst->featureExtractionParams.thresFluctLrt = (float)0.05; - //limit on the max and min values for the feature thresholds + // limit on the max and min values for the feature thresholds inst->featureExtractionParams.maxLrt = (float)1.0; inst->featureExtractionParams.minLrt = (float)0.20; @@ -60,19 +63,19 @@ void WebRtcNs_set_feature_extraction_parameters(NSinst_t* inst) { inst->featureExtractionParams.maxSpecDiff = (float)1.0; inst->featureExtractionParams.minSpecDiff = (float)0.16; - //criteria of weight of histogram peak to accept/reject feature - inst->featureExtractionParams.thresWeightSpecFlat = (int)(0.3 - * (inst->modelUpdatePars[1])); //for spectral flatness - inst->featureExtractionParams.thresWeightSpecDiff = (int)(0.3 - * (inst->modelUpdatePars[1])); //for spectral difference + // criteria of weight of histogram peak to accept/reject feature + inst->featureExtractionParams.thresWeightSpecFlat = + (int)(0.3 * (inst->modelUpdatePars[1])); // for spectral flatness + inst->featureExtractionParams.thresWeightSpecDiff = + (int)(0.3 * (inst->modelUpdatePars[1])); // for spectral difference } // Initialize state int WebRtcNs_InitCore(NSinst_t* inst, uint32_t fs) { int i; - //We only support 10ms frames + // We only support 10ms frames - //check for valid pointer + // check for valid pointer if (inst == NULL) { return -1; } @@ -106,20 +109,21 @@ int WebRtcNs_InitCore(NSinst_t* inst, uint32_t fs) { inst->window = kBlocks160w256; inst->outLen = 0; } - inst->magnLen = inst->anaLen / 2 + 1; // Number of frequency bins + inst->magnLen = inst->anaLen / 2 + 1; // Number of frequency bins // Initialize fft work arrays. - inst->ip[0] = 0; // Setting this triggers initialization. + inst->ip[0] = 0; // Setting this triggers initialization. memset(inst->dataBuf, 0, sizeof(float) * ANAL_BLOCKL_MAX); WebRtc_rdft(inst->anaLen, 1, inst->dataBuf, inst->ip, inst->wfft); + memset(inst->analyzeBuf, 0, sizeof(float) * ANAL_BLOCKL_MAX); memset(inst->dataBuf, 0, sizeof(float) * ANAL_BLOCKL_MAX); memset(inst->syntBuf, 0, sizeof(float) * ANAL_BLOCKL_MAX); - //for HB processing + // for HB processing memset(inst->dataBufHB, 0, sizeof(float) * ANAL_BLOCKL_MAX); - //for quantile noise estimation + // for quantile noise estimation memset(inst->quantile, 0, sizeof(float) * HALF_ANAL_BLOCKL); for (i = 0; i < SIMULT * HALF_ANAL_BLOCKL; i++) { inst->lquantile[i] = (float)8.0; @@ -127,7 +131,8 @@ int WebRtcNs_InitCore(NSinst_t* inst, uint32_t fs) { } for (i = 0; i < SIMULT; i++) { - inst->counter[i] = (int)floor((float)(END_STARTUP_LONG * (i + 1)) / (float)SIMULT); + inst->counter[i] = + (int)floor((float)(END_STARTUP_LONG * (i + 1)) / (float)SIMULT); } inst->updates = 0; @@ -140,52 +145,63 @@ int WebRtcNs_InitCore(NSinst_t* inst, uint32_t fs) { // Set the aggressiveness: default inst->aggrMode = 0; - //initialize variables for new method - inst->priorSpeechProb = (float)0.5; //prior prob for speech/noise + // initialize variables for new method + inst->priorSpeechProb = (float)0.5; // prior prob for speech/noise for (i = 0; i < HALF_ANAL_BLOCKL; i++) { - inst->magnPrev[i] = (float)0.0; //previous mag spectrum - inst->noisePrev[i] = (float)0.0; //previous noise-spectrum - inst->logLrtTimeAvg[i] = LRT_FEATURE_THR; //smooth LR ratio (same as threshold) - inst->magnAvgPause[i] = (float)0.0; //conservative noise spectrum estimate - inst->speechProbHB[i] = (float)0.0; //for estimation of HB in second pass - inst->initMagnEst[i] = (float)0.0; //initial average mag spectrum + inst->magnPrev[i] = (float)0.0; // previous mag spectrum + inst->noisePrev[i] = (float)0.0; // previous noise-spectrum + inst->logLrtTimeAvg[i] = + LRT_FEATURE_THR; // smooth LR ratio (same as threshold) + inst->magnAvgPause[i] = (float)0.0; // conservative noise spectrum estimate + inst->speechProb[i] = (float)0.0; // for estimation of HB in second pass + inst->initMagnEst[i] = (float)0.0; // initial average mag spectrum } - //feature quantities - inst->featureData[0] = SF_FEATURE_THR; //spectral flatness (start on threshold) - inst->featureData[1] = (float)0.0; //spectral entropy: not used in this version - inst->featureData[2] = (float)0.0; //spectral variance: not used in this version - inst->featureData[3] = LRT_FEATURE_THR; //average lrt factor (start on threshold) - inst->featureData[4] = SF_FEATURE_THR; //spectral template diff (start on threshold) - inst->featureData[5] = (float)0.0; //normalization for spectral-diff - inst->featureData[6] = (float)0.0; //window time-average of input magnitude spectrum - - //histogram quantities: used to estimate/update thresholds for features + // feature quantities + inst->featureData[0] = + SF_FEATURE_THR; // spectral flatness (start on threshold) + inst->featureData[1] = + (float)0.0; // spectral entropy: not used in this version + inst->featureData[2] = + (float)0.0; // spectral variance: not used in this version + inst->featureData[3] = + LRT_FEATURE_THR; // average lrt factor (start on threshold) + inst->featureData[4] = + SF_FEATURE_THR; // spectral template diff (start on threshold) + inst->featureData[5] = (float)0.0; // normalization for spectral-diff + inst->featureData[6] = + (float)0.0; // window time-average of input magnitude spectrum + + // histogram quantities: used to estimate/update thresholds for features for (i = 0; i < HIST_PAR_EST; i++) { inst->histLrt[i] = 0; inst->histSpecFlat[i] = 0; inst->histSpecDiff[i] = 0; } - inst->blockInd = -1; //frame counter - inst->priorModelPars[0] = LRT_FEATURE_THR; //default threshold for lrt feature - inst->priorModelPars[1] = (float)0.5; //threshold for spectral flatness: + inst->blockInd = -1; // frame counter + inst->priorModelPars[0] = + LRT_FEATURE_THR; // default threshold for lrt feature + inst->priorModelPars[1] = (float)0.5; // threshold for spectral flatness: // determined on-line - inst->priorModelPars[2] = (float)1.0; //sgn_map par for spectral measure: + inst->priorModelPars[2] = (float)1.0; // sgn_map par for spectral measure: // 1 for flatness measure - inst->priorModelPars[3] = (float)0.5; //threshold for template-difference feature: + inst->priorModelPars[3] = + (float)0.5; // threshold for template-difference feature: // determined on-line - inst->priorModelPars[4] = (float)1.0; //default weighting parameter for lrt feature - inst->priorModelPars[5] = (float)0.0; //default weighting parameter for + inst->priorModelPars[4] = + (float)1.0; // default weighting parameter for lrt feature + inst->priorModelPars[5] = (float)0.0; // default weighting parameter for // spectral flatness feature - inst->priorModelPars[6] = (float)0.0; //default weighting parameter for + inst->priorModelPars[6] = (float)0.0; // default weighting parameter for // spectral difference feature - inst->modelUpdatePars[0] = 2; //update flag for parameters: + inst->modelUpdatePars[0] = 2; // update flag for parameters: // 0 no update, 1=update once, 2=update every window - inst->modelUpdatePars[1] = 500; //window for update - inst->modelUpdatePars[2] = 0; //counter for update of conservative noise spectrum - //counter if the feature thresholds are updated during the sequence + inst->modelUpdatePars[1] = 500; // window for update + inst->modelUpdatePars[2] = + 0; // counter for update of conservative noise spectrum + // counter if the feature thresholds are updated during the sequence inst->modelUpdatePars[3] = inst->modelUpdatePars[1]; inst->signalEnergy = 0.0; @@ -194,12 +210,11 @@ int WebRtcNs_InitCore(NSinst_t* inst, uint32_t fs) { inst->pinkNoiseNumerator = 0.0; inst->pinkNoiseExp = 0.0; - WebRtcNs_set_feature_extraction_parameters(inst); // Set feature configuration + WebRtcNs_set_feature_extraction_parameters(inst); - //default mode + // default mode WebRtcNs_set_policy_core(inst, 0); - memset(inst->outBuf, 0, sizeof(float) * 3 * BLOCKL_MAX); inst->initFlag = 1; @@ -218,17 +233,17 @@ int WebRtcNs_set_policy_core(NSinst_t* inst, int mode) { inst->denoiseBound = (float)0.5; inst->gainmap = 0; } else if (mode == 1) { - //inst->overdrive = (float)1.25; + // inst->overdrive = (float)1.25; inst->overdrive = (float)1.0; inst->denoiseBound = (float)0.25; inst->gainmap = 1; } else if (mode == 2) { - //inst->overdrive = (float)1.25; + // inst->overdrive = (float)1.25; inst->overdrive = (float)1.1; inst->denoiseBound = (float)0.125; inst->gainmap = 1; } else if (mode == 3) { - //inst->overdrive = (float)1.30; + // inst->overdrive = (float)1.30; inst->overdrive = (float)1.25; inst->denoiseBound = (float)0.09; inst->gainmap = 1; @@ -264,17 +279,19 @@ void WebRtcNs_NoiseEstimation(NSinst_t* inst, float* magn, float* noise) { // update log quantile estimate if (lmagn[i] > inst->lquantile[offset + i]) { - inst->lquantile[offset + i] += QUANTILE * delta - / (float)(inst->counter[s] + 1); + inst->lquantile[offset + i] += + QUANTILE * delta / (float)(inst->counter[s] + 1); } else { - inst->lquantile[offset + i] -= ((float)1.0 - QUANTILE) * delta - / (float)(inst->counter[s] + 1); + inst->lquantile[offset + i] -= + ((float)1.0 - QUANTILE) * delta / (float)(inst->counter[s] + 1); } // update density estimate if (fabs(lmagn[i] - inst->lquantile[offset + i]) < WIDTH) { - inst->density[offset + i] = ((float)inst->counter[s] * inst->density[offset - + i] + (float)1.0 / ((float)2.0 * WIDTH)) / (float)(inst->counter[s] + 1); + inst->density[offset + i] = + ((float)inst->counter[s] * inst->density[offset + i] + + (float)1.0 / ((float)2.0 * WIDTH)) / + (float)(inst->counter[s] + 1); } } // end loop over magnitude spectrum @@ -304,57 +321,66 @@ void WebRtcNs_NoiseEstimation(NSinst_t* inst, float* magn, float* noise) { } // Extract thresholds for feature parameters -// histograms are computed over some window_size (given by inst->modelUpdatePars[1]) +// histograms are computed over some window_size (given by +// inst->modelUpdatePars[1]) // thresholds and weights are extracted every window -// flag 0 means update histogram only, flag 1 means compute the thresholds/weights +// flag 0 means update histogram only, flag 1 means compute the +// thresholds/weights // threshold and weights are returned in: inst->priorModelPars void WebRtcNs_FeatureParameterExtraction(NSinst_t* inst, int flag) { int i, useFeatureSpecFlat, useFeatureSpecDiff, numHistLrt; int maxPeak1, maxPeak2; - int weightPeak1SpecFlat, weightPeak2SpecFlat, weightPeak1SpecDiff, weightPeak2SpecDiff; + int weightPeak1SpecFlat, weightPeak2SpecFlat, weightPeak1SpecDiff, + weightPeak2SpecDiff; float binMid, featureSum; float posPeak1SpecFlat, posPeak2SpecFlat, posPeak1SpecDiff, posPeak2SpecDiff; float fluctLrt, avgHistLrt, avgSquareHistLrt, avgHistLrtCompl; - //3 features: lrt, flatness, difference - //lrt_feature = inst->featureData[3]; - //flat_feature = inst->featureData[0]; - //diff_feature = inst->featureData[4]; + // 3 features: lrt, flatness, difference + // lrt_feature = inst->featureData[3]; + // flat_feature = inst->featureData[0]; + // diff_feature = inst->featureData[4]; - //update histograms + // update histograms if (flag == 0) { // LRT - if ((inst->featureData[3] < HIST_PAR_EST * inst->featureExtractionParams.binSizeLrt) - && (inst->featureData[3] >= 0.0)) { - i = (int)(inst->featureData[3] / inst->featureExtractionParams.binSizeLrt); + if ((inst->featureData[3] < + HIST_PAR_EST * inst->featureExtractionParams.binSizeLrt) && + (inst->featureData[3] >= 0.0)) { + i = (int)(inst->featureData[3] / + inst->featureExtractionParams.binSizeLrt); inst->histLrt[i]++; } // Spectral flatness - if ((inst->featureData[0] < HIST_PAR_EST - * inst->featureExtractionParams.binSizeSpecFlat) - && (inst->featureData[0] >= 0.0)) { - i = (int)(inst->featureData[0] / inst->featureExtractionParams.binSizeSpecFlat); + if ((inst->featureData[0] < + HIST_PAR_EST * inst->featureExtractionParams.binSizeSpecFlat) && + (inst->featureData[0] >= 0.0)) { + i = (int)(inst->featureData[0] / + inst->featureExtractionParams.binSizeSpecFlat); inst->histSpecFlat[i]++; } // Spectral difference - if ((inst->featureData[4] < HIST_PAR_EST - * inst->featureExtractionParams.binSizeSpecDiff) - && (inst->featureData[4] >= 0.0)) { - i = (int)(inst->featureData[4] / inst->featureExtractionParams.binSizeSpecDiff); + if ((inst->featureData[4] < + HIST_PAR_EST * inst->featureExtractionParams.binSizeSpecDiff) && + (inst->featureData[4] >= 0.0)) { + i = (int)(inst->featureData[4] / + inst->featureExtractionParams.binSizeSpecDiff); inst->histSpecDiff[i]++; } } // extract parameters for speech/noise probability if (flag == 1) { - //lrt feature: compute the average over inst->featureExtractionParams.rangeAvgHistLrt + // lrt feature: compute the average over + // inst->featureExtractionParams.rangeAvgHistLrt avgHistLrt = 0.0; avgHistLrtCompl = 0.0; avgSquareHistLrt = 0.0; numHistLrt = 0; for (i = 0; i < HIST_PAR_EST; i++) { - binMid = ((float)i + (float)0.5) * inst->featureExtractionParams.binSizeLrt; + binMid = + ((float)i + (float)0.5) * inst->featureExtractionParams.binSizeLrt; if (binMid <= inst->featureExtractionParams.rangeAvgHistLrt) { avgHistLrt += inst->histLrt[i] * binMid; numHistLrt += inst->histLrt[i]; @@ -370,11 +396,11 @@ void WebRtcNs_FeatureParameterExtraction(NSinst_t* inst, int flag) { fluctLrt = avgSquareHistLrt - avgHistLrt * avgHistLrtCompl; // get threshold for lrt feature: if (fluctLrt < inst->featureExtractionParams.thresFluctLrt) { - //very low fluct, so likely noise + // very low fluct, so likely noise inst->priorModelPars[0] = inst->featureExtractionParams.maxLrt; } else { - inst->priorModelPars[0] = inst->featureExtractionParams.factor1ModelPars - * avgHistLrt; + inst->priorModelPars[0] = + inst->featureExtractionParams.factor1ModelPars * avgHistLrt; // check if value is within min/max range if (inst->priorModelPars[0] < inst->featureExtractionParams.minLrt) { inst->priorModelPars[0] = inst->featureExtractionParams.minLrt; @@ -385,8 +411,8 @@ void WebRtcNs_FeatureParameterExtraction(NSinst_t* inst, int flag) { } // done with lrt feature - // - // for spectral flatness and spectral difference: compute the main peaks of histogram + // for spectral flatness and spectral difference: compute the main peaks of + // histogram maxPeak1 = 0; maxPeak2 = 0; posPeak1SpecFlat = 0.0; @@ -396,7 +422,8 @@ void WebRtcNs_FeatureParameterExtraction(NSinst_t* inst, int flag) { // peaks for flatness for (i = 0; i < HIST_PAR_EST; i++) { - binMid = ((float)i + (float)0.5) * inst->featureExtractionParams.binSizeSpecFlat; + binMid = ((float)i + (float)0.5) * + inst->featureExtractionParams.binSizeSpecFlat; if (inst->histSpecFlat[i] > maxPeak1) { // Found new "first" peak maxPeak2 = maxPeak1; @@ -414,7 +441,7 @@ void WebRtcNs_FeatureParameterExtraction(NSinst_t* inst, int flag) { } } - //compute two peaks for spectral difference + // compute two peaks for spectral difference maxPeak1 = 0; maxPeak2 = 0; posPeak1SpecDiff = 0.0; @@ -423,7 +450,8 @@ void WebRtcNs_FeatureParameterExtraction(NSinst_t* inst, int flag) { weightPeak2SpecDiff = 0; // peaks for spectral difference for (i = 0; i < HIST_PAR_EST; i++) { - binMid = ((float)i + (float)0.5) * inst->featureExtractionParams.binSizeSpecDiff; + binMid = ((float)i + (float)0.5) * + inst->featureExtractionParams.binSizeSpecDiff; if (inst->histSpecDiff[i] > maxPeak1) { // Found new "first" peak maxPeak2 = maxPeak1; @@ -444,25 +472,26 @@ void WebRtcNs_FeatureParameterExtraction(NSinst_t* inst, int flag) { // for spectrum flatness feature useFeatureSpecFlat = 1; // merge the two peaks if they are close - if ((fabs(posPeak2SpecFlat - posPeak1SpecFlat) - < inst->featureExtractionParams.limitPeakSpacingSpecFlat) - && (weightPeak2SpecFlat - > inst->featureExtractionParams.limitPeakWeightsSpecFlat - * weightPeak1SpecFlat)) { + if ((fabs(posPeak2SpecFlat - posPeak1SpecFlat) < + inst->featureExtractionParams.limitPeakSpacingSpecFlat) && + (weightPeak2SpecFlat > + inst->featureExtractionParams.limitPeakWeightsSpecFlat * + weightPeak1SpecFlat)) { weightPeak1SpecFlat += weightPeak2SpecFlat; posPeak1SpecFlat = (float)0.5 * (posPeak1SpecFlat + posPeak2SpecFlat); } - //reject if weight of peaks is not large enough, or peak value too small - if (weightPeak1SpecFlat < inst->featureExtractionParams.thresWeightSpecFlat - || posPeak1SpecFlat < inst->featureExtractionParams.thresPosSpecFlat) { + // reject if weight of peaks is not large enough, or peak value too small + if (weightPeak1SpecFlat < + inst->featureExtractionParams.thresWeightSpecFlat || + posPeak1SpecFlat < inst->featureExtractionParams.thresPosSpecFlat) { useFeatureSpecFlat = 0; } // if selected, get the threshold if (useFeatureSpecFlat == 1) { // compute the threshold - inst->priorModelPars[1] = inst->featureExtractionParams.factor2ModelPars - * posPeak1SpecFlat; - //check if value is within min/max range + inst->priorModelPars[1] = + inst->featureExtractionParams.factor2ModelPars * posPeak1SpecFlat; + // check if value is within min/max range if (inst->priorModelPars[1] < inst->featureExtractionParams.minSpecFlat) { inst->priorModelPars[1] = inst->featureExtractionParams.minSpecFlat; } @@ -475,22 +504,23 @@ void WebRtcNs_FeatureParameterExtraction(NSinst_t* inst, int flag) { // for template feature useFeatureSpecDiff = 1; // merge the two peaks if they are close - if ((fabs(posPeak2SpecDiff - posPeak1SpecDiff) - < inst->featureExtractionParams.limitPeakSpacingSpecDiff) - && (weightPeak2SpecDiff - > inst->featureExtractionParams.limitPeakWeightsSpecDiff - * weightPeak1SpecDiff)) { + if ((fabs(posPeak2SpecDiff - posPeak1SpecDiff) < + inst->featureExtractionParams.limitPeakSpacingSpecDiff) && + (weightPeak2SpecDiff > + inst->featureExtractionParams.limitPeakWeightsSpecDiff * + weightPeak1SpecDiff)) { weightPeak1SpecDiff += weightPeak2SpecDiff; posPeak1SpecDiff = (float)0.5 * (posPeak1SpecDiff + posPeak2SpecDiff); } // get the threshold value - inst->priorModelPars[3] = inst->featureExtractionParams.factor1ModelPars - * posPeak1SpecDiff; - //reject if weight of peaks is not large enough - if (weightPeak1SpecDiff < inst->featureExtractionParams.thresWeightSpecDiff) { + inst->priorModelPars[3] = + inst->featureExtractionParams.factor1ModelPars * posPeak1SpecDiff; + // reject if weight of peaks is not large enough + if (weightPeak1SpecDiff < + inst->featureExtractionParams.thresWeightSpecDiff) { useFeatureSpecDiff = 0; } - //check if value is within min/max range + // check if value is within min/max range if (inst->priorModelPars[3] < inst->featureExtractionParams.minSpecDiff) { inst->priorModelPars[3] = inst->featureExtractionParams.minSpecDiff; } @@ -530,7 +560,7 @@ void WebRtcNs_FeatureParameterExtraction(NSinst_t* inst, int flag) { // spectral flatness is returned in inst->featureData[0] void WebRtcNs_ComputeSpectralFlatness(NSinst_t* inst, float* magnIn) { int i; - int shiftLP = 1; //option to remove first bin(s) from spectral measures + int shiftLP = 1; // option to remove first bin(s) from spectral measures float avgSpectralFlatnessNum, avgSpectralFlatnessDen, spectralTmp; // comute spectral measures @@ -540,7 +570,8 @@ void WebRtcNs_ComputeSpectralFlatness(NSinst_t* inst, float* magnIn) { for (i = 0; i < shiftLP; i++) { avgSpectralFlatnessDen -= magnIn[i]; } - // compute log of ratio of the geometric to arithmetic mean: check for log(0) case + // compute log of ratio of the geometric to arithmetic mean: check for log(0) + // case for (i = shiftLP; i < inst->magnLen; i++) { if (magnIn[i] > 0.0) { avgSpectralFlatnessNum += (float)log(magnIn[i]); @@ -549,24 +580,26 @@ void WebRtcNs_ComputeSpectralFlatness(NSinst_t* inst, float* magnIn) { return; } } - //normalize + // normalize avgSpectralFlatnessDen = avgSpectralFlatnessDen / inst->magnLen; avgSpectralFlatnessNum = avgSpectralFlatnessNum / inst->magnLen; - //ratio and inverse log: check for case of log(0) + // ratio and inverse log: check for case of log(0) spectralTmp = (float)exp(avgSpectralFlatnessNum) / avgSpectralFlatnessDen; - //time-avg update of spectral flatness feature + // time-avg update of spectral flatness feature inst->featureData[0] += SPECT_FL_TAVG * (spectralTmp - inst->featureData[0]); // done with flatness feature } -// Compute the difference measure between input spectrum and a template/learned noise spectrum +// Compute the difference measure between input spectrum and a template/learned +// noise spectrum // magnIn is the input spectrum // the reference/template spectrum is inst->magnAvgPause[i] // returns (normalized) spectral difference in inst->featureData[4] void WebRtcNs_ComputeSpectralDifference(NSinst_t* inst, float* magnIn) { - // avgDiffNormMagn = var(magnIn) - cov(magnIn, magnAvgPause)^2 / var(magnAvgPause) + // avgDiffNormMagn = var(magnIn) - cov(magnIn, magnAvgPause)^2 / + // var(magnAvgPause) int i; float avgPause, avgMagn, covMagnPause, varPause, varMagn, avgDiffNormMagn; @@ -574,7 +607,7 @@ void WebRtcNs_ComputeSpectralDifference(NSinst_t* inst, float* magnIn) { avgMagn = inst->sumMagn; // compute average quantities for (i = 0; i < inst->magnLen; i++) { - //conservative smooth noise spectrum from pause frames + // conservative smooth noise spectrum from pause frames avgPause += inst->magnAvgPause[i]; } avgPause = avgPause / ((float)inst->magnLen); @@ -586,7 +619,8 @@ void WebRtcNs_ComputeSpectralDifference(NSinst_t* inst, float* magnIn) { // compute variance and covariance quantities for (i = 0; i < inst->magnLen; i++) { covMagnPause += (magnIn[i] - avgMagn) * (inst->magnAvgPause[i] - avgPause); - varPause += (inst->magnAvgPause[i] - avgPause) * (inst->magnAvgPause[i] - avgPause); + varPause += + (inst->magnAvgPause[i] - avgPause) * (inst->magnAvgPause[i] - avgPause); varMagn += (magnIn[i] - avgMagn) * (magnIn[i] - avgMagn); } covMagnPause = covMagnPause / ((float)inst->magnLen); @@ -595,19 +629,24 @@ void WebRtcNs_ComputeSpectralDifference(NSinst_t* inst, float* magnIn) { // update of average magnitude spectrum inst->featureData[6] += inst->signalEnergy; - avgDiffNormMagn = varMagn - (covMagnPause * covMagnPause) / (varPause + (float)0.0001); + avgDiffNormMagn = + varMagn - (covMagnPause * covMagnPause) / (varPause + (float)0.0001); // normalize and compute time-avg update of difference feature - avgDiffNormMagn = (float)(avgDiffNormMagn / (inst->featureData[5] + (float)0.0001)); - inst->featureData[4] += SPECT_DIFF_TAVG * (avgDiffNormMagn - inst->featureData[4]); + avgDiffNormMagn = + (float)(avgDiffNormMagn / (inst->featureData[5] + (float)0.0001)); + inst->featureData[4] += + SPECT_DIFF_TAVG * (avgDiffNormMagn - inst->featureData[4]); } // Compute speech/noise probability // speech/noise probability is returned in: probSpeechFinal -//magn is the input magnitude spectrum -//noise is the noise spectrum -//snrLocPrior is the prior snr for each freq. -//snr loc_post is the post snr for each freq. -void WebRtcNs_SpeechNoiseProb(NSinst_t* inst, float* probSpeechFinal, float* snrLocPrior, +// magn is the input magnitude spectrum +// noise is the noise spectrum +// snrLocPrior is the prior snr for each freq. +// snr loc_post is the post snr for each freq. +void WebRtcNs_SpeechNoiseProb(NSinst_t* inst, + float* probSpeechFinal, + float* snrLocPrior, float* snrLocPost) { int i, sgnMap; float invLrt, gainPrior, indPrior; @@ -619,19 +658,19 @@ void WebRtcNs_SpeechNoiseProb(NSinst_t* inst, float* probSpeechFinal, float* snr float widthPrior, widthPrior0, widthPrior1, widthPrior2; widthPrior0 = WIDTH_PR_MAP; - widthPrior1 = (float)2.0 * WIDTH_PR_MAP; //width for pause region: + widthPrior1 = (float)2.0 * WIDTH_PR_MAP; // width for pause region: // lower range, so increase width in tanh map - widthPrior2 = (float)2.0 * WIDTH_PR_MAP; //for spectral-difference measure + widthPrior2 = (float)2.0 * WIDTH_PR_MAP; // for spectral-difference measure - //threshold parameters for features + // threshold parameters for features threshPrior0 = inst->priorModelPars[0]; threshPrior1 = inst->priorModelPars[1]; threshPrior2 = inst->priorModelPars[3]; - //sign for flatness feature + // sign for flatness feature sgnMap = (int)(inst->priorModelPars[2]); - //weight parameters for features + // weight parameters for features weightIndPrior0 = inst->priorModelPars[4]; weightIndPrior1 = inst->priorModelPars[5]; weightIndPrior2 = inst->priorModelPars[6]; @@ -643,8 +682,8 @@ void WebRtcNs_SpeechNoiseProb(NSinst_t* inst, float* probSpeechFinal, float* snr tmpFloat1 = (float)1.0 + (float)2.0 * snrLocPrior[i]; tmpFloat2 = (float)2.0 * snrLocPrior[i] / (tmpFloat1 + (float)0.0001); besselTmp = (snrLocPost[i] + (float)1.0) * tmpFloat2; - inst->logLrtTimeAvg[i] += LRT_TAVG * (besselTmp - (float)log(tmpFloat1) - - inst->logLrtTimeAvg[i]); + inst->logLrtTimeAvg[i] += + LRT_TAVG * (besselTmp - (float)log(tmpFloat1) - inst->logLrtTimeAvg[i]); logLrtTimeAvgKsum += inst->logLrtTimeAvg[i]; } logLrtTimeAvgKsum = (float)logLrtTimeAvgKsum / (inst->magnLen); @@ -652,23 +691,24 @@ void WebRtcNs_SpeechNoiseProb(NSinst_t* inst, float* probSpeechFinal, float* snr // done with computation of LR factor // - //compute the indicator functions + // compute the indicator functions // // average lrt feature widthPrior = widthPrior0; - //use larger width in tanh map for pause regions + // use larger width in tanh map for pause regions if (logLrtTimeAvgKsum < threshPrior0) { widthPrior = widthPrior1; } // compute indicator function: sigmoid map - indicator0 = (float)0.5 * ((float)tanh(widthPrior * - (logLrtTimeAvgKsum - threshPrior0)) + (float)1.0); + indicator0 = (float)0.5 * + ((float)tanh(widthPrior * (logLrtTimeAvgKsum - threshPrior0)) + + (float)1.0); - //spectral flatness feature + // spectral flatness feature tmpFloat1 = inst->featureData[0]; widthPrior = widthPrior0; - //use larger width in tanh map for pause regions + // use larger width in tanh map for pause regions if (sgnMap == 1 && (tmpFloat1 > threshPrior1)) { widthPrior = widthPrior1; } @@ -676,26 +716,29 @@ void WebRtcNs_SpeechNoiseProb(NSinst_t* inst, float* probSpeechFinal, float* snr widthPrior = widthPrior1; } // compute indicator function: sigmoid map - indicator1 = (float)0.5 * ((float)tanh((float)sgnMap * - widthPrior * (threshPrior1 - tmpFloat1)) + (float)1.0); + indicator1 = + (float)0.5 * + ((float)tanh((float)sgnMap * widthPrior * (threshPrior1 - tmpFloat1)) + + (float)1.0); - //for template spectrum-difference + // for template spectrum-difference tmpFloat1 = inst->featureData[4]; widthPrior = widthPrior0; - //use larger width in tanh map for pause regions + // use larger width in tanh map for pause regions if (tmpFloat1 < threshPrior2) { widthPrior = widthPrior2; } // compute indicator function: sigmoid map - indicator2 = (float)0.5 * ((float)tanh(widthPrior * (tmpFloat1 - threshPrior2)) - + (float)1.0); + indicator2 = + (float)0.5 * + ((float)tanh(widthPrior * (tmpFloat1 - threshPrior2)) + (float)1.0); - //combine the indicator function with the feature weights - indPrior = weightIndPrior0 * indicator0 + weightIndPrior1 * indicator1 + weightIndPrior2 - * indicator2; + // combine the indicator function with the feature weights + indPrior = weightIndPrior0 * indicator0 + weightIndPrior1 * indicator1 + + weightIndPrior2 * indicator2; // done with computing indicator function - //compute the prior probability + // compute the prior probability inst->priorSpeechProb += PRIOR_UPDATE * (indPrior - inst->priorSpeechProb); // make sure probabilities are within range: keep floor to 0.01 if (inst->priorSpeechProb > 1.0) { @@ -705,8 +748,9 @@ void WebRtcNs_SpeechNoiseProb(NSinst_t* inst, float* probSpeechFinal, float* snr inst->priorSpeechProb = (float)0.01; } - //final speech probability: combine prior model with LR factor: - gainPrior = ((float)1.0 - inst->priorSpeechProb) / (inst->priorSpeechProb + (float)0.0001); + // final speech probability: combine prior model with LR factor: + gainPrior = ((float)1.0 - inst->priorSpeechProb) / + (inst->priorSpeechProb + (float)0.0001); for (i = 0; i < inst->magnLen; i++) { invLrt = (float)exp(-inst->logLrtTimeAvg[i]); invLrt = (float)gainPrior * invLrt; @@ -714,135 +758,65 @@ void WebRtcNs_SpeechNoiseProb(NSinst_t* inst, float* probSpeechFinal, float* snr } } -int WebRtcNs_AnalyzeCore(NSinst_t* inst, float* inFrame) { - return 0; -} - -int WebRtcNs_ProcessCore(NSinst_t* inst, - float* speechFrame, - float* speechFrameHB, - float* outFrame, - float* outFrameHB) { - // main routine for noise reduction - - int flagHB = 0; - int i; - const int kStartBand = 5; // Skip first frequency bins during estimation. - int updateParsFlag; - - float energy1, energy2, gain, factor, factor1, factor2; - float signalEnergy, sumMagn; - float snrPrior, currentEstimateStsa; - float tmpFloat1, tmpFloat2, tmpFloat3, probSpeech, probNonSpeech; - float gammaNoiseTmp, gammaNoiseOld; - float noiseUpdateTmp, fTmp; - float fout[BLOCKL_MAX]; - float winData[ANAL_BLOCKL_MAX]; - float magn[HALF_ANAL_BLOCKL], noise[HALF_ANAL_BLOCKL]; - float theFilter[HALF_ANAL_BLOCKL], theFilterTmp[HALF_ANAL_BLOCKL]; - float snrLocPost[HALF_ANAL_BLOCKL], snrLocPrior[HALF_ANAL_BLOCKL]; - float probSpeechFinal[HALF_ANAL_BLOCKL] = { 0 }; - float previousEstimateStsa[HALF_ANAL_BLOCKL]; - float real[ANAL_BLOCKL_MAX], imag[HALF_ANAL_BLOCKL]; +int WebRtcNs_AnalyzeCore(NSinst_t* inst, float* speechFrame) { + int i; + const int kStartBand = 5; // Skip first frequency bins during estimation. + int updateParsFlag; + float energy; + float signalEnergy, sumMagn; + float tmpFloat1, tmpFloat2, tmpFloat3, probSpeech, probNonSpeech; + float gammaNoiseTmp, gammaNoiseOld; + float noiseUpdateTmp, fTmp; + float winData[ANAL_BLOCKL_MAX]; + float magn[HALF_ANAL_BLOCKL], noise[HALF_ANAL_BLOCKL]; + float snrLocPost[HALF_ANAL_BLOCKL], snrLocPrior[HALF_ANAL_BLOCKL]; + float real[ANAL_BLOCKL_MAX], imag[HALF_ANAL_BLOCKL]; // Variables during startup - float sum_log_i = 0.0; - float sum_log_i_square = 0.0; - float sum_log_magn = 0.0; - float sum_log_i_log_magn = 0.0; - float parametric_noise = 0.0; - float parametric_exp = 0.0; - float parametric_num = 0.0; - - // SWB variables - int deltaBweHB = 1; - int deltaGainHB = 1; - float decayBweHB = 1.0; - float gainMapParHB = 1.0; - float gainTimeDomainHB = 1.0; - float avgProbSpeechHB, avgProbSpeechHBTmp, avgFilterGainHB, gainModHB; + float sum_log_i = 0.0; + float sum_log_i_square = 0.0; + float sum_log_magn = 0.0; + float sum_log_i_log_magn = 0.0; + float parametric_exp = 0.0; + float parametric_num = 0.0; // Check that initiation has been done if (inst->initFlag != 1) { return (-1); } - // Check for valid pointers based on sampling rate - if (inst->fs == 32000) { - if (speechFrameHB == NULL) { - return -1; - } - flagHB = 1; - // range for averaging low band quantities for H band gain - deltaBweHB = (int)inst->magnLen / 4; - deltaGainHB = deltaBweHB; - } // updateParsFlag = inst->modelUpdatePars[0]; // // update analysis buffer for L band - memcpy(inst->dataBuf, inst->dataBuf + inst->blockLen10ms, + memcpy(inst->analyzeBuf, + inst->analyzeBuf + inst->blockLen10ms, sizeof(float) * (inst->anaLen - inst->blockLen10ms)); - memcpy(inst->dataBuf + inst->anaLen - inst->blockLen10ms, speechFrame, + memcpy(inst->analyzeBuf + inst->anaLen - inst->blockLen10ms, + speechFrame, sizeof(float) * inst->blockLen10ms); - if (flagHB == 1) { - // update analysis buffer for H band - memcpy(inst->dataBufHB, inst->dataBufHB + inst->blockLen10ms, - sizeof(float) * (inst->anaLen - inst->blockLen10ms)); - memcpy(inst->dataBufHB + inst->anaLen - inst->blockLen10ms, speechFrameHB, - sizeof(float) * inst->blockLen10ms); - } - // check if processing needed if (inst->outLen == 0) { // windowing - energy1 = 0.0; + energy = 0.0; for (i = 0; i < inst->anaLen; i++) { - winData[i] = inst->window[i] * inst->dataBuf[i]; - energy1 += winData[i] * winData[i]; + winData[i] = inst->window[i] * inst->analyzeBuf[i]; + energy += winData[i] * winData[i]; } - if (energy1 == 0.0) { - // synthesize the special case of zero input + if (energy == 0.0) { // we want to avoid updating statistics in this case: - // Updating feature statistics when we have zeros only will cause thresholds to - // move towards zero signal situations. This in turn has the effect that once the - // signal is "turned on" (non-zero values) everything will be treated as speech - // and there is no noise suppression effect. Depending on the duration of the - // inactive signal it takes a considerable amount of time for the system to learn - // what is noise and what is speech. - - // read out fully processed segment - for (i = inst->windShift; i < inst->blockLen + inst->windShift; i++) { - fout[i - inst->windShift] = inst->syntBuf[i]; - } - // update synthesis buffer - memcpy(inst->syntBuf, inst->syntBuf + inst->blockLen, - sizeof(float) * (inst->anaLen - inst->blockLen)); - memset(inst->syntBuf + inst->anaLen - inst->blockLen, 0, - sizeof(float) * inst->blockLen); - - // out buffer - inst->outLen = inst->blockLen - inst->blockLen10ms; - if (inst->blockLen > inst->blockLen10ms) { - for (i = 0; i < inst->outLen; i++) { - inst->outBuf[i] = fout[i + inst->blockLen10ms]; - } - } - for (i = 0; i < inst->blockLen10ms; ++i) - outFrame[i] = WEBRTC_SPL_SAT( - WEBRTC_SPL_WORD16_MAX, fout[i], WEBRTC_SPL_WORD16_MIN); - - // for time-domain gain of HB - if (flagHB == 1) - for (i = 0; i < inst->blockLen10ms; ++i) - outFrameHB[i] = WEBRTC_SPL_SAT( - WEBRTC_SPL_WORD16_MAX, inst->dataBufHB[i], WEBRTC_SPL_WORD16_MIN); - + // Updating feature statistics when we have zeros only will cause + // thresholds to move towards zero signal situations. This in turn has the + // effect that once the signal is "turned on" (non-zero values) everything + // will be treated as speech and there is no noise suppression effect. + // Depending on the duration of the inactive signal it takes a + // considerable amount of time for the system to learn what is noise and + // what is speech. return 0; } // - inst->blockInd++; // Update the block index only when we process a block. + inst->blockInd++; // Update the block index only when we process a block. // FFT WebRtc_rdft(inst->anaLen, 1, winData, inst->ip, inst->wfft); @@ -856,8 +830,6 @@ int WebRtcNs_ProcessCore(NSinst_t* inst, (float)(real[inst->magnLen - 1] * real[inst->magnLen - 1]); sumMagn = magn[0] + magn[inst->magnLen - 1]; if (inst->blockInd < END_STARTUP_SHORT) { - inst->initMagnEst[0] += magn[0]; - inst->initMagnEst[inst->magnLen - 1] += magn[inst->magnLen - 1]; tmpFloat2 = log((float)(inst->magnLen - 1)); sum_log_i = tmpFloat2; sum_log_i_square = tmpFloat2 * tmpFloat2; @@ -875,7 +847,6 @@ int WebRtcNs_ProcessCore(NSinst_t* inst, magn[i] = ((float)sqrt(fTmp)) + 1.0f; sumMagn += magn[i]; if (inst->blockInd < END_STARTUP_SHORT) { - inst->initMagnEst[i] += magn[i]; if (i >= kStartBand) { tmpFloat2 = log((float)i); sum_log_i += tmpFloat2; @@ -890,18 +861,20 @@ int WebRtcNs_ProcessCore(NSinst_t* inst, inst->signalEnergy = signalEnergy; inst->sumMagn = sumMagn; - //compute spectral flatness on input spectrum + // compute spectral flatness on input spectrum WebRtcNs_ComputeSpectralFlatness(inst, magn); // quantile noise estimate WebRtcNs_NoiseEstimation(inst, magn, noise); - //compute simplified noise model during startup + // compute simplified noise model during startup if (inst->blockInd < END_STARTUP_SHORT) { // Estimate White noise - inst->whiteNoiseLevel += sumMagn / ((float)inst->magnLen) * inst->overdrive; + inst->whiteNoiseLevel += + sumMagn / ((float)inst->magnLen) * inst->overdrive; // Estimate Pink noise parameters tmpFloat1 = sum_log_i_square * ((float)(inst->magnLen - kStartBand)); tmpFloat1 -= (sum_log_i * sum_log_i); - tmpFloat2 = (sum_log_i_square * sum_log_magn - sum_log_i * sum_log_i_log_magn); + tmpFloat2 = + (sum_log_i_square * sum_log_magn - sum_log_i * sum_log_i_log_magn); tmpFloat3 = tmpFloat2 / tmpFloat1; // Constrain the estimated spectrum to be positive if (tmpFloat3 < 0.0f) { @@ -921,32 +894,34 @@ int WebRtcNs_ProcessCore(NSinst_t* inst, inst->pinkNoiseExp += tmpFloat3; // Calculate frequency independent parts of parametric noise estimate. - if (inst->pinkNoiseExp == 0.0f) { - // Use white noise estimate - parametric_noise = inst->whiteNoiseLevel; - } else { + if (inst->pinkNoiseExp > 0.0f) { // Use pink noise estimate - parametric_num = exp(inst->pinkNoiseNumerator / (float)(inst->blockInd + 1)); + parametric_num = + exp(inst->pinkNoiseNumerator / (float)(inst->blockInd + 1)); parametric_num *= (float)(inst->blockInd + 1); parametric_exp = inst->pinkNoiseExp / (float)(inst->blockInd + 1); - parametric_noise = parametric_num / pow((float)kStartBand, parametric_exp); } for (i = 0; i < inst->magnLen; i++) { - // Estimate the background noise using the white and pink noise parameters - if ((inst->pinkNoiseExp > 0.0f) && (i >= kStartBand)) { + // Estimate the background noise using the white and pink noise + // parameters + if (inst->pinkNoiseExp == 0.0f) { + // Use white noise estimate + inst->parametricNoise[i] = inst->whiteNoiseLevel; + } else { // Use pink noise estimate - parametric_noise = parametric_num / pow((float)i, parametric_exp); + float use_band = (float)(i < kStartBand ? kStartBand : i); + inst->parametricNoise[i] = + parametric_num / pow(use_band, parametric_exp); } - theFilterTmp[i] = (inst->initMagnEst[i] - inst->overdrive * parametric_noise); - theFilterTmp[i] /= (inst->initMagnEst[i] + (float)0.0001); // Weight quantile noise with modeled noise noise[i] *= (inst->blockInd); - tmpFloat2 = parametric_noise * (END_STARTUP_SHORT - inst->blockInd); + tmpFloat2 = + inst->parametricNoise[i] * (END_STARTUP_SHORT - inst->blockInd); noise[i] += (tmpFloat2 / (float)(inst->blockInd + 1)); noise[i] /= END_STARTUP_SHORT; } } - //compute average signal during END_STARTUP_LONG time: + // compute average signal during END_STARTUP_LONG time: // used to normalize spectral difference measure if (inst->blockInd < END_STARTUP_LONG) { inst->featureData[5] *= inst->blockInd; @@ -954,11 +929,8 @@ int WebRtcNs_ProcessCore(NSinst_t* inst, inst->featureData[5] /= (inst->blockInd + 1); } - //start processing at frames == converged+1 - // + // start processing at frames == converged+1 // STEP 1: compute prior and post snr based on quantile noise est - // - // compute DD estimate of prior SNR: needed for new method for (i = 0; i < inst->magnLen; i++) { // post snr @@ -968,23 +940,25 @@ int WebRtcNs_ProcessCore(NSinst_t* inst, } // previous post snr // previous estimate: based on previous frame with gain filter - previousEstimateStsa[i] = inst->magnPrev[i] / (inst->noisePrev[i] + (float)0.0001) - * (inst->smooth[i]); + inst->previousEstimateStsa[i] = inst->magnPrev[i] / + (inst->noisePrev[i] + (float)0.0001) * + (inst->smooth[i]); // DD estimate is sum of two terms: current estimate and previous estimate // directed decision update of snrPrior - snrLocPrior[i] = DD_PR_SNR * previousEstimateStsa[i] + ((float)1.0 - DD_PR_SNR) - * snrLocPost[i]; + snrLocPrior[i] = DD_PR_SNR * inst->previousEstimateStsa[i] + + ((float)1.0 - DD_PR_SNR) * snrLocPost[i]; // post and prior snr needed for step 2 } // end of loop over freqs - // done with step 1: dd computation of prior and post snr + // done with step 1: dd computation of prior and post snr - // - //STEP 2: compute speech/noise likelihood - // - // compute difference of input spectrum with learned/estimated noise spectrum + // STEP 2: compute speech/noise likelihood + // compute difference of input spectrum with learned/estimated noise + // spectrum WebRtcNs_ComputeSpectralDifference(inst, magn); - // compute histograms for parameter decisions (thresholds and weights for features) - // parameters are extracted once every window time (=inst->modelUpdatePars[1]) + // compute histograms for parameter decisions (thresholds and weights for + // features) + // parameters are extracted once every window time + // (=inst->modelUpdatePars[1]) if (updateParsFlag >= 1) { // counter update inst->modelUpdatePars[3]--; @@ -1002,25 +976,27 @@ int WebRtcNs_ProcessCore(NSinst_t* inst, } else { // update every window: // get normalization for spectral difference for next window estimate - inst->featureData[6] = inst->featureData[6] - / ((float)inst->modelUpdatePars[1]); - inst->featureData[5] = (float)0.5 * (inst->featureData[6] - + inst->featureData[5]); + inst->featureData[6] = + inst->featureData[6] / ((float)inst->modelUpdatePars[1]); + inst->featureData[5] = + (float)0.5 * (inst->featureData[6] + inst->featureData[5]); inst->featureData[6] = (float)0.0; } } } // compute speech/noise probability - WebRtcNs_SpeechNoiseProb(inst, probSpeechFinal, snrLocPrior, snrLocPost); + WebRtcNs_SpeechNoiseProb(inst, inst->speechProb, snrLocPrior, snrLocPost); // time-avg parameter for noise update gammaNoiseTmp = NOISE_UPDATE; for (i = 0; i < inst->magnLen; i++) { - probSpeech = probSpeechFinal[i]; + probSpeech = inst->speechProb[i]; probNonSpeech = (float)1.0 - probSpeech; // temporary noise update: // use it for speech frames if update value is less than previous - noiseUpdateTmp = gammaNoiseTmp * inst->noisePrev[i] + ((float)1.0 - gammaNoiseTmp) - * (probNonSpeech * magn[i] + probSpeech * inst->noisePrev[i]); + noiseUpdateTmp = + gammaNoiseTmp * inst->noisePrev[i] + + ((float)1.0 - gammaNoiseTmp) * + (probNonSpeech * magn[i] + probSpeech * inst->noisePrev[i]); // // time-constant based on speech/noise state gammaNoiseOld = gammaNoiseTmp; @@ -1031,16 +1007,20 @@ int WebRtcNs_ProcessCore(NSinst_t* inst, } // conservative noise update if (probSpeech < PROB_RANGE) { - inst->magnAvgPause[i] += GAMMA_PAUSE * (magn[i] - inst->magnAvgPause[i]); + inst->magnAvgPause[i] += + GAMMA_PAUSE * (magn[i] - inst->magnAvgPause[i]); } // noise update if (gammaNoiseTmp == gammaNoiseOld) { noise[i] = noiseUpdateTmp; } else { - noise[i] = gammaNoiseTmp * inst->noisePrev[i] + ((float)1.0 - gammaNoiseTmp) - * (probNonSpeech * magn[i] + probSpeech * inst->noisePrev[i]); + noise[i] = + gammaNoiseTmp * inst->noisePrev[i] + + ((float)1.0 - gammaNoiseTmp) * + (probNonSpeech * magn[i] + probSpeech * inst->noisePrev[i]); // allow for noise update downwards: - // if noise update decreases the noise, it is safe, so allow it to happen + // if noise update decreases the noise, it is safe, so allow it to + // happen if (noiseUpdateTmp < noise[i]) { noise[i] = noiseUpdateTmp; } @@ -1048,25 +1028,160 @@ int WebRtcNs_ProcessCore(NSinst_t* inst, } // end of freq loop // done with step 2: noise update - // - // STEP 3: compute dd update of prior snr and post snr based on new noise estimate - // + // keep track of noise spectrum for next frame + for (i = 0; i < inst->magnLen; i++) { + inst->noisePrev[i] = noise[i]; + } + } // end of if inst->outLen == 0 + + return 0; +} + +int WebRtcNs_ProcessCore(NSinst_t* inst, + float* speechFrame, + float* speechFrameHB, + float* outFrame, + float* outFrameHB) { + // main routine for noise reduction + int flagHB = 0; + int i; + + float energy1, energy2, gain, factor, factor1, factor2; + float snrPrior, currentEstimateStsa; + float tmpFloat1, tmpFloat2; + float fTmp; + float fout[BLOCKL_MAX]; + float winData[ANAL_BLOCKL_MAX]; + float magn[HALF_ANAL_BLOCKL]; + float theFilter[HALF_ANAL_BLOCKL], theFilterTmp[HALF_ANAL_BLOCKL]; + float real[ANAL_BLOCKL_MAX], imag[HALF_ANAL_BLOCKL]; + + // SWB variables + int deltaBweHB = 1; + int deltaGainHB = 1; + float decayBweHB = 1.0; + float gainMapParHB = 1.0; + float gainTimeDomainHB = 1.0; + float avgProbSpeechHB, avgProbSpeechHBTmp, avgFilterGainHB, gainModHB; + + // Check that initiation has been done + if (inst->initFlag != 1) { + return (-1); + } + // Check for valid pointers based on sampling rate + if (inst->fs == 32000) { + if (speechFrameHB == NULL) { + return -1; + } + flagHB = 1; + // range for averaging low band quantities for H band gain + deltaBweHB = (int)inst->magnLen / 4; + deltaGainHB = deltaBweHB; + } + + // update analysis buffer for L band + memcpy(inst->dataBuf, + inst->dataBuf + inst->blockLen10ms, + sizeof(float) * (inst->anaLen - inst->blockLen10ms)); + memcpy(inst->dataBuf + inst->anaLen - inst->blockLen10ms, + speechFrame, + sizeof(float) * inst->blockLen10ms); + + if (flagHB == 1) { + // update analysis buffer for H band + memcpy(inst->dataBufHB, + inst->dataBufHB + inst->blockLen10ms, + sizeof(float) * (inst->anaLen - inst->blockLen10ms)); + memcpy(inst->dataBufHB + inst->anaLen - inst->blockLen10ms, + speechFrameHB, + sizeof(float) * inst->blockLen10ms); + } + + // check if processing needed + if (inst->outLen == 0) { + // windowing + energy1 = 0.0; + for (i = 0; i < inst->anaLen; i++) { + winData[i] = inst->window[i] * inst->dataBuf[i]; + energy1 += winData[i] * winData[i]; + } + if (energy1 == 0.0) { + // synthesize the special case of zero input + // read out fully processed segment + for (i = inst->windShift; i < inst->blockLen + inst->windShift; i++) { + fout[i - inst->windShift] = inst->syntBuf[i]; + } + // update synthesis buffer + memcpy(inst->syntBuf, + inst->syntBuf + inst->blockLen, + sizeof(float) * (inst->anaLen - inst->blockLen)); + memset(inst->syntBuf + inst->anaLen - inst->blockLen, + 0, + sizeof(float) * inst->blockLen); + + // out buffer + inst->outLen = inst->blockLen - inst->blockLen10ms; + if (inst->blockLen > inst->blockLen10ms) { + for (i = 0; i < inst->outLen; i++) { + inst->outBuf[i] = fout[i + inst->blockLen10ms]; + } + } + for (i = 0; i < inst->blockLen10ms; ++i) + outFrame[i] = WEBRTC_SPL_SAT( + WEBRTC_SPL_WORD16_MAX, fout[i], WEBRTC_SPL_WORD16_MIN); + + // for time-domain gain of HB + if (flagHB == 1) + for (i = 0; i < inst->blockLen10ms; ++i) + outFrameHB[i] = WEBRTC_SPL_SAT( + WEBRTC_SPL_WORD16_MAX, inst->dataBufHB[i], WEBRTC_SPL_WORD16_MIN); + + return 0; + } + + // FFT + WebRtc_rdft(inst->anaLen, 1, winData, inst->ip, inst->wfft); + + imag[0] = 0; + real[0] = winData[0]; + magn[0] = (float)(fabs(real[0]) + 1.0f); + imag[inst->magnLen - 1] = 0; + real[inst->magnLen - 1] = winData[1]; + magn[inst->magnLen - 1] = (float)(fabs(real[inst->magnLen - 1]) + 1.0f); + if (inst->blockInd < END_STARTUP_SHORT) { + inst->initMagnEst[0] += magn[0]; + inst->initMagnEst[inst->magnLen - 1] += magn[inst->magnLen - 1]; + } + for (i = 1; i < inst->magnLen - 1; i++) { + real[i] = winData[2 * i]; + imag[i] = winData[2 * i + 1]; + // magnitude spectrum + fTmp = real[i] * real[i]; + fTmp += imag[i] * imag[i]; + magn[i] = ((float)sqrt(fTmp)) + 1.0f; + if (inst->blockInd < END_STARTUP_SHORT) { + inst->initMagnEst[i] += magn[i]; + } + } + + // Compute dd update of prior snr and post snr based on new noise estimate for (i = 0; i < inst->magnLen; i++) { // post and prior snr currentEstimateStsa = (float)0.0; - if (magn[i] > noise[i]) { - currentEstimateStsa = magn[i] / (noise[i] + (float)0.0001) - (float)1.0; + if (magn[i] > inst->noisePrev[i]) { + currentEstimateStsa = + magn[i] / (inst->noisePrev[i] + (float)0.0001) - (float)1.0; } - // DD estimate is sume of two terms: current estimate and previous estimate + // DD estimate is sume of two terms: current estimate and previous + // estimate // directed decision update of snrPrior - snrPrior = DD_PR_SNR * previousEstimateStsa[i] + ((float)1.0 - DD_PR_SNR) - * currentEstimateStsa; + snrPrior = DD_PR_SNR * inst->previousEstimateStsa[i] + + ((float)1.0 - DD_PR_SNR) * currentEstimateStsa; // gain filter tmpFloat1 = inst->overdrive + snrPrior; tmpFloat2 = (float)snrPrior / tmpFloat1; theFilter[i] = (float)tmpFloat2; } // end of loop over freqs - // done with step3 for (i = 0; i < inst->magnLen; i++) { // flooring bottom @@ -1078,6 +1193,9 @@ int WebRtcNs_ProcessCore(NSinst_t* inst, theFilter[i] = 1.0; } if (inst->blockInd < END_STARTUP_SHORT) { + theFilterTmp[i] = + (inst->initMagnEst[i] - inst->overdrive * inst->parametricNoise[i]); + theFilterTmp[i] /= (inst->initMagnEst[i] + (float)0.0001); // flooring bottom if (theFilterTmp[i] < inst->denoiseBound) { theFilterTmp[i] = inst->denoiseBound; @@ -1097,9 +1215,8 @@ int WebRtcNs_ProcessCore(NSinst_t* inst, real[i] *= inst->smooth[i]; imag[i] *= inst->smooth[i]; } - // keep track of noise and magn spectrum for next frame + // keep track of magn spectrum for next frame for (i = 0; i < inst->magnLen; i++) { - inst->noisePrev[i] = noise[i]; inst->magnPrev[i] = magn[i]; } // back to time domain @@ -1112,10 +1229,10 @@ int WebRtcNs_ProcessCore(NSinst_t* inst, WebRtc_rdft(inst->anaLen, -1, winData, inst->ip, inst->wfft); for (i = 0; i < inst->anaLen; i++) { - real[i] = 2.0f * winData[i] / inst->anaLen; // fft scaling + real[i] = 2.0f * winData[i] / inst->anaLen; // fft scaling } - //scale factor: only do it after END_STARTUP_LONG time + // scale factor: only do it after END_STARTUP_LONG time factor = (float)1.0; if (inst->gainmap == 1 && inst->blockInd > END_STARTUP_LONG) { factor1 = (float)1.0; @@ -1135,17 +1252,17 @@ int WebRtcNs_ProcessCore(NSinst_t* inst, } } if (gain < B_LIM) { - //don't reduce scale too much for pause regions: + // don't reduce scale too much for pause regions: // attenuation here should be controlled by flooring if (gain <= inst->denoiseBound) { gain = inst->denoiseBound; } factor2 = (float)1.0 - (float)0.3 * (B_LIM - gain); } - //combine both scales with speech/noise prob: + // combine both scales with speech/noise prob: // note prior (priorSpeechProb) is not frequency dependent - factor = inst->priorSpeechProb * factor1 + ((float)1.0 - inst->priorSpeechProb) - * factor2; + factor = inst->priorSpeechProb * factor1 + + ((float)1.0 - inst->priorSpeechProb) * factor2; } // out of inst->gainmap==1 // synthesis @@ -1157,9 +1274,11 @@ int WebRtcNs_ProcessCore(NSinst_t* inst, fout[i - inst->windShift] = inst->syntBuf[i]; } // update synthesis buffer - memcpy(inst->syntBuf, inst->syntBuf + inst->blockLen, + memcpy(inst->syntBuf, + inst->syntBuf + inst->blockLen, sizeof(float) * (inst->anaLen - inst->blockLen)); - memset(inst->syntBuf + inst->anaLen - inst->blockLen, 0, + memset(inst->syntBuf + inst->anaLen - inst->blockLen, + 0, sizeof(float) * inst->blockLen); // out buffer @@ -1174,27 +1293,26 @@ int WebRtcNs_ProcessCore(NSinst_t* inst, for (i = 0; i < inst->blockLen10ms; i++) { fout[i] = inst->outBuf[i]; } - memcpy(inst->outBuf, inst->outBuf + inst->blockLen10ms, + memcpy(inst->outBuf, + inst->outBuf + inst->blockLen10ms, sizeof(float) * (inst->outLen - inst->blockLen10ms)); - memset(inst->outBuf + inst->outLen - inst->blockLen10ms, 0, + memset(inst->outBuf + inst->outLen - inst->blockLen10ms, + 0, sizeof(float) * inst->blockLen10ms); inst->outLen -= inst->blockLen10ms; } for (i = 0; i < inst->blockLen10ms; ++i) - outFrame[i] = WEBRTC_SPL_SAT( - WEBRTC_SPL_WORD16_MAX, fout[i], WEBRTC_SPL_WORD16_MIN); + outFrame[i] = + WEBRTC_SPL_SAT(WEBRTC_SPL_WORD16_MAX, fout[i], WEBRTC_SPL_WORD16_MIN); // for time-domain gain of HB if (flagHB == 1) { - for (i = 0; i < inst->magnLen; i++) { - inst->speechProbHB[i] = probSpeechFinal[i]; - } // average speech prob from low band // avg over second half (i.e., 4->8kHz) of freq. spectrum avgProbSpeechHB = 0.0; for (i = inst->magnLen - deltaBweHB - 1; i < inst->magnLen - 1; i++) { - avgProbSpeechHB += inst->speechProbHB[i]; + avgProbSpeechHB += inst->speechProb[i]; } avgProbSpeechHB = avgProbSpeechHB / ((float)deltaBweHB); // average filter gain from low band @@ -1206,14 +1324,16 @@ int WebRtcNs_ProcessCore(NSinst_t* inst, avgFilterGainHB = avgFilterGainHB / ((float)(deltaGainHB)); avgProbSpeechHBTmp = (float)2.0 * avgProbSpeechHB - (float)1.0; // gain based on speech prob: - gainModHB = (float)0.5 * ((float)1.0 + (float)tanh(gainMapParHB * avgProbSpeechHBTmp)); - //combine gain with low band gain + gainModHB = (float)0.5 * + ((float)1.0 + (float)tanh(gainMapParHB * avgProbSpeechHBTmp)); + // combine gain with low band gain gainTimeDomainHB = (float)0.5 * gainModHB + (float)0.5 * avgFilterGainHB; if (avgProbSpeechHB >= (float)0.5) { - gainTimeDomainHB = (float)0.25 * gainModHB + (float)0.75 * avgFilterGainHB; + gainTimeDomainHB = + (float)0.25 * gainModHB + (float)0.75 * avgFilterGainHB; } gainTimeDomainHB = gainTimeDomainHB * decayBweHB; - //make sure gain is within flooring range + // make sure gain is within flooring range // flooring bottom if (gainTimeDomainHB < inst->denoiseBound) { gainTimeDomainHB = inst->denoiseBound; @@ -1222,11 +1342,11 @@ int WebRtcNs_ProcessCore(NSinst_t* inst, if (gainTimeDomainHB > (float)1.0) { gainTimeDomainHB = 1.0; } - //apply gain + // apply gain for (i = 0; i < inst->blockLen10ms; i++) { float o = gainTimeDomainHB * inst->dataBufHB[i]; - outFrameHB[i] = WEBRTC_SPL_SAT( - WEBRTC_SPL_WORD16_MAX, o, WEBRTC_SPL_WORD16_MIN); + outFrameHB[i] = + WEBRTC_SPL_SAT(WEBRTC_SPL_WORD16_MAX, o, WEBRTC_SPL_WORD16_MIN); } } // end of H band gain computation // diff --git a/modules/audio_processing/ns/ns_core.h b/modules/audio_processing/ns/ns_core.h index 2d4b23cf..c5ca13f0 100644 --- a/modules/audio_processing/ns/ns_core.h +++ b/modules/audio_processing/ns/ns_core.h @@ -8,106 +8,108 @@ * be found in the AUTHORS file in the root of the source tree. */ -#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_NS_MAIN_SOURCE_NS_CORE_H_ -#define WEBRTC_MODULES_AUDIO_PROCESSING_NS_MAIN_SOURCE_NS_CORE_H_ +#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_NS_NS_CORE_H_ +#define WEBRTC_MODULES_AUDIO_PROCESSING_NS_NS_CORE_H_ #include "webrtc/modules/audio_processing/ns/defines.h" typedef struct NSParaExtract_t_ { - - //bin size of histogram + // bin size of histogram float binSizeLrt; float binSizeSpecFlat; float binSizeSpecDiff; - //range of histogram over which lrt threshold is computed + // range of histogram over which lrt threshold is computed float rangeAvgHistLrt; - //scale parameters: multiply dominant peaks of the histograms by scale factor to obtain - //thresholds for prior model - float factor1ModelPars; //for lrt and spectral difference - float factor2ModelPars; //for spectral_flatness: used when noise is flatter than speech - //peak limit for spectral flatness (varies between 0 and 1) + // scale parameters: multiply dominant peaks of the histograms by scale factor + // to obtain thresholds for prior model + float factor1ModelPars; // for lrt and spectral difference + float factor2ModelPars; // for spectral_flatness: used when noise is flatter + // than speech + // peak limit for spectral flatness (varies between 0 and 1) float thresPosSpecFlat; - //limit on spacing of two highest peaks in histogram: spacing determined by bin size + // limit on spacing of two highest peaks in histogram: spacing determined by + // bin size float limitPeakSpacingSpecFlat; float limitPeakSpacingSpecDiff; - //limit on relevance of second peak: + // limit on relevance of second peak: float limitPeakWeightsSpecFlat; float limitPeakWeightsSpecDiff; - //limit on fluctuation of lrt feature + // limit on fluctuation of lrt feature float thresFluctLrt; - //limit on the max and min values for the feature thresholds + // limit on the max and min values for the feature thresholds float maxLrt; float minLrt; float maxSpecFlat; float minSpecFlat; float maxSpecDiff; float minSpecDiff; - //criteria of weight of histogram peak to accept/reject feature + // criteria of weight of histogram peak to accept/reject feature int thresWeightSpecFlat; int thresWeightSpecDiff; } NSParaExtract_t; typedef struct NSinst_t_ { - - uint32_t fs; - int blockLen; - int blockLen10ms; - int windShift; - int outLen; - int anaLen; - int magnLen; - int aggrMode; - const float* window; - float dataBuf[ANAL_BLOCKL_MAX]; - float syntBuf[ANAL_BLOCKL_MAX]; - float outBuf[3 * BLOCKL_MAX]; - - int initFlag; + uint32_t fs; + int blockLen; + int blockLen10ms; + int windShift; + int outLen; + int anaLen; + int magnLen; + int aggrMode; + const float* window; + float analyzeBuf[ANAL_BLOCKL_MAX]; + float dataBuf[ANAL_BLOCKL_MAX]; + float syntBuf[ANAL_BLOCKL_MAX]; + float outBuf[3 * BLOCKL_MAX]; + + int initFlag; // parameters for quantile noise estimation - float density[SIMULT* HALF_ANAL_BLOCKL]; - float lquantile[SIMULT* HALF_ANAL_BLOCKL]; - float quantile[HALF_ANAL_BLOCKL]; - int counter[SIMULT]; - int updates; + float density[SIMULT * HALF_ANAL_BLOCKL]; + float lquantile[SIMULT * HALF_ANAL_BLOCKL]; + float quantile[HALF_ANAL_BLOCKL]; + int counter[SIMULT]; + int updates; // parameters for Wiener filter - float smooth[HALF_ANAL_BLOCKL]; - float overdrive; - float denoiseBound; - int gainmap; + float previousEstimateStsa[HALF_ANAL_BLOCKL]; + float smooth[HALF_ANAL_BLOCKL]; + float overdrive; + float denoiseBound; + int gainmap; // fft work arrays. - int ip[IP_LENGTH]; - float wfft[W_LENGTH]; + int ip[IP_LENGTH]; + float wfft[W_LENGTH]; // parameters for new method: some not needed, will reduce/cleanup later - int32_t blockInd; //frame index counter - int modelUpdatePars[4]; //parameters for updating or estimating + int32_t blockInd; // frame index counter + int modelUpdatePars[4]; // parameters for updating or estimating // thresholds/weights for prior model - float priorModelPars[7]; //parameters for prior model - float noisePrev[HALF_ANAL_BLOCKL]; //noise spectrum from previous frame - float magnPrev[HALF_ANAL_BLOCKL]; //magnitude spectrum of previous frame - float logLrtTimeAvg[HALF_ANAL_BLOCKL]; //log lrt factor with time-smoothing - float priorSpeechProb; //prior speech/noise probability - float featureData[7]; //data for features - float magnAvgPause[HALF_ANAL_BLOCKL]; //conservative noise spectrum estimate - float signalEnergy; //energy of magn - float sumMagn; //sum of magn - float whiteNoiseLevel; //initial noise estimate - float initMagnEst[HALF_ANAL_BLOCKL]; //initial magnitude spectrum estimate - float pinkNoiseNumerator; //pink noise parameter: numerator - float pinkNoiseExp; //pink noise parameter: power of freq - NSParaExtract_t featureExtractionParams; //parameters for feature extraction - //histograms for parameter estimation - int histLrt[HIST_PAR_EST]; - int histSpecFlat[HIST_PAR_EST]; - int histSpecDiff[HIST_PAR_EST]; - //quantities for high band estimate - float speechProbHB[HALF_ANAL_BLOCKL]; //final speech/noise prob: prior + LRT - float dataBufHB[ANAL_BLOCKL_MAX]; //buffering data for HB + float priorModelPars[7]; // parameters for prior model + float noisePrev[HALF_ANAL_BLOCKL]; // noise spectrum from previous frame + float magnPrev[HALF_ANAL_BLOCKL]; // magnitude spectrum of previous frame + float logLrtTimeAvg[HALF_ANAL_BLOCKL]; // log lrt factor with time-smoothing + float priorSpeechProb; // prior speech/noise probability + float featureData[7]; // data for features + float magnAvgPause[HALF_ANAL_BLOCKL]; // conservative noise spectrum estimate + float signalEnergy; // energy of magn + float sumMagn; // sum of magn + float whiteNoiseLevel; // initial noise estimate + float initMagnEst[HALF_ANAL_BLOCKL]; // initial magnitude spectrum estimate + float pinkNoiseNumerator; // pink noise parameter: numerator + float pinkNoiseExp; // pink noise parameter: power of freq + float parametricNoise[HALF_ANAL_BLOCKL]; + NSParaExtract_t featureExtractionParams; // parameters for feature extraction + // histograms for parameter estimation + int histLrt[HIST_PAR_EST]; + int histSpecFlat[HIST_PAR_EST]; + int histSpecDiff[HIST_PAR_EST]; + // quantities for high band estimate + float speechProb[HALF_ANAL_BLOCKL]; // final speech/noise prob: prior + LRT + float dataBufHB[ANAL_BLOCKL_MAX]; // buffering data for HB } NSinst_t; - #ifdef __cplusplus extern "C" { #endif @@ -136,7 +138,7 @@ int WebRtcNs_InitCore(NSinst_t* inst, uint32_t fs); * * Input: * - inst : Instance that should be initialized - * - mode : 0: Mild (6 dB), 1: Medium (10 dB), 2: Aggressive (15 dB) + * - mode : 0: Mild (6dB), 1: Medium (10dB), 2: Aggressive (15dB) * * Output: * - NS_inst : Initialized instance @@ -153,7 +155,7 @@ int WebRtcNs_set_policy_core(NSinst_t* inst, int mode); * * Input: * - inst : Instance that should be initialized - * - inFrame : Input speech frame for lower band + * - speechFrame : Input speech frame for lower band * * Output: * - inst : Updated instance @@ -161,7 +163,7 @@ int WebRtcNs_set_policy_core(NSinst_t* inst, int mode); * Return value : 0 - OK * -1 - Error */ -int WebRtcNs_AnalyzeCore(NSinst_t* inst, float* inFrame); +int WebRtcNs_AnalyzeCore(NSinst_t* inst, float* speechFrame); /**************************************************************************** * WebRtcNs_ProcessCore @@ -187,8 +189,7 @@ int WebRtcNs_ProcessCore(NSinst_t* inst, float* outFrameLow, float* outFrameHigh); - #ifdef __cplusplus } #endif -#endif // WEBRTC_MODULES_AUDIO_PROCESSING_NS_MAIN_SOURCE_NS_CORE_H_ +#endif // WEBRTC_MODULES_AUDIO_PROCESSING_NS_NS_CORE_H_ diff --git a/modules/audio_processing/test/audio_processing_unittest.cc b/modules/audio_processing/test/audio_processing_unittest.cc index d7f7cf4c..af63bde3 100644 --- a/modules/audio_processing/test/audio_processing_unittest.cc +++ b/modules/audio_processing/test/audio_processing_unittest.cc @@ -214,6 +214,7 @@ void WriteStatsMessage(const AudioProcessing::Statistic& output, void OpenFileAndWriteMessage(const std::string filename, const ::google::protobuf::MessageLite& msg) { +#if defined(WEBRTC_LINUX) && !defined(WEBRTC_ANDROID) FILE* file = fopen(filename.c_str(), "wb"); ASSERT_TRUE(file != NULL); @@ -226,6 +227,10 @@ void OpenFileAndWriteMessage(const std::string filename, ASSERT_EQ(static_cast<size_t>(size), fwrite(array.get(), sizeof(array[0]), size, file)); fclose(file); +#else + std::cout << "Warning: Writing new reference is only allowed on Linux!" + << std::endl; +#endif } std::string ResourceFilePath(std::string name, int sample_rate_hz) { @@ -1735,14 +1740,7 @@ TEST_F(ApmTest, FloatAndIntInterfacesGiveIdenticalResults) { // TODO(andrew): Add a test to process a few frames with different combinations // of enabled components. -// TODO(bjornv): Investigate if simply increasing the slack is a good way to -// make this test work on Android. When running the test on a N7 we get a {2, 6} -// difference of |has_voice_count| and |max_output_average| is up to 18 higher. -// All numbers being consistently higher on N7 compare to ref_data, evaluated on -// linux. Simply increasing the slack is one way forward. Adding an offset to -// the metrics mentioned above, but keeping the same slack, is also an -// alternative. -TEST_F(ApmTest, DISABLED_ON_ANDROID(Process)) { +TEST_F(ApmTest, Process) { GOOGLE_PROTOBUF_VERIFY_VERSION; audioproc::OutputData ref_data; @@ -1857,12 +1855,33 @@ TEST_F(ApmTest, DISABLED_ON_ANDROID(Process)) { if (!write_ref_data) { const int kIntNear = 1; + // When running the test on a N7 we get a {2, 6} difference of + // |has_voice_count| and |max_output_average| is up to 18 higher. + // All numbers being consistently higher on N7 compare to ref_data. + // TODO(bjornv): If we start getting more of these offsets on Android we + // should consider a different approach. Either using one slack for all, + // or generate a separate android reference. +#if defined(WEBRTC_ANDROID) + const int kHasVoiceCountOffset = 3; + const int kHasVoiceCountNear = 3; + const int kMaxOutputAverageOffset = 9; + const int kMaxOutputAverageNear = 9; +#else + const int kHasVoiceCountOffset = 0; + const int kHasVoiceCountNear = kIntNear; + const int kMaxOutputAverageOffset = 0; + const int kMaxOutputAverageNear = kIntNear; +#endif EXPECT_NEAR(test->has_echo_count(), has_echo_count, kIntNear); - EXPECT_NEAR(test->has_voice_count(), has_voice_count, kIntNear); + EXPECT_NEAR(test->has_voice_count(), + has_voice_count - kHasVoiceCountOffset, + kHasVoiceCountNear); EXPECT_NEAR(test->is_saturated_count(), is_saturated_count, kIntNear); EXPECT_NEAR(test->analog_level_average(), analog_level_average, kIntNear); - EXPECT_NEAR(test->max_output_average(), max_output_average, kIntNear); + EXPECT_NEAR(test->max_output_average(), + max_output_average - kMaxOutputAverageOffset, + kMaxOutputAverageNear); #if defined(WEBRTC_AUDIOPROC_FLOAT_PROFILE) audioproc::Test::EchoMetrics reference = test->echo_metrics(); diff --git a/modules/audio_processing/test/process_test.cc b/modules/audio_processing/test/process_test.cc index b3f6a775..b6d51e47 100644 --- a/modules/audio_processing/test/process_test.cc +++ b/modules/audio_processing/test/process_test.cc @@ -874,7 +874,7 @@ void void_main(int argc, char* argv[]) { near_frame.samples_per_channel_ = samples_per_channel; if (!raw_output) { - // The WAV file needs to be reset every time, because it cant change + // The WAV file needs to be reset every time, because it can't change // it's sample rate or number of channels. output_wav_file.reset(new WavFile(out_filename + ".wav", sample_rate_hz, @@ -1028,6 +1028,11 @@ void void_main(int argc, char* argv[]) { if (raw_output && !output_raw_file) { output_raw_file.reset(new RawFile(out_filename + ".pcm")); } + if (!raw_output && !output_wav_file) { + output_wav_file.reset(new WavFile(out_filename + ".wav", + sample_rate_hz, + num_capture_output_channels)); + } WriteIntData(near_frame.data_, size, output_wav_file.get(), diff --git a/modules/modules.gyp b/modules/modules.gyp index 4086299a..777523ab 100644 --- a/modules/modules.gyp +++ b/modules/modules.gyp @@ -104,7 +104,9 @@ 'sources': [ 'audio_coding/main/acm2/acm_opus_unittest.cc', 'audio_coding/main/acm2/acm_receiver_unittest.cc', + 'audio_coding/main/acm2/acm_receiver_unittest_oldapi.cc', 'audio_coding/main/acm2/audio_coding_module_unittest.cc', + 'audio_coding/main/acm2/audio_coding_module_unittest_oldapi.cc', 'audio_coding/main/acm2/call_statistics_unittest.cc', 'audio_coding/main/acm2/initial_delay_manager_unittest.cc', 'audio_coding/main/acm2/nack_unittest.cc', diff --git a/modules/pacing/include/paced_sender.h b/modules/pacing/include/paced_sender.h index ddd8e53b..14a3a3d4 100644 --- a/modules/pacing/include/paced_sender.h +++ b/modules/pacing/include/paced_sender.h @@ -14,9 +14,9 @@ #include <list> #include <set> +#include "webrtc/base/thread_annotations.h" #include "webrtc/modules/interface/module.h" #include "webrtc/system_wrappers/interface/scoped_ptr.h" -#include "webrtc/system_wrappers/interface/thread_annotations.h" #include "webrtc/typedefs.h" namespace webrtc { diff --git a/modules/rtp_rtcp/source/rtcp_sender.h b/modules/rtp_rtcp/source/rtcp_sender.h index 26c44b04..c0b8ebdb 100644 --- a/modules/rtp_rtcp/source/rtcp_sender.h +++ b/modules/rtp_rtcp/source/rtcp_sender.h @@ -15,6 +15,7 @@ #include <sstream> #include <string> +#include "webrtc/base/thread_annotations.h" #include "webrtc/modules/remote_bitrate_estimator/include/bwe_defines.h" #include "webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h" #include "webrtc/modules/rtp_rtcp/interface/receive_statistics.h" @@ -23,7 +24,6 @@ #include "webrtc/modules/rtp_rtcp/source/rtp_utility.h" #include "webrtc/modules/rtp_rtcp/source/tmmbr_help.h" #include "webrtc/system_wrappers/interface/scoped_ptr.h" -#include "webrtc/system_wrappers/interface/thread_annotations.h" #include "webrtc/typedefs.h" namespace webrtc { diff --git a/modules/rtp_rtcp/source/rtp_packet_history.h b/modules/rtp_rtcp/source/rtp_packet_history.h index 190e5057..253f6d0a 100644 --- a/modules/rtp_rtcp/source/rtp_packet_history.h +++ b/modules/rtp_rtcp/source/rtp_packet_history.h @@ -15,10 +15,10 @@ #include <vector> +#include "webrtc/base/thread_annotations.h" #include "webrtc/modules/interface/module_common_types.h" #include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h" #include "webrtc/typedefs.h" -#include "webrtc/system_wrappers/interface/thread_annotations.h" namespace webrtc { diff --git a/modules/rtp_rtcp/source/rtp_sender.h b/modules/rtp_rtcp/source/rtp_sender.h index e4d4fca3..b2f2e0c4 100644 --- a/modules/rtp_rtcp/source/rtp_sender.h +++ b/modules/rtp_rtcp/source/rtp_sender.h @@ -16,6 +16,7 @@ #include <map> +#include "webrtc/base/thread_annotations.h" #include "webrtc/common_types.h" #include "webrtc/modules/pacing/include/paced_sender.h" #include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h" @@ -25,7 +26,6 @@ #include "webrtc/modules/rtp_rtcp/source/rtp_rtcp_config.h" #include "webrtc/modules/rtp_rtcp/source/ssrc_database.h" #include "webrtc/modules/rtp_rtcp/source/video_codec_information.h" -#include "webrtc/system_wrappers/interface/thread_annotations.h" #define MAX_INIT_RTP_SEQ_NUMBER 32767 // 2^15 -1. diff --git a/modules/video_capture/BUILD.gn b/modules/video_capture/BUILD.gn index 773708bc..b9f3d2f3 100644 --- a/modules/video_capture/BUILD.gn +++ b/modules/video_capture/BUILD.gn @@ -100,7 +100,10 @@ source_set("video_capture_internal_impl") { "QTKit.framework", ] } - if (is_win) { + # winsdk_samples isn't pulled into Chromium, so it is disabled for Chromium + # builds. This is not a problem since the internal video capture + # implementation should not be used in chrome - issue 3831. + if (is_win && !build_with_chromium) { sources = [ "windows/device_info_ds.cc", "windows/device_info_ds.h", diff --git a/modules/video_coding/codecs/vp8/vp8_impl.cc b/modules/video_coding/codecs/vp8/vp8_impl.cc index 16d105d1..2a2a9d04 100644 --- a/modules/video_coding/codecs/vp8/vp8_impl.cc +++ b/modules/video_coding/codecs/vp8/vp8_impl.cc @@ -175,7 +175,7 @@ int VP8EncoderImpl::InitEncode(const VideoCodec* inst, // Creating a wrapper to the image - setting image data to NULL. Actual // pointer will be set in encode. Setting align to 1, as it is meaningless // (actual memory is not allocated). - raw_ = vpx_img_wrap(NULL, IMG_FMT_I420, codec_.width, codec_.height, + raw_ = vpx_img_wrap(NULL, VPX_IMG_FMT_I420, codec_.width, codec_.height, 1, NULL); // populate encoder configuration with default values if (vpx_codec_enc_config_default(vpx_codec_vp8_cx(), config_, 0)) { @@ -356,9 +356,9 @@ int VP8EncoderImpl::Encode(const I420VideoFrame& input_frame, } // Image in vpx_image_t format. // Input frame is const. VP8's raw frame is not defined as const. - raw_->planes[PLANE_Y] = const_cast<uint8_t*>(frame.buffer(kYPlane)); - raw_->planes[PLANE_U] = const_cast<uint8_t*>(frame.buffer(kUPlane)); - raw_->planes[PLANE_V] = const_cast<uint8_t*>(frame.buffer(kVPlane)); + raw_->planes[VPX_PLANE_Y] = const_cast<uint8_t*>(frame.buffer(kYPlane)); + raw_->planes[VPX_PLANE_U] = const_cast<uint8_t*>(frame.buffer(kUPlane)); + raw_->planes[VPX_PLANE_V] = const_cast<uint8_t*>(frame.buffer(kVPlane)); // TODO(mikhal): Stride should be set in initialization. raw_->stride[VPX_PLANE_Y] = frame.stride(kYPlane); raw_->stride[VPX_PLANE_U] = frame.stride(kUPlane); diff --git a/modules/video_coding/main/source/timing.h b/modules/video_coding/main/source/timing.h index 116639eb..bb44a93b 100644 --- a/modules/video_coding/main/source/timing.h +++ b/modules/video_coding/main/source/timing.h @@ -11,9 +11,9 @@ #ifndef WEBRTC_MODULES_VIDEO_CODING_MAIN_SOURCE_TIMING_H_ #define WEBRTC_MODULES_VIDEO_CODING_MAIN_SOURCE_TIMING_H_ +#include "webrtc/base/thread_annotations.h" #include "webrtc/modules/video_coding/main/source/codec_timer.h" #include "webrtc/system_wrappers/interface/critical_section_wrapper.h" -#include "webrtc/system_wrappers/interface/thread_annotations.h" #include "webrtc/typedefs.h" namespace webrtc { diff --git a/modules/video_coding/main/source/video_coding_impl.h b/modules/video_coding/main/source/video_coding_impl.h index 816552fe..5b3fe2eb 100644 --- a/modules/video_coding/main/source/video_coding_impl.h +++ b/modules/video_coding/main/source/video_coding_impl.h @@ -15,6 +15,7 @@ #include <vector> +#include "webrtc/base/thread_annotations.h" #include "webrtc/modules/video_coding/main/source/codec_database.h" #include "webrtc/modules/video_coding/main/source/frame_buffer.h" #include "webrtc/modules/video_coding/main/source/generic_decoder.h" @@ -25,7 +26,6 @@ #include "webrtc/modules/video_coding/main/source/timing.h" #include "webrtc/system_wrappers/interface/clock.h" #include "webrtc/system_wrappers/interface/critical_section_wrapper.h" -#include "webrtc/system_wrappers/interface/thread_annotations.h" namespace webrtc { diff --git a/modules/video_render/video_render.gypi b/modules/video_render/video_render.gypi index da51588a..5db851d9 100644 --- a/modules/video_render/video_render.gypi +++ b/modules/video_render/video_render.gypi @@ -141,23 +141,7 @@ 'windows/video_render_windows_impl.h', 'windows/video_render_direct3d9.cc', 'windows/video_render_windows_impl.cc', - ], - 'variables': { - # 'directx_sdk_path' will be overridden in the condition block - # below, but it must not be declared as empty here since gyp - # will check if the first character is '/' for some reason. - # If it's empty, we'll get an out-of-bounds error. - 'directx_sdk_path': 'will_be_overridden', - 'directx_sdk_default_path': '<(DEPTH)/third_party/directxsdk/files', - 'conditions': [ - ['"<!(python <(DEPTH)/build/dir_exists.py <(directx_sdk_default_path))"=="True"', { - 'directx_sdk_path': '<(DEPTH)/third_party/directxsdk/files', - }, { - 'directx_sdk_path': '$(DXSDK_DIR)', - }], - ], - }, - + ], 'include_dirs': [ '<(directx_sdk_path)/Include', ], diff --git a/system_wrappers/BUILD.gn b/system_wrappers/BUILD.gn index 76a0f340..262189c1 100644 --- a/system_wrappers/BUILD.gn +++ b/system_wrappers/BUILD.gn @@ -45,7 +45,6 @@ static_library("system_wrappers") { "interface/static_instance.h", "interface/stl_util.h", "interface/stringize_macros.h", - "interface/thread_annotations.h", "interface/thread_wrapper.h", "interface/tick_util.h", "interface/timestamp_extrapolator.h", diff --git a/system_wrappers/interface/critical_section_wrapper.h b/system_wrappers/interface/critical_section_wrapper.h index 4979b5c7..e93a249e 100644 --- a/system_wrappers/interface/critical_section_wrapper.h +++ b/system_wrappers/interface/critical_section_wrapper.h @@ -14,8 +14,8 @@ // If the critical section is heavily contended it may be beneficial to use // read/write locks instead. +#include "webrtc/base/thread_annotations.h" #include "webrtc/common_types.h" -#include "webrtc/system_wrappers/interface/thread_annotations.h" namespace webrtc { class LOCKABLE CriticalSectionWrapper { diff --git a/system_wrappers/interface/rw_lock_wrapper.h b/system_wrappers/interface/rw_lock_wrapper.h index 91126e5d..dbe6d6c7 100644 --- a/system_wrappers/interface/rw_lock_wrapper.h +++ b/system_wrappers/interface/rw_lock_wrapper.h @@ -11,7 +11,7 @@ #ifndef WEBRTC_SYSTEM_WRAPPERS_INTERFACE_RW_LOCK_WRAPPER_H_ #define WEBRTC_SYSTEM_WRAPPERS_INTERFACE_RW_LOCK_WRAPPER_H_ -#include "webrtc/system_wrappers/interface/thread_annotations.h" +#include "webrtc/base/thread_annotations.h" // Note, Windows pre-Vista version of RW locks are not supported natively. For // these OSs regular critical sections have been used to approximate RW lock diff --git a/system_wrappers/source/system_wrappers.gyp b/system_wrappers/source/system_wrappers.gyp index 82addeb5..e6c4216d 100644 --- a/system_wrappers/source/system_wrappers.gyp +++ b/system_wrappers/source/system_wrappers.gyp @@ -54,7 +54,6 @@ '../interface/static_instance.h', '../interface/stl_util.h', '../interface/stringize_macros.h', - '../interface/thread_annotations.h', '../interface/thread_wrapper.h', '../interface/tick_util.h', '../interface/timestamp_extrapolator.h', diff --git a/test/call_test.cc b/test/call_test.cc index ccad7551..d49f6e3f 100644 --- a/test/call_test.cc +++ b/test/call_test.cc @@ -16,7 +16,6 @@ namespace test { CallTest::CallTest() : clock_(Clock::GetRealTimeClock()), - encoder_settings_(NULL), send_stream_(NULL), fake_encoder_(clock_) { } @@ -40,8 +39,7 @@ void CallTest::RunBaseTest(BaseTest* test) { if (test->ShouldCreateReceivers()) { CreateMatchingReceiveConfigs(); } - encoder_settings_ = test->GetEncoderSettings(); - test->ModifyConfigs(&send_config_, &receive_configs_, &video_streams_); + test->ModifyConfigs(&send_config_, &receive_configs_, &encoder_config_); CreateStreams(); test->OnStreamsCreated(send_stream_, receive_streams_); @@ -92,7 +90,7 @@ void CallTest::CreateSendConfig(size_t num_streams) { send_config_.encoder_settings.encoder = &fake_encoder_; send_config_.encoder_settings.payload_name = "FAKE"; send_config_.encoder_settings.payload_type = kFakeSendPayloadType; - video_streams_ = test::CreateVideoStreams(num_streams); + encoder_config_.streams = test::CreateVideoStreams(num_streams); for (size_t i = 0; i < num_streams; ++i) send_config_.rtp.ssrcs.push_back(kSendSsrcs[i]); } @@ -123,7 +121,7 @@ void CallTest::CreateMatchingReceiveConfigs() { } void CallTest::CreateFrameGeneratorCapturer() { - VideoStream stream = video_streams_.back(); + VideoStream stream = encoder_config_.streams.back(); frame_generator_capturer_.reset( test::FrameGeneratorCapturer::Create(send_stream_->Input(), stream.width, @@ -135,8 +133,8 @@ void CallTest::CreateStreams() { assert(send_stream_ == NULL); assert(receive_streams_.empty()); - send_stream_ = sender_call_->CreateVideoSendStream( - send_config_, video_streams_, encoder_settings_); + send_stream_ = + sender_call_->CreateVideoSendStream(send_config_, encoder_config_); for (size_t i = 0; i < receive_configs_.size(); ++i) { receive_streams_.push_back( @@ -193,14 +191,10 @@ size_t BaseTest::GetNumStreams() const { return 1; } -const void* BaseTest::GetEncoderSettings() { - return NULL; -} - void BaseTest::ModifyConfigs( VideoSendStream::Config* send_config, std::vector<VideoReceiveStream::Config>* receive_configs, - std::vector<VideoStream>* video_streams) { + VideoEncoderConfig* encoder_config) { } void BaseTest::OnStreamsCreated( diff --git a/test/call_test.h b/test/call_test.h index a03c3bbe..695fb2ac 100644 --- a/test/call_test.h +++ b/test/call_test.h @@ -65,8 +65,7 @@ class CallTest : public ::testing::Test { scoped_ptr<Call> sender_call_; VideoSendStream::Config send_config_; - std::vector<VideoStream> video_streams_; - const void* encoder_settings_; + VideoEncoderConfig encoder_config_; VideoSendStream* send_stream_; scoped_ptr<Call> receiver_call_; @@ -93,11 +92,10 @@ class BaseTest : public RtpRtcpObserver { virtual Call::Config GetReceiverCallConfig(); virtual void OnCallsCreated(Call* sender_call, Call* receiver_call); - virtual const void* GetEncoderSettings(); virtual void ModifyConfigs( VideoSendStream::Config* send_config, std::vector<VideoReceiveStream::Config>* receive_configs, - std::vector<VideoStream>* video_streams); + VideoEncoderConfig* encoder_config); virtual void OnStreamsCreated( VideoSendStream* send_stream, const std::vector<VideoReceiveStream*>& receive_streams); diff --git a/test/webrtc_test_common.gyp b/test/webrtc_test_common.gyp index 71b875d1..d422a4b0 100644 --- a/test/webrtc_test_common.gyp +++ b/test/webrtc_test_common.gyp @@ -104,22 +104,6 @@ 'sources!': [ 'null_platform_renderer.cc', ], - 'variables': { - # 'directx_sdk_path' will be overridden in the condition block - # below, but it must not be declared as empty here since gyp - # will check if the first character is '/' for some reason. - # If it's empty, we'll get an out-of-bounds error. - 'directx_sdk_path': 'will_be_overridden', - 'directx_sdk_default_path': '<(DEPTH)/third_party/directxsdk/files', - 'conditions': [ - ['"<!(python <(DEPTH)/build/dir_exists.py <(directx_sdk_default_path))"=="True"', { - 'directx_sdk_path': '<(DEPTH)/third_party/directxsdk/files', - }, { - 'directx_sdk_path': '$(DXSDK_DIR)', - }], - ], - }, - 'include_dirs': [ '<(directx_sdk_path)/Include', ], diff --git a/tools/rtcbot/bot/browser/bot.js b/tools/rtcbot/bot/browser/bot.js index 130d006e..88398021 100644 --- a/tools/rtcbot/bot/browser/bot.js +++ b/tools/rtcbot/bot/browser/bot.js @@ -5,23 +5,80 @@ // tree. An additional intellectual property rights grant can be found // in the file PATENTS. All contributing project authors may // be found in the AUTHORS file in the root of the source tree. +// +var localStreams = []; +var remoteStreams = []; + +function ping(callback) { + callback("pong"); +} + +function getUserMedia(constraints, onSuccessCallback, onFailCallback){ + console.log("Getting user media."); + navigator.webkitGetUserMedia(constraints, + onSuccessCallbackWraper, onFailCallback); + + function onSuccessCallbackWraper(stream) { + console.log("GetUserMedia success."); + localStreams[stream.id] = stream; + onSuccessCallback(stream); + } +} + +function createPeerConnection(doneCallback, failCallback) { + console.log("Creating peer connection"); + var obj = {}; + var pc = new webkitRTCPeerConnection(null); -var botExposedApi = { - ping: function (callback) { - callback("pong"); - }, - - createPeerConnection: function (doneCallback) { - console.log("Creating peer connection"); - var pc = new webkitRTCPeerConnection(null); - var obj = {}; - expose(obj, pc, "close"); - expose(obj, pc, "createOffer"); - expose(obj, pc, "createAnswer"); - expose(obj, pc, "setRemoteDescription", { 0: RTCSessionDescription }); - expose(obj, pc, "setLocalDescription", { 0: RTCSessionDescription }); - doneCallback(obj); - }, + expose(obj, pc, "close"); + expose(obj, pc, "createOffer"); + expose(obj, pc, "createAnswer"); + expose(obj, pc, "addEventListener"); + expose(obj, pc, "addIceCandidate", { 0: RTCIceCandidate}); + expose(obj, pc, "setRemoteDescription", { 0: RTCSessionDescription }); + expose(obj, pc, "setLocalDescription", { 0: RTCSessionDescription }); + + obj.addStream = function(stream) { + console.log("Adding local stream."); + var tempStream = localStreams[stream.id]; + if (!tempStream) { + console.log("Undefined stream!"); + return; + } + pc.addStream(tempStream); + }; + + pc.addEventListener('addstream', function(event) { + remoteStreams[event.stream.id] = event.stream; + }); + + doneCallback(obj); +} + +function showStream(streamId, autoplay, muted) { + var stream = getStreamFromIdentifier_(streamId); + var video = document.createElement('video'); + video.autoplay = autoplay; + video.muted = muted; + document.body.appendChild(video); + video.src = URL.createObjectURL(stream); + console.log("Stream " + stream.id + " attached to video element"); }; -connectToServer(botExposedApi); +function getStreamFromIdentifier_(id) { + var tempStream = localStreams[id]; + if (tempStream) + return tempStream; + tempStream = remoteStreams[id]; + if (tempStream) + return tempStream; + console.log(id + " is not id for stream."); + return null; +} + +connectToServer({ + ping: ping, + getUserMedia: getUserMedia, + createPeerConnection: createPeerConnection, + showStream: showStream, +}); diff --git a/tools/rtcbot/test.js b/tools/rtcbot/test.js index c1db4fc0..36675ccb 100644 --- a/tools/rtcbot/test.js +++ b/tools/rtcbot/test.js @@ -17,10 +17,10 @@ var vm = require('vm'); var BotManager = require('./botmanager.js'); function Test(botType) { - // Make the test fail if not completed in 3 seconds. + // TODO(houssainy) set the time out. this.timeout_ = setTimeout( this.fail.bind(this, "Test timeout!"), - 5000); + 10000); this.botType_ = botType; } @@ -76,10 +76,10 @@ Test.prototype = { }, } -function runTest(testfile) { +function runTest(botType, testfile) { console.log("Running test: " + testfile); var script = vm.createScript(fs.readFileSync(testfile), testfile); - script.runInNewContext({ test: new Test(process.argv[2]) }); + script.runInNewContext({ test: new Test(botType) }); } -runTest("./test/simple_offer_answer.js"); +runTest(process.argv[2], process.argv[3]); diff --git a/tools/rtcbot/test/webrtc_video_streaming.js b/tools/rtcbot/test/webrtc_video_streaming.js new file mode 100644 index 00000000..a9079763 --- /dev/null +++ b/tools/rtcbot/test/webrtc_video_streaming.js @@ -0,0 +1,94 @@ +// Copyright (c) 2014 The WebRTC project authors. All Rights Reserved. +// +// Use of this source code is governed by a BSD-style license +// that can be found in the LICENSE file in the root of the source +// tree. An additional intellectual property rights grant can be found +// in the file PATENTS. All contributing project authors may +// be found in the AUTHORS file in the root of the source tree. +// +// A unidirectional video and audio flowing test from bot 1 to bot 2. +// +// Note: the source of the video and audio stream is getUserMedia(). +// +// TODO(houssainy): get a condition to terminate the test. +// +function testVideoStreaming(bot1, bot2) { + var pc1 = null; + var pc2 = null; + + test.wait([ + createPeerConnection.bind(bot1), + createPeerConnection.bind(bot2) ], + onPeerConnectionCreated); + + function createPeerConnection(done) { + this.createPeerConnection(done, test.fail); + } + + function onPeerConnectionCreated(peer1, peer2) { + test.log("RTC Peers created."); + pc1 = peer1; + pc2 = peer2; + pc1.addEventListener('addstream', test.fail); + pc2.addEventListener('addstream', onAddStream); + pc1.addEventListener('icecandidate', onIceCandidate.bind(pc2)); + pc2.addEventListener('icecandidate', onIceCandidate.bind(pc1)); + + bot1.getUserMedia({video:true, audio:true}, onUserMediaSuccess, test.fail); + + function onUserMediaSuccess(stream) { + test.log("User has granted access to local media."); + pc1.addStream(stream); + bot1.showStream(stream.id, true, true); + + createOfferAndAnswer(); + } + } + + function onAddStream(event) { + test.log("On Add stream."); + bot2.showStream(event.stream.id, true, false); + } + + function onIceCandidate(event) { + if(event.candidate){ + test.log(event.candidate.candidate); + this.addIceCandidate(event.candidate, + onAddIceCandidateSuccess, test.fail); + }; + + function onAddIceCandidateSuccess() { + test.log("Candidate added successfully"); + }; + } + + function createOfferAndAnswer() { + test.log("Creating offer."); + pc1.createOffer(gotOffer, test.fail); + + function gotOffer(offer) { + test.log("Got offer"); + pc1.setLocalDescription(offer, onSetSessionDescriptionSuccess, test.fail); + pc2.setRemoteDescription(offer, onSetSessionDescriptionSuccess, + test.fail); + test.log("Creating answer"); + pc2.createAnswer(gotAnswer, test.fail); + } + + function gotAnswer(answer) { + test.log("Got answer"); + pc2.setLocalDescription(answer, onSetSessionDescriptionSuccess, + test.fail); + pc1.setRemoteDescription(answer, onSetSessionDescriptionSuccess, + test.fail); + } + + function onSetSessionDescriptionSuccess() { + test.log("Set session description success."); + } + } +} + +test.wait( [ test.spawnBot.bind(test, "alice"), + test.spawnBot.bind(test, "bob") ], + testVideoStreaming); diff --git a/video/bitrate_estimator_tests.cc b/video/bitrate_estimator_tests.cc index 40c1ed68..cb7c391e 100644 --- a/video/bitrate_estimator_tests.cc +++ b/video/bitrate_estimator_tests.cc @@ -13,11 +13,11 @@ #include "testing/gtest/include/gtest/gtest.h" +#include "webrtc/base/thread_annotations.h" #include "webrtc/call.h" #include "webrtc/system_wrappers/interface/critical_section_wrapper.h" #include "webrtc/system_wrappers/interface/event_wrapper.h" #include "webrtc/system_wrappers/interface/scoped_ptr.h" -#include "webrtc/system_wrappers/interface/thread_annotations.h" #include "webrtc/system_wrappers/interface/trace.h" #include "webrtc/test/call_test.h" #include "webrtc/test/direct_transport.h" @@ -148,7 +148,7 @@ class BitrateEstimatorTest : public test::CallTest { send_config_.encoder_settings.encoder = NULL; send_config_.encoder_settings.payload_name = "FAKE"; send_config_.encoder_settings.payload_type = kFakeSendPayloadType; - video_streams_ = test::CreateVideoStreams(1); + encoder_config_.streams = test::CreateVideoStreams(1); receive_config_ = VideoReceiveStream::Config(); assert(receive_config_.codecs.empty()); @@ -195,14 +195,14 @@ class BitrateEstimatorTest : public test::CallTest { test_->send_config_.rtp.ssrcs[0]++; test_->send_config_.encoder_settings.encoder = &fake_encoder_; send_stream_ = test_->sender_call_->CreateVideoSendStream( - test_->send_config_, test_->video_streams_, NULL); - assert(test_->video_streams_.size() == 1); - frame_generator_capturer_.reset( - test::FrameGeneratorCapturer::Create(send_stream_->Input(), - test_->video_streams_[0].width, - test_->video_streams_[0].height, - 30, - Clock::GetRealTimeClock())); + test_->send_config_, test_->encoder_config_); + assert(test_->encoder_config_.streams.size() == 1); + frame_generator_capturer_.reset(test::FrameGeneratorCapturer::Create( + send_stream_->Input(), + test_->encoder_config_.streams[0].width, + test_->encoder_config_.streams[0].height, + 30, + Clock::GetRealTimeClock())); send_stream_->Start(); frame_generator_capturer_->Start(); diff --git a/video/call.cc b/video/call.cc index b4adafd7..25853f8b 100644 --- a/video/call.cc +++ b/video/call.cc @@ -14,6 +14,7 @@ #include <map> #include <vector> +#include "webrtc/base/thread_annotations.h" #include "webrtc/call.h" #include "webrtc/common.h" #include "webrtc/config.h" @@ -22,7 +23,6 @@ #include "webrtc/system_wrappers/interface/critical_section_wrapper.h" #include "webrtc/system_wrappers/interface/rw_lock_wrapper.h" #include "webrtc/system_wrappers/interface/scoped_ptr.h" -#include "webrtc/system_wrappers/interface/thread_annotations.h" #include "webrtc/system_wrappers/interface/trace.h" #include "webrtc/video/video_receive_stream.h" #include "webrtc/video/video_send_stream.h" @@ -87,8 +87,7 @@ class Call : public webrtc::Call, public PacketReceiver { virtual VideoSendStream* CreateVideoSendStream( const VideoSendStream::Config& config, - const std::vector<VideoStream>& video_streams, - const void* encoder_settings) OVERRIDE; + const VideoEncoderConfig& encoder_config) OVERRIDE; virtual void DestroyVideoSendStream(webrtc::VideoSendStream* send_stream) OVERRIDE; @@ -196,8 +195,7 @@ PacketReceiver* Call::Receiver() { return this; } VideoSendStream* Call::CreateVideoSendStream( const VideoSendStream::Config& config, - const std::vector<VideoStream>& video_streams, - const void* encoder_settings) { + const VideoEncoderConfig& encoder_config) { assert(config.rtp.ssrcs.size() > 0); // TODO(mflodman): Base the start bitrate on a current bandwidth estimate, if @@ -207,8 +205,7 @@ VideoSendStream* Call::CreateVideoSendStream( overuse_observer_proxy_.get(), video_engine_, config, - video_streams, - encoder_settings, + encoder_config, suspended_send_ssrcs_, base_channel_id_, config_.start_bitrate_bps != -1 ? config_.start_bitrate_bps diff --git a/video/call_perf_tests.cc b/video/call_perf_tests.cc index 557c5149..cd83698f 100644 --- a/video/call_perf_tests.cc +++ b/video/call_perf_tests.cc @@ -15,6 +15,7 @@ #include "testing/gtest/include/gtest/gtest.h" +#include "webrtc/base/thread_annotations.h" #include "webrtc/call.h" #include "webrtc/modules/audio_coding/main/interface/audio_coding_module.h" #include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h" @@ -22,7 +23,6 @@ #include "webrtc/system_wrappers/interface/critical_section_wrapper.h" #include "webrtc/system_wrappers/interface/rtp_to_ntp.h" #include "webrtc/system_wrappers/interface/scoped_ptr.h" -#include "webrtc/system_wrappers/interface/thread_annotations.h" #include "webrtc/test/call_test.h" #include "webrtc/test/direct_transport.h" #include "webrtc/test/encoder_settings.h" @@ -399,7 +399,7 @@ void CallPerfTest::TestCaptureNtpTime(const FakeNetworkPipe::Config& net_config, virtual void ModifyConfigs( VideoSendStream::Config* send_config, std::vector<VideoReceiveStream::Config>* receive_configs, - std::vector<VideoStream>* video_streams) OVERRIDE { + VideoEncoderConfig* encoder_config) OVERRIDE { (*receive_configs)[0].renderer = this; // Enable the receiver side rtt calculation. (*receive_configs)[0].rtp.rtcp_xr.receiver_reference_time_report = true; @@ -546,7 +546,7 @@ void CallPerfTest::TestMinTransmitBitrate(bool pad_to_min_bitrate) { virtual void ModifyConfigs( VideoSendStream::Config* send_config, std::vector<VideoReceiveStream::Config>* receive_configs, - std::vector<VideoStream>* video_streams) OVERRIDE { + VideoEncoderConfig* encoder_config) OVERRIDE { if (pad_to_min_bitrate_) { send_config->rtp.min_transmit_bitrate_bps = kMinTransmitBitrateBps; } else { diff --git a/video/end_to_end_tests.cc b/video/end_to_end_tests.cc index 46420ecc..0e87a033 100644 --- a/video/end_to_end_tests.cc +++ b/video/end_to_end_tests.cc @@ -208,7 +208,7 @@ TEST_F(EndToEndTest, TransmitsFirstFrame) { Start(); scoped_ptr<test::FrameGenerator> frame_generator(test::FrameGenerator::Create( - video_streams_[0].width, video_streams_[0].height)); + encoder_config_.streams[0].width, encoder_config_.streams[0].height)); send_stream_->Input()->SwapFrame(frame_generator->NextFrame()); EXPECT_EQ(kEventSignaled, renderer.Wait()) @@ -238,13 +238,13 @@ TEST_F(EndToEndTest, SendsAndReceivesH264) { virtual void ModifyConfigs( VideoSendStream::Config* send_config, std::vector<VideoReceiveStream::Config>* receive_configs, - std::vector<VideoStream>* video_streams) { + VideoEncoderConfig* encoder_config) OVERRIDE { send_config->encoder_settings.encoder = &fake_encoder_; send_config->encoder_settings.payload_name = "H264"; send_config->encoder_settings.payload_type = kFakeSendPayloadType; - (*video_streams)[0].min_bitrate_bps = 50000; - (*video_streams)[0].target_bitrate_bps = - (*video_streams)[0].max_bitrate_bps = 2000000; + encoder_config->streams[0].min_bitrate_bps = 50000; + encoder_config->streams[0].target_bitrate_bps = + encoder_config->streams[0].max_bitrate_bps = 2000000; (*receive_configs)[0].renderer = this; VideoCodec codec = @@ -369,7 +369,7 @@ TEST_F(EndToEndTest, ReceivesAndRetransmitsNack) { virtual void ModifyConfigs( VideoSendStream::Config* send_config, std::vector<VideoReceiveStream::Config>* receive_configs, - std::vector<VideoStream>* video_streams) OVERRIDE { + VideoEncoderConfig* encoder_config) OVERRIDE { send_config->rtp.nack.rtp_history_ms = kNackRtpHistoryMs; (*receive_configs)[0].rtp.nack.rtp_history_ms = kNackRtpHistoryMs; } @@ -463,7 +463,7 @@ TEST_F(EndToEndTest, DISABLED_CanReceiveFec) { virtual void ModifyConfigs( VideoSendStream::Config* send_config, std::vector<VideoReceiveStream::Config>* receive_configs, - std::vector<VideoStream>* video_streams) OVERRIDE { + VideoEncoderConfig* encoder_config) OVERRIDE { // TODO(pbos): Run this test with combined NACK/FEC enabled as well. // int rtp_history_ms = 1000; // (*receive_configs)[0].rtp.nack.rtp_history_ms = rtp_history_ms; @@ -541,7 +541,7 @@ void EndToEndTest::DecodesRetransmittedFrame(bool retransmit_over_rtx) { virtual void ModifyConfigs( VideoSendStream::Config* send_config, std::vector<VideoReceiveStream::Config>* receive_configs, - std::vector<VideoStream>* video_streams) OVERRIDE { + VideoEncoderConfig* encoder_config) OVERRIDE { send_config->rtp.nack.rtp_history_ms = kNackRtpHistoryMs; (*receive_configs)[0].pre_render_callback = this; (*receive_configs)[0].rtp.nack.rtp_history_ms = kNackRtpHistoryMs; @@ -578,8 +578,7 @@ TEST_F(EndToEndTest, DecodesRetransmittedFrameOverRtx) { DecodesRetransmittedFrame(true); } -// Disabled due to: https://code.google.com/p/webrtc/issues/detail?id=3770 -TEST_F(EndToEndTest, DISABLED_ON_ANDROID(UsesFrameCallbacks)) { +TEST_F(EndToEndTest, UsesFrameCallbacks) { static const int kWidth = 320; static const int kHeight = 240; @@ -648,9 +647,9 @@ TEST_F(EndToEndTest, DISABLED_ON_ANDROID(UsesFrameCallbacks)) { VideoEncoder::Create(VideoEncoder::kVp8)); send_config_.encoder_settings.encoder = encoder.get(); send_config_.encoder_settings.payload_name = "VP8"; - ASSERT_EQ(1u, video_streams_.size()) << "Test setup error."; - video_streams_[0].width = kWidth; - video_streams_[0].height = kHeight; + ASSERT_EQ(1u, encoder_config_.streams.size()) << "Test setup error."; + encoder_config_.streams[0].width = kWidth; + encoder_config_.streams[0].height = kHeight; send_config_.pre_encode_callback = &pre_encode_callback; CreateMatchingReceiveConfigs(); @@ -745,7 +744,7 @@ void EndToEndTest::ReceivesPliAndRecovers(int rtp_history_ms) { virtual void ModifyConfigs( VideoSendStream::Config* send_config, std::vector<VideoReceiveStream::Config>* receive_configs, - std::vector<VideoStream>* video_streams) OVERRIDE { + VideoEncoderConfig* encoder_config) OVERRIDE { send_config->rtp.nack.rtp_history_ms = rtp_history_ms_; (*receive_configs)[0].rtp.nack.rtp_history_ms = rtp_history_ms_; (*receive_configs)[0].renderer = this; @@ -892,7 +891,7 @@ void EndToEndTest::RespectsRtcpMode(newapi::RtcpMode rtcp_mode) { virtual void ModifyConfigs( VideoSendStream::Config* send_config, std::vector<VideoReceiveStream::Config>* receive_configs, - std::vector<VideoStream>* video_streams) OVERRIDE { + VideoEncoderConfig* encoder_config) OVERRIDE { send_config->rtp.nack.rtp_history_ms = kNackRtpHistoryMs; (*receive_configs)[0].rtp.nack.rtp_history_ms = kNackRtpHistoryMs; (*receive_configs)[0].rtp.rtcp_mode = rtcp_mode_; @@ -925,9 +924,7 @@ TEST_F(EndToEndTest, UsesRtcpReducedSizeMode) { // Another is set up to receive all three of these with different renderers. // Each renderer verifies that it receives the expected resolution, and as soon // as every renderer has received a frame, the test finishes. -// -// Disabled due to: https://code.google.com/p/webrtc/issues/detail?id=3770 -TEST_F(EndToEndTest, DISABLED_ON_ANDROID(SendsAndReceivesMultipleStreams)) { +TEST_F(EndToEndTest, SendsAndReceivesMultipleStreams) { static const size_t kNumStreams = 3; class VideoOutputObserver : public VideoRenderer { @@ -991,15 +988,16 @@ TEST_F(EndToEndTest, DISABLED_ON_ANDROID(SendsAndReceivesMultipleStreams)) { send_config.encoder_settings.encoder = encoders[i].get(); send_config.encoder_settings.payload_name = "VP8"; send_config.encoder_settings.payload_type = 124; - std::vector<VideoStream> video_streams = test::CreateVideoStreams(1); - VideoStream* stream = &video_streams[0]; + VideoEncoderConfig encoder_config; + encoder_config.streams = test::CreateVideoStreams(1); + VideoStream* stream = &encoder_config.streams[0]; stream->width = width; stream->height = height; stream->max_framerate = 5; stream->min_bitrate_bps = stream->target_bitrate_bps = stream->max_bitrate_bps = 100000; send_streams[i] = - sender_call->CreateVideoSendStream(send_config, video_streams, NULL); + sender_call->CreateVideoSendStream(send_config, encoder_config); send_streams[i]->Start(); VideoReceiveStream::Config receive_config; @@ -1090,7 +1088,7 @@ TEST_F(EndToEndTest, ObserversEncodedFrames) { Start(); scoped_ptr<test::FrameGenerator> frame_generator(test::FrameGenerator::Create( - video_streams_[0].width, video_streams_[0].height)); + encoder_config_.streams[0].width, encoder_config_.streams[0].height)); send_stream_->Input()->SwapFrame(frame_generator->NextFrame()); EXPECT_EQ(kEventSignaled, post_encode_observer.Wait()) @@ -1215,7 +1213,7 @@ void EndToEndTest::TestXrReceiverReferenceTimeReport(bool enable_rrtr) { virtual void ModifyConfigs( VideoSendStream::Config* send_config, std::vector<VideoReceiveStream::Config>* receive_configs, - std::vector<VideoStream>* video_streams) OVERRIDE { + VideoEncoderConfig* encoder_config) OVERRIDE { (*receive_configs)[0].rtp.rtcp_mode = newapi::kRtcpReducedSize; (*receive_configs)[0].rtp.rtcp_xr.receiver_reference_time_report = enable_rrtr_; @@ -1283,19 +1281,19 @@ void EndToEndTest::TestSendsSetSsrcs(size_t num_ssrcs, virtual void ModifyConfigs( VideoSendStream::Config* send_config, std::vector<VideoReceiveStream::Config>* receive_configs, - std::vector<VideoStream>* video_streams) OVERRIDE { + VideoEncoderConfig* encoder_config) OVERRIDE { if (num_ssrcs_ > 1) { // Set low simulcast bitrates to not have to wait for bandwidth ramp-up. - for (size_t i = 0; i < video_streams->size(); ++i) { - (*video_streams)[i].min_bitrate_bps = 10000; - (*video_streams)[i].target_bitrate_bps = 15000; - (*video_streams)[i].max_bitrate_bps = 20000; + for (size_t i = 0; i < encoder_config->streams.size(); ++i) { + encoder_config->streams[i].min_bitrate_bps = 10000; + encoder_config->streams[i].target_bitrate_bps = 15000; + encoder_config->streams[i].max_bitrate_bps = 20000; } } - all_streams_ = *video_streams; + encoder_config_all_streams_ = *encoder_config; if (send_single_ssrc_first_) - video_streams->resize(1); + encoder_config->streams.resize(1); } virtual void OnStreamsCreated( @@ -1311,7 +1309,7 @@ void EndToEndTest::TestSendsSetSsrcs(size_t num_ssrcs, if (send_single_ssrc_first_) { // Set full simulcast and continue with the rest of the SSRCs. - send_stream_->ReconfigureVideoEncoder(all_streams_, NULL); + send_stream_->ReconfigureVideoEncoder(encoder_config_all_streams_); EXPECT_EQ(kEventSignaled, Wait()) << "Timed out while waiting on additional SSRCs."; } @@ -1328,7 +1326,7 @@ void EndToEndTest::TestSendsSetSsrcs(size_t num_ssrcs, bool expect_single_ssrc_; VideoSendStream* send_stream_; - std::vector<VideoStream> all_streams_; + VideoEncoderConfig encoder_config_all_streams_; } test(kSendSsrcs, num_ssrcs, send_single_ssrc_first); RunBaseTest(&test); @@ -1472,7 +1470,7 @@ TEST_F(EndToEndTest, GetStats) { virtual void ModifyConfigs( VideoSendStream::Config* send_config, std::vector<VideoReceiveStream::Config>* receive_configs, - std::vector<VideoStream>* video_streams) OVERRIDE { + VideoEncoderConfig* encoder_config) OVERRIDE { send_config->pre_encode_callback = this; // Used to inject delay. send_config->rtp.c_name = "SomeCName"; @@ -1648,12 +1646,12 @@ TEST_F(EndToEndTest, DISABLED_RedundantPayloadsTransmittedOnAllSsrcs) { virtual void ModifyConfigs( VideoSendStream::Config* send_config, std::vector<VideoReceiveStream::Config>* receive_configs, - std::vector<VideoStream>* video_streams) OVERRIDE { + VideoEncoderConfig* encoder_config) OVERRIDE { // Set low simulcast bitrates to not have to wait for bandwidth ramp-up. - for (size_t i = 0; i < video_streams->size(); ++i) { - (*video_streams)[i].min_bitrate_bps = 10000; - (*video_streams)[i].target_bitrate_bps = 15000; - (*video_streams)[i].max_bitrate_bps = 20000; + for (size_t i = 0; i < encoder_config->streams.size(); ++i) { + encoder_config->streams[i].min_bitrate_bps = 10000; + encoder_config->streams[i].target_bitrate_bps = 15000; + encoder_config->streams[i].max_bitrate_bps = 20000; } // Significantly higher than max bitrates for all video streams -> forcing // padding to trigger redundant padding on all RTX SSRCs. @@ -1787,10 +1785,10 @@ void EndToEndTest::TestRtpStatePreservation(bool use_rtx) { } // Lower bitrates so that all streams send initially. - for (size_t i = 0; i < video_streams_.size(); ++i) { - video_streams_[i].min_bitrate_bps = 10000; - video_streams_[i].target_bitrate_bps = 15000; - video_streams_[i].max_bitrate_bps = 20000; + for (size_t i = 0; i < encoder_config_.streams.size(); ++i) { + encoder_config_.streams[i].min_bitrate_bps = 10000; + encoder_config_.streams[i].target_bitrate_bps = 15000; + encoder_config_.streams[i].max_bitrate_bps = 20000; } CreateMatchingReceiveConfigs(); @@ -1809,10 +1807,10 @@ void EndToEndTest::TestRtpStatePreservation(bool use_rtx) { sender_call_->DestroyVideoSendStream(send_stream_); // Re-create VideoSendStream with only one stream. - std::vector<VideoStream> one_stream = video_streams_; - one_stream.resize(1); + VideoEncoderConfig one_stream = encoder_config_; + one_stream.streams.resize(1); send_stream_ = - sender_call_->CreateVideoSendStream(send_config_, one_stream, NULL); + sender_call_->CreateVideoSendStream(send_config_, one_stream); send_stream_->Start(); CreateFrameGeneratorCapturer(); frame_generator_capturer_->Start(); @@ -1822,19 +1820,19 @@ void EndToEndTest::TestRtpStatePreservation(bool use_rtx) { << "Timed out waiting for single RTP packet."; // Reconfigure back to use all streams. - send_stream_->ReconfigureVideoEncoder(video_streams_, NULL); + send_stream_->ReconfigureVideoEncoder(encoder_config_); observer.ResetExpectedSsrcs(kNumSsrcs); EXPECT_EQ(kEventSignaled, observer.Wait()) << "Timed out waiting for all SSRCs to send packets."; // Reconfigure down to one stream. - send_stream_->ReconfigureVideoEncoder(one_stream, NULL); + send_stream_->ReconfigureVideoEncoder(one_stream); observer.ResetExpectedSsrcs(1); EXPECT_EQ(kEventSignaled, observer.Wait()) << "Timed out waiting for single RTP packet."; // Reconfigure back to use all streams. - send_stream_->ReconfigureVideoEncoder(video_streams_, NULL); + send_stream_->ReconfigureVideoEncoder(encoder_config_); observer.ResetExpectedSsrcs(kNumSsrcs); EXPECT_EQ(kEventSignaled, observer.Wait()) << "Timed out waiting for all SSRCs to send packets."; @@ -1937,7 +1935,7 @@ TEST_F(EndToEndTest, RespectsNetworkState) { virtual void ModifyConfigs( VideoSendStream::Config* send_config, std::vector<VideoReceiveStream::Config>* receive_configs, - std::vector<VideoStream>* video_streams) OVERRIDE { + VideoEncoderConfig* encoder_config) OVERRIDE { send_config->encoder_settings.encoder = this; } diff --git a/video/full_stack.cc b/video/full_stack.cc index cd1190cc..f2d71a97 100644 --- a/video/full_stack.cc +++ b/video/full_stack.cc @@ -14,6 +14,7 @@ #include "testing/gtest/include/gtest/gtest.h" +#include "webrtc/base/thread_annotations.h" #include "webrtc/call.h" #include "webrtc/common_video/libyuv/include/webrtc_libyuv.h" #include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h" @@ -22,7 +23,6 @@ #include "webrtc/system_wrappers/interface/event_wrapper.h" #include "webrtc/system_wrappers/interface/scoped_ptr.h" #include "webrtc/system_wrappers/interface/sleep.h" -#include "webrtc/system_wrappers/interface/thread_annotations.h" #include "webrtc/test/call_test.h" #include "webrtc/test/direct_transport.h" #include "webrtc/test/encoder_settings.h" @@ -393,7 +393,7 @@ void FullStackTest::RunTest(const FullStackTestParams& params) { send_config_.encoder_settings.payload_name = "VP8"; send_config_.encoder_settings.payload_type = 124; - VideoStream* stream = &video_streams_[0]; + VideoStream* stream = &encoder_config_.streams[0]; stream->width = params.clip.width; stream->height = params.clip.height; stream->min_bitrate_bps = params.min_bitrate_bps; diff --git a/video/loopback.cc b/video/loopback.cc index 29a3c782..488adf41 100644 --- a/video/loopback.cc +++ b/video/loopback.cc @@ -129,8 +129,9 @@ void Loopback() { send_config.encoder_settings.encoder = encoder.get(); send_config.encoder_settings.payload_name = flags::Codec(); send_config.encoder_settings.payload_type = 124; - std::vector<VideoStream> video_streams = test::CreateVideoStreams(1); - VideoStream* stream = &video_streams[0]; + VideoEncoderConfig encoder_config; + encoder_config.streams = test::CreateVideoStreams(1); + VideoStream* stream = &encoder_config.streams[0]; stream->width = flags::Width(); stream->height = flags::Height(); stream->min_bitrate_bps = static_cast<int>(flags::MinBitrate()) * 1000; @@ -140,7 +141,7 @@ void Loopback() { stream->max_qp = 56; VideoSendStream* send_stream = - call->CreateVideoSendStream(send_config, video_streams, NULL); + call->CreateVideoSendStream(send_config, encoder_config); Clock* test_clock = Clock::GetRealTimeClock(); diff --git a/video/rampup_tests.cc b/video/rampup_tests.cc index e1dd95a6..10bcb7f1 100644 --- a/video/rampup_tests.cc +++ b/video/rampup_tests.cc @@ -411,8 +411,8 @@ void RampUpTest::RunRampUpTest(bool rtx, receiver_transport.SetReceiver(sender_call_->Receiver()); if (num_streams == 1) { - video_streams_[0].target_bitrate_bps = 2000000; - video_streams_[0].max_bitrate_bps = 2000000; + encoder_config_.streams[0].target_bitrate_bps = 2000000; + encoder_config_.streams[0].max_bitrate_bps = 2000000; } send_config_.rtp.nack.rtp_history_ms = kNackRtpHistoryMs; @@ -430,9 +430,9 @@ void RampUpTest::RunRampUpTest(bool rtx, // For multi stream rampup until all streams are being sent. That means // enough birate to send all the target streams plus the min bitrate of // the last one. - int expected_bitrate_bps = video_streams_.back().min_bitrate_bps; - for (size_t i = 0; i < video_streams_.size() - 1; ++i) { - expected_bitrate_bps += video_streams_[i].target_bitrate_bps; + int expected_bitrate_bps = encoder_config_.streams.back().min_bitrate_bps; + for (size_t i = 0; i < encoder_config_.streams.size() - 1; ++i) { + expected_bitrate_bps += encoder_config_.streams[i].target_bitrate_bps; } stream_observer.set_expected_bitrate_bps(expected_bitrate_bps); } diff --git a/video/receive_statistics_proxy.h b/video/receive_statistics_proxy.h index 35e5cc3e..b5fbf860 100644 --- a/video/receive_statistics_proxy.h +++ b/video/receive_statistics_proxy.h @@ -13,10 +13,10 @@ #include <string> +#include "webrtc/base/thread_annotations.h" #include "webrtc/common_types.h" #include "webrtc/frame_callback.h" #include "webrtc/modules/remote_bitrate_estimator/rate_statistics.h" -#include "webrtc/system_wrappers/interface/thread_annotations.h" #include "webrtc/video_engine/include/vie_codec.h" #include "webrtc/video_engine/include/vie_rtp_rtcp.h" #include "webrtc/video_receive_stream.h" diff --git a/video/send_statistics_proxy.h b/video/send_statistics_proxy.h index 1b888b03..ef459dab 100644 --- a/video/send_statistics_proxy.h +++ b/video/send_statistics_proxy.h @@ -13,12 +13,12 @@ #include <string> +#include "webrtc/base/thread_annotations.h" #include "webrtc/common_types.h" #include "webrtc/video_engine/include/vie_codec.h" #include "webrtc/video_engine/include/vie_capture.h" #include "webrtc/video_send_stream.h" #include "webrtc/system_wrappers/interface/scoped_ptr.h" -#include "webrtc/system_wrappers/interface/thread_annotations.h" namespace webrtc { diff --git a/video/video_send_stream.cc b/video/video_send_stream.cc index 7d64755d..6e8f2381 100644 --- a/video/video_send_stream.cc +++ b/video/video_send_stream.cc @@ -114,8 +114,7 @@ VideoSendStream::VideoSendStream( CpuOveruseObserver* overuse_observer, webrtc::VideoEngine* video_engine, const VideoSendStream::Config& config, - const std::vector<VideoStream> video_streams, - const void* encoder_settings, + const VideoEncoderConfig& encoder_config, const std::map<uint32_t, RtpState>& suspended_ssrcs, int base_channel, int start_bitrate_bps) @@ -211,7 +210,7 @@ VideoSendStream::VideoSendStream( } codec_ = ViECodec::GetInterface(video_engine); - if (!ReconfigureVideoEncoder(video_streams, encoder_settings)) + if (!ReconfigureVideoEncoder(encoder_config)) abort(); if (overuse_observer) @@ -297,8 +296,8 @@ void VideoSendStream::Stop() { } bool VideoSendStream::ReconfigureVideoEncoder( - const std::vector<VideoStream>& streams, - const void* encoder_settings) { + const VideoEncoderConfig& config) { + const std::vector<VideoStream>& streams = config.streams; assert(!streams.empty()); assert(config_.rtp.ssrcs.size() >= streams.size()); @@ -311,6 +310,14 @@ bool VideoSendStream::ReconfigureVideoEncoder( } else { video_codec.codecType = kVideoCodecGeneric; } + switch (config.content_type) { + case VideoEncoderConfig::kRealtimeVideo: + video_codec.mode = kRealtimeVideo; + break; + case VideoEncoderConfig::kScreenshare: + video_codec.mode = kScreensharing; + break; + } if (video_codec.codecType == kVideoCodecVP8) { video_codec.codecSpecific.VP8 = VideoEncoder::GetDefaultVp8Settings(); @@ -319,13 +326,15 @@ bool VideoSendStream::ReconfigureVideoEncoder( } if (video_codec.codecType == kVideoCodecVP8) { - if (encoder_settings != NULL) { - video_codec.codecSpecific.VP8 = - *reinterpret_cast<const VideoCodecVP8*>(encoder_settings); + if (config.encoder_specific_settings != NULL) { + video_codec.codecSpecific.VP8 = *reinterpret_cast<const VideoCodecVP8*>( + config.encoder_specific_settings); } + video_codec.codecSpecific.VP8.numberOfTemporalLayers = + static_cast<unsigned char>(streams.back().temporal_layers.size()); } else { // TODO(pbos): Support encoder_settings codec-agnostically. - assert(encoder_settings == NULL); + assert(config.encoder_specific_settings == NULL); } strncpy(video_codec.plName, @@ -355,8 +364,8 @@ bool VideoSendStream::ReconfigureVideoEncoder( sim_stream->targetBitrate = streams[i].target_bitrate_bps / 1000; sim_stream->maxBitrate = streams[i].max_bitrate_bps / 1000; sim_stream->qpMax = streams[i].max_qp; - // TODO(pbos): Implement mapping for temporal layers. - assert(streams[i].temporal_layers.empty()); + sim_stream->numberOfTemporalLayers = + static_cast<unsigned char>(streams[i].temporal_layers.size()); video_codec.width = std::max(video_codec.width, static_cast<unsigned short>(streams[i].width)); diff --git a/video/video_send_stream.h b/video/video_send_stream.h index 130c1c5c..8a77852e 100644 --- a/video/video_send_stream.h +++ b/video/video_send_stream.h @@ -46,8 +46,7 @@ class VideoSendStream : public webrtc::VideoSendStream, CpuOveruseObserver* overuse_observer, webrtc::VideoEngine* video_engine, const VideoSendStream::Config& config, - const std::vector<VideoStream> video_streams, - const void* encoder_settings, + const VideoEncoderConfig& encoder_config, const std::map<uint32_t, RtpState>& suspended_ssrcs, int base_channel, int start_bitrate); @@ -57,8 +56,8 @@ class VideoSendStream : public webrtc::VideoSendStream, virtual void Start() OVERRIDE; virtual void Stop() OVERRIDE; - virtual bool ReconfigureVideoEncoder(const std::vector<VideoStream>& streams, - const void* encoder_settings) OVERRIDE; + virtual bool ReconfigureVideoEncoder( + const VideoEncoderConfig& config) OVERRIDE; virtual Stats GetStats() const OVERRIDE; diff --git a/video/video_send_stream_tests.cc b/video/video_send_stream_tests.cc index 7a641bd4..6c2fa39f 100644 --- a/video/video_send_stream_tests.cc +++ b/video/video_send_stream_tests.cc @@ -114,7 +114,7 @@ TEST_F(VideoSendStreamTest, SupportsCName) { virtual void ModifyConfigs( VideoSendStream::Config* send_config, std::vector<VideoReceiveStream::Config>* receive_configs, - std::vector<VideoStream>* video_streams) OVERRIDE { + VideoEncoderConfig* encoder_config) OVERRIDE { send_config->rtp.c_name = kCName; } @@ -152,7 +152,7 @@ TEST_F(VideoSendStreamTest, SupportsAbsoluteSendTime) { virtual void ModifyConfigs( VideoSendStream::Config* send_config, std::vector<VideoReceiveStream::Config>* receive_configs, - std::vector<VideoStream>* video_streams) OVERRIDE { + VideoEncoderConfig* encoder_config) OVERRIDE { send_config->rtp.extensions.push_back( RtpExtension(RtpExtension::kAbsSendTime, kAbsSendTimeExtensionId)); } @@ -193,7 +193,7 @@ TEST_F(VideoSendStreamTest, SupportsTransmissionTimeOffset) { virtual void ModifyConfigs( VideoSendStream::Config* send_config, std::vector<VideoReceiveStream::Config>* receive_configs, - std::vector<VideoStream>* video_streams) OVERRIDE { + VideoEncoderConfig* encoder_config) OVERRIDE { send_config->encoder_settings.encoder = &encoder_; send_config->rtp.extensions.push_back( RtpExtension(RtpExtension::kTOffset, kTOffsetExtensionId)); @@ -360,7 +360,7 @@ TEST_F(VideoSendStreamTest, SupportsFec) { virtual void ModifyConfigs( VideoSendStream::Config* send_config, std::vector<VideoReceiveStream::Config>* receive_configs, - std::vector<VideoStream>* video_streams) OVERRIDE { + VideoEncoderConfig* encoder_config) OVERRIDE { send_config->rtp.fec.red_payload_type = kRedPayloadType; send_config->rtp.fec.ulpfec_payload_type = kUlpfecPayloadType; } @@ -440,7 +440,7 @@ void VideoSendStreamTest::TestNackRetransmission( virtual void ModifyConfigs( VideoSendStream::Config* send_config, std::vector<VideoReceiveStream::Config>* receive_configs, - std::vector<VideoStream>* video_streams) OVERRIDE { + VideoEncoderConfig* encoder_config) OVERRIDE { send_config->rtp.nack.rtp_history_ms = kNackRtpHistoryMs; send_config->rtp.rtx.payload_type = retransmit_payload_type_; if (retransmit_ssrc_ != kSendSsrcs[0]) @@ -613,7 +613,7 @@ void VideoSendStreamTest::TestPacketFragmentationSize(VideoFormat format, virtual void ModifyConfigs( VideoSendStream::Config* send_config, std::vector<VideoReceiveStream::Config>* receive_configs, - std::vector<VideoStream>* video_streams) OVERRIDE { + VideoEncoderConfig* encoder_config) OVERRIDE { if (use_fec_) { send_config->rtp.fec.red_payload_type = kRedPayloadType; send_config->rtp.fec.ulpfec_payload_type = kUlpfecPayloadType; @@ -788,14 +788,14 @@ TEST_F(VideoSendStreamTest, SuspendBelowMinBitrate) { virtual void ModifyConfigs( VideoSendStream::Config* send_config, std::vector<VideoReceiveStream::Config>* receive_configs, - std::vector<VideoStream>* video_streams) OVERRIDE { + VideoEncoderConfig* encoder_config) OVERRIDE { send_config->rtp.nack.rtp_history_ms = kNackRtpHistoryMs; send_config->pre_encode_callback = this; send_config->suspend_below_min_bitrate = true; - int min_bitrate_bps = (*video_streams)[0].min_bitrate_bps; + int min_bitrate_bps = encoder_config->streams[0].min_bitrate_bps; set_low_remb_bps(min_bitrate_bps - 10000); int threshold_window = std::max(min_bitrate_bps / 10, 10000); - ASSERT_GT((*video_streams)[0].max_bitrate_bps, + ASSERT_GT(encoder_config->streams[0].max_bitrate_bps, min_bitrate_bps + threshold_window + 5000); set_high_remb_bps(min_bitrate_bps + threshold_window + 5000); } @@ -977,7 +977,7 @@ TEST_F(VideoSendStreamTest, ProducesStats) { virtual void ModifyConfigs( VideoSendStream::Config* send_config, std::vector<VideoReceiveStream::Config>* receive_configs, - std::vector<VideoStream>* video_streams) OVERRIDE { + VideoEncoderConfig* encoder_config) OVERRIDE { SetConfig(*send_config); } @@ -1074,7 +1074,7 @@ TEST_F(VideoSendStreamTest, MinTransmitBitrateRespectsRemb) { virtual void ModifyConfigs( VideoSendStream::Config* send_config, std::vector<VideoReceiveStream::Config>* receive_configs, - std::vector<VideoStream>* video_streams) OVERRIDE { + VideoEncoderConfig* encoder_config) OVERRIDE { send_config->rtp.min_transmit_bitrate_bps = kMinTransmitBitrateBps; } @@ -1133,8 +1133,8 @@ TEST_F(VideoSendStreamTest, CapturesTextureAndI420VideoFrames) { // Prepare five input frames. Send I420VideoFrame and TextureVideoFrame // alternatively. ScopedVector<I420VideoFrame> input_frames; - int width = static_cast<int>(video_streams_[0].width); - int height = static_cast<int>(video_streams_[0].height); + int width = static_cast<int>(encoder_config_.streams[0].width); + int height = static_cast<int>(encoder_config_.streams[0].height); webrtc::RefCountImpl<FakeNativeHandle>* handle1 = new webrtc::RefCountImpl<FakeNativeHandle>(); webrtc::RefCountImpl<FakeNativeHandle>* handle2 = @@ -1154,7 +1154,7 @@ TEST_F(VideoSendStreamTest, CapturesTextureAndI420VideoFrames) { send_stream_->Input()->SwapFrame(frame.get()); // Do not send the next frame too fast, so the frame dropper won't drop it. if (i < input_frames.size() - 1) - SleepMs(1000 / video_streams_[0].max_framerate); + SleepMs(1000 / encoder_config_.streams[0].max_framerate); // Wait until the output frame is received before sending the next input // frame. Or the previous input frame may be replaced without delivering. observer.WaitOutputFrame(); @@ -1327,16 +1327,16 @@ TEST_F(VideoSendStreamTest, EncoderIsProperlyInitializedAndDestroyed) { virtual void ModifyConfigs( VideoSendStream::Config* send_config, std::vector<VideoReceiveStream::Config>* receive_configs, - std::vector<VideoStream>* video_streams) OVERRIDE { + VideoEncoderConfig* encoder_config) OVERRIDE { send_config->encoder_settings.encoder = this; - video_streams_ = *video_streams; + encoder_config_ = *encoder_config; } virtual void PerformTest() OVERRIDE { EXPECT_EQ(kEventSignaled, Wait()) << "Timed out while waiting for Encode."; EXPECT_EQ(0u, num_releases()); - stream_->ReconfigureVideoEncoder(video_streams_, NULL); + stream_->ReconfigureVideoEncoder(encoder_config_); EXPECT_EQ(0u, num_releases()); stream_->Stop(); // Encoder should not be released before destroying the VideoSendStream. @@ -1354,7 +1354,7 @@ TEST_F(VideoSendStreamTest, EncoderIsProperlyInitializedAndDestroyed) { bool callback_registered_ GUARDED_BY(crit_); size_t num_releases_ GUARDED_BY(crit_); bool released_ GUARDED_BY(crit_); - std::vector<VideoStream> video_streams_; + VideoEncoderConfig encoder_config_; } test_encoder; RunBaseTest(&test_encoder); @@ -1363,7 +1363,64 @@ TEST_F(VideoSendStreamTest, EncoderIsProperlyInitializedAndDestroyed) { EXPECT_EQ(1u, test_encoder.num_releases()); } +TEST_F(VideoSendStreamTest, EncoderSetupPropagatesCommonEncoderConfigValues) { + class VideoCodecConfigObserver : public test::SendTest, + public test::FakeEncoder { + public: + VideoCodecConfigObserver() + : SendTest(kDefaultTimeoutMs), + FakeEncoder(Clock::GetRealTimeClock()), + num_initializations_(0) {} + + private: + virtual void ModifyConfigs( + VideoSendStream::Config* send_config, + std::vector<VideoReceiveStream::Config>* receive_configs, + VideoEncoderConfig* encoder_config) OVERRIDE { + send_config->encoder_settings.encoder = this; + encoder_config_ = *encoder_config; + } + + virtual void OnStreamsCreated( + VideoSendStream* send_stream, + const std::vector<VideoReceiveStream*>& receive_streams) OVERRIDE { + stream_ = send_stream; + } + + virtual int32_t InitEncode(const VideoCodec* config, + int32_t number_of_cores, + uint32_t max_payload_size) OVERRIDE { + if (num_initializations_ == 0) { + // Verify default values. + EXPECT_EQ(kRealtimeVideo, config->mode); + } else { + // Verify that changed values are propagated. + EXPECT_EQ(kScreensharing, config->mode); + } + ++num_initializations_; + return FakeEncoder::InitEncode(config, number_of_cores, max_payload_size); + } + + virtual void PerformTest() OVERRIDE { + EXPECT_EQ(1u, num_initializations_) << "VideoEncoder not initialized."; + + encoder_config_.content_type = VideoEncoderConfig::kScreenshare; + stream_->ReconfigureVideoEncoder(encoder_config_); + EXPECT_EQ(2u, num_initializations_) + << "ReconfigureVideoEncoder did not reinitialize the encoder with " + "new encoder settings."; + } + + size_t num_initializations_; + VideoSendStream* stream_; + VideoEncoderConfig encoder_config_; + } test; + + RunBaseTest(&test); +} + TEST_F(VideoSendStreamTest, EncoderSetupPropagatesVp8Config) { + static const size_t kNumberOfTemporalLayers = 4; class VideoCodecConfigObserver : public test::SendTest, public test::FakeEncoder { public: @@ -1378,11 +1435,17 @@ TEST_F(VideoSendStreamTest, EncoderSetupPropagatesVp8Config) { virtual void ModifyConfigs( VideoSendStream::Config* send_config, std::vector<VideoReceiveStream::Config>* receive_configs, - std::vector<VideoStream>* video_streams) OVERRIDE { + VideoEncoderConfig* encoder_config) OVERRIDE { send_config->encoder_settings.encoder = this; send_config->encoder_settings.payload_name = "VP8"; - video_streams_ = *video_streams; + for (size_t i = 0; i < encoder_config->streams.size(); ++i) { + encoder_config->streams[i].temporal_layers.resize( + kNumberOfTemporalLayers); + } + + encoder_config->encoder_specific_settings = &vp8_settings_; + encoder_config_ = *encoder_config; } virtual void OnStreamsCreated( @@ -1395,6 +1458,20 @@ TEST_F(VideoSendStreamTest, EncoderSetupPropagatesVp8Config) { int32_t number_of_cores, uint32_t max_payload_size) OVERRIDE { EXPECT_EQ(kVideoCodecVP8, config->codecType); + + // Check that the number of temporal layers has propagated properly to + // VideoCodec. + EXPECT_EQ(kNumberOfTemporalLayers, + config->codecSpecific.VP8.numberOfTemporalLayers); + + for (unsigned char i = 0; i < config->numberOfSimulcastStreams; ++i) { + EXPECT_EQ(kNumberOfTemporalLayers, + config->simulcastStream[i].numberOfTemporalLayers); + } + + // Set expected temporal layers as they should have been set when + // reconfiguring the encoder and not match the set config. + vp8_settings_.numberOfTemporalLayers = kNumberOfTemporalLayers; EXPECT_EQ(0, memcmp(&config->codecSpecific.VP8, &vp8_settings_, @@ -1407,26 +1484,23 @@ TEST_F(VideoSendStreamTest, EncoderSetupPropagatesVp8Config) { EXPECT_EQ(1u, num_initializations_) << "VideoEncoder not initialized."; vp8_settings_.denoisingOn = true; - stream_->ReconfigureVideoEncoder(video_streams_, &vp8_settings_); + stream_->ReconfigureVideoEncoder(encoder_config_); EXPECT_EQ(2u, num_initializations_) << "ReconfigureVideoEncoder did not reinitialize the encoder with " "new encoder settings."; } - int32_t Encode( - const I420VideoFrame& input_image, - const CodecSpecificInfo* codec_specific_info, - const std::vector<VideoFrameType>* frame_types) { + int32_t Encode(const I420VideoFrame& input_image, + const CodecSpecificInfo* codec_specific_info, + const std::vector<VideoFrameType>* frame_types) OVERRIDE { // Silently skip the encode, FakeEncoder::Encode doesn't produce VP8. return 0; } - virtual const void* GetEncoderSettings() OVERRIDE { return &vp8_settings_; } - VideoCodecVP8 vp8_settings_; size_t num_initializations_; VideoSendStream* stream_; - std::vector<VideoStream> video_streams_; + VideoEncoderConfig encoder_config_; } test; RunBaseTest(&test); diff --git a/video_engine/vie_capturer.h b/video_engine/vie_capturer.h index 2464ca04..fdee5d88 100644 --- a/video_engine/vie_capturer.h +++ b/video_engine/vie_capturer.h @@ -13,6 +13,7 @@ #include <vector> +#include "webrtc/base/thread_annotations.h" #include "webrtc/common_types.h" #include "webrtc/engine_configurations.h" #include "webrtc/modules/video_capture/include/video_capture.h" @@ -20,7 +21,6 @@ #include "webrtc/modules/video_coding/main/interface/video_coding.h" #include "webrtc/modules/video_processing/main/interface/video_processing.h" #include "webrtc/system_wrappers/interface/scoped_ptr.h" -#include "webrtc/system_wrappers/interface/thread_annotations.h" #include "webrtc/typedefs.h" #include "webrtc/video_engine/include/vie_base.h" #include "webrtc/video_engine/include/vie_capture.h" diff --git a/video_engine/vie_channel_group.cc b/video_engine/vie_channel_group.cc index 6b2d3570..c6b3f741 100644 --- a/video_engine/vie_channel_group.cc +++ b/video_engine/vie_channel_group.cc @@ -10,6 +10,7 @@ #include "webrtc/video_engine/vie_channel_group.h" +#include "webrtc/base/thread_annotations.h" #include "webrtc/common.h" #include "webrtc/experiments.h" #include "webrtc/modules/bitrate_controller/include/bitrate_controller.h" @@ -18,7 +19,6 @@ #include "webrtc/modules/utility/interface/process_thread.h" #include "webrtc/system_wrappers/interface/critical_section_wrapper.h" #include "webrtc/system_wrappers/interface/logging.h" -#include "webrtc/system_wrappers/interface/thread_annotations.h" #include "webrtc/video_engine/call_stats.h" #include "webrtc/video_engine/encoder_state_feedback.h" #include "webrtc/video_engine/vie_channel.h" diff --git a/video_engine/vie_encoder.h b/video_engine/vie_encoder.h index e2e691e0..51c1d016 100644 --- a/video_engine/vie_encoder.h +++ b/video_engine/vie_encoder.h @@ -14,13 +14,13 @@ #include <list> #include <map> +#include "webrtc/base/thread_annotations.h" #include "webrtc/common_types.h" #include "webrtc/modules/bitrate_controller/include/bitrate_controller.h" #include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h" #include "webrtc/modules/video_coding/main/interface/video_coding_defines.h" #include "webrtc/modules/video_processing/main/interface/video_processing.h" #include "webrtc/system_wrappers/interface/scoped_ptr.h" -#include "webrtc/system_wrappers/interface/thread_annotations.h" #include "webrtc/typedefs.h" #include "webrtc/frame_callback.h" #include "webrtc/video_engine/vie_defines.h" diff --git a/video_engine/vie_manager_base.h b/video_engine/vie_manager_base.h index f9f008f3..c5e92106 100644 --- a/video_engine/vie_manager_base.h +++ b/video_engine/vie_manager_base.h @@ -11,7 +11,7 @@ #ifndef WEBRTC_VIDEO_ENGINE_VIE_MANAGER_BASE_H_ #define WEBRTC_VIDEO_ENGINE_VIE_MANAGER_BASE_H_ -#include "webrtc/system_wrappers/interface/thread_annotations.h" +#include "webrtc/base/thread_annotations.h" namespace webrtc { diff --git a/video_send_stream.h b/video_send_stream.h index 8c9d5b7f..dd2bec12 100644 --- a/video_send_stream.h +++ b/video_send_stream.h @@ -153,8 +153,7 @@ class VideoSendStream { // Set which streams to send. Must have at least as many SSRCs as configured // in the config. Encoder settings are passed on to the encoder instance along // with the VideoStream settings. - virtual bool ReconfigureVideoEncoder(const std::vector<VideoStream>& streams, - const void* encoder_settings) = 0; + virtual bool ReconfigureVideoEncoder(const VideoEncoderConfig& config) = 0; virtual Stats GetStats() const = 0; diff --git a/voice_engine/channel.cc b/voice_engine/channel.cc index 6d6b099c..93a241be 100644 --- a/voice_engine/channel.cc +++ b/voice_engine/channel.cc @@ -1392,61 +1392,6 @@ Channel::StopReceiving() } int32_t -Channel::SetNetEQPlayoutMode(NetEqModes mode) -{ - WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId), - "Channel::SetNetEQPlayoutMode()"); - AudioPlayoutMode playoutMode(voice); - switch (mode) - { - case kNetEqDefault: - playoutMode = voice; - break; - case kNetEqStreaming: - playoutMode = streaming; - break; - case kNetEqFax: - playoutMode = fax; - break; - case kNetEqOff: - playoutMode = off; - break; - } - if (audio_coding_->SetPlayoutMode(playoutMode) != 0) - { - _engineStatisticsPtr->SetLastError( - VE_AUDIO_CODING_MODULE_ERROR, kTraceError, - "SetNetEQPlayoutMode() failed to set playout mode"); - return -1; - } - return 0; -} - -int32_t -Channel::GetNetEQPlayoutMode(NetEqModes& mode) -{ - const AudioPlayoutMode playoutMode = audio_coding_->PlayoutMode(); - switch (playoutMode) - { - case voice: - mode = kNetEqDefault; - break; - case streaming: - mode = kNetEqStreaming; - break; - case fax: - mode = kNetEqFax; - break; - case off: - mode = kNetEqOff; - } - WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, - VoEId(_instanceId,_channelId), - "Channel::GetNetEQPlayoutMode() => mode=%u", mode); - return 0; -} - -int32_t Channel::RegisterVoiceEngineObserver(VoiceEngineObserver& observer) { WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId), diff --git a/voice_engine/channel.h b/voice_engine/channel.h index d030daf3..535922fa 100644 --- a/voice_engine/channel.h +++ b/voice_engine/channel.h @@ -194,8 +194,6 @@ public: int32_t StartReceiving(); int32_t StopReceiving(); - int32_t SetNetEQPlayoutMode(NetEqModes mode); - int32_t GetNetEQPlayoutMode(NetEqModes& mode); int32_t RegisterVoiceEngineObserver(VoiceEngineObserver& observer); int32_t DeRegisterVoiceEngineObserver(); diff --git a/voice_engine/include/voe_base.h b/voice_engine/include/voe_base.h index 3a78f7b6..65a71758 100644 --- a/voice_engine/include/voe_base.h +++ b/voice_engine/include/voe_base.h @@ -181,9 +181,6 @@ public: OnHoldModes mode = kHoldSendAndPlay) { return -1; } virtual int GetOnHoldStatus(int channel, bool& enabled, OnHoldModes& mode) { return -1; } - virtual int SetNetEQPlayoutMode(int channel, NetEqModes mode) { return -1; } - virtual int GetNetEQPlayoutMode(int channel, - NetEqModes& mode) { return -1; } protected: VoEBase() {} |