diff options
author | henrik.lundin@webrtc.org <henrik.lundin@webrtc.org> | 2014-11-06 15:27:43 +0000 |
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committer | henrik.lundin@webrtc.org <henrik.lundin@webrtc.org> | 2014-11-06 15:27:43 +0000 |
commit | f1e9d8b8a9438b6f645d2dca01cb290d53cb5f06 (patch) | |
tree | 294657f5e770eb0994bfa6560c256aa6409b5c93 | |
parent | eb0e23196dcb82b90c94202d1ed39e6c8abbe96a (diff) | |
download | talk-f1e9d8b8a9438b6f645d2dca01cb290d53cb5f06.tar.gz |
Revert "Advertise G722 as 8 kHz rather than 16 kHz"
This reverts r7645.
TBR=pthatcher@webrtc.org
BUG=3951
Review URL: https://webrtc-codereview.appspot.com/24199004
git-svn-id: http://webrtc.googlecode.com/svn/trunk/talk@7653 4adac7df-926f-26a2-2b94-8c16560cd09d
-rw-r--r-- | media/webrtc/webrtcvoiceengine.cc | 29 | ||||
-rw-r--r-- | media/webrtc/webrtcvoiceengine.h | 1 | ||||
-rw-r--r-- | media/webrtc/webrtcvoiceengine_unittest.cc | 26 |
3 files changed, 7 insertions, 49 deletions
diff --git a/media/webrtc/webrtcvoiceengine.cc b/media/webrtc/webrtcvoiceengine.cc index 6394f09..95e16e4 100644 --- a/media/webrtc/webrtcvoiceengine.cc +++ b/media/webrtc/webrtcvoiceengine.cc @@ -110,7 +110,6 @@ static const int kDefaultAudioDeviceId = 0; static const char kIsacCodecName[] = "ISAC"; static const char kL16CodecName[] = "L16"; -static const char kG722CodecName[] = "G722"; // Parameter used for NACK. // This value is equivalent to 5 seconds of audio data at 20 ms per packet. @@ -486,24 +485,12 @@ static void GetOpusConfig(const AudioCodec& codec, webrtc::CodecInst* voe_codec, voe_codec->rate = GetOpusBitrate(codec, *max_playback_rate); } -// Changes RTP timestamp rate of G722. This is due to the "bug" in the RFC -// which says that G722 should be advertised as 8 kHz although it is a 16 kHz -// codec. -static void MaybeFixupG722(webrtc::CodecInst* voe_codec, int new_plfreq) { - if (_stricmp(voe_codec->plname, kG722CodecName) == 0) { - // If the ASSERT triggers, the codec definition in WebRTC VoiceEngine - // has changed, and this special case is no longer needed. - ASSERT(voe_codec->plfreq != new_plfreq); - voe_codec->plfreq = new_plfreq; - } -} - void WebRtcVoiceEngine::ConstructCodecs() { LOG(LS_INFO) << "WebRtc VoiceEngine codecs:"; int ncodecs = voe_wrapper_->codec()->NumOfCodecs(); for (int i = 0; i < ncodecs; ++i) { webrtc::CodecInst voe_codec; - if (GetVoeCodec(i, voe_codec)) { + if (voe_wrapper_->codec()->GetCodec(i, voe_codec) != -1) { // Skip uncompressed formats. if (_stricmp(voe_codec.plname, kL16CodecName) == 0) { continue; @@ -553,15 +540,6 @@ void WebRtcVoiceEngine::ConstructCodecs() { std::sort(codecs_.begin(), codecs_.end(), &AudioCodec::Preferable); } -bool WebRtcVoiceEngine::GetVoeCodec(int index, webrtc::CodecInst& codec) { - if (voe_wrapper_->codec()->GetCodec(index, codec) != -1) { - // Change the sample rate of G722 to 8000 to match SDP. - MaybeFixupG722(&codec, 8000); - return true; - } - return false; -} - WebRtcVoiceEngine::~WebRtcVoiceEngine() { LOG(LS_VERBOSE) << "WebRtcVoiceEngine::~WebRtcVoiceEngine"; if (voe_wrapper_->base()->DeRegisterVoiceEngineObserver() == -1) { @@ -1246,7 +1224,7 @@ bool WebRtcVoiceEngine::FindWebRtcCodec(const AudioCodec& in, int ncodecs = voe_wrapper_->codec()->NumOfCodecs(); for (int i = 0; i < ncodecs; ++i) { webrtc::CodecInst voe_codec; - if (GetVoeCodec(i, voe_codec)) { + if (voe_wrapper_->codec()->GetCodec(i, voe_codec) != -1) { AudioCodec codec(voe_codec.pltype, voe_codec.plname, voe_codec.plfreq, voe_codec.rate, voe_codec.channels, 0); bool multi_rate = IsCodecMultiRate(voe_codec); @@ -1265,9 +1243,6 @@ bool WebRtcVoiceEngine::FindWebRtcCodec(const AudioCodec& in, voe_codec.rate = in.bitrate; } - // Reset G722 sample rate to 16000 to match WebRTC. - MaybeFixupG722(&voe_codec, 16000); - // Apply codec-specific settings. if (IsIsac(codec)) { // If ISAC and an explicit bitrate is not specified, diff --git a/media/webrtc/webrtcvoiceengine.h b/media/webrtc/webrtcvoiceengine.h index 34b9f3c..f19059b 100644 --- a/media/webrtc/webrtcvoiceengine.h +++ b/media/webrtc/webrtcvoiceengine.h @@ -199,7 +199,6 @@ class WebRtcVoiceEngine void Construct(); void ConstructCodecs(); - bool GetVoeCodec(int index, webrtc::CodecInst& codec); bool InitInternal(); bool EnsureSoundclipEngineInit(); void SetTraceFilter(int filter); diff --git a/media/webrtc/webrtcvoiceengine_unittest.cc b/media/webrtc/webrtcvoiceengine_unittest.cc index 5eb6e24..5deabd2 100644 --- a/media/webrtc/webrtcvoiceengine_unittest.cc +++ b/media/webrtc/webrtcvoiceengine_unittest.cc @@ -52,16 +52,14 @@ static const cricket::AudioCodec kPcmuCodec(0, "PCMU", 8000, 64000, 1, 0); static const cricket::AudioCodec kIsacCodec(103, "ISAC", 16000, 32000, 1, 0); static const cricket::AudioCodec kCeltCodec(110, "CELT", 32000, 64000, 2, 0); static const cricket::AudioCodec kOpusCodec(111, "opus", 48000, 64000, 2, 0); -static const cricket::AudioCodec kG722CodecVoE(9, "G722", 16000, 64000, 1, 0); -static const cricket::AudioCodec kG722CodecSdp(9, "G722", 8000, 64000, 1, 0); static const cricket::AudioCodec kRedCodec(117, "red", 8000, 0, 1, 0); static const cricket::AudioCodec kCn8000Codec(13, "CN", 8000, 0, 1, 0); static const cricket::AudioCodec kCn16000Codec(105, "CN", 16000, 0, 1, 0); static const cricket::AudioCodec kTelephoneEventCodec(106, "telephone-event", 8000, 0, 1, 0); static const cricket::AudioCodec* const kAudioCodecs[] = { - &kPcmuCodec, &kIsacCodec, &kCeltCodec, &kOpusCodec, &kG722CodecVoE, - &kRedCodec, &kCn8000Codec, &kCn16000Codec, &kTelephoneEventCodec, + &kPcmuCodec, &kIsacCodec, &kCeltCodec, &kOpusCodec, &kRedCodec, + &kCn8000Codec, &kCn16000Codec, &kTelephoneEventCodec, }; const char kRingbackTone[] = "RIFF____WAVE____ABCD1234"; static uint32 kSsrc1 = 0x99; @@ -772,20 +770,6 @@ TEST_F(WebRtcVoiceEngineTestFake, DontResetSetSendCodec) { EXPECT_EQ(1, voe_.GetNumSetSendCodecs()); } -// Verify that G722 is set with 16000 samples per second to WebRTC. -TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecG722) { - EXPECT_TRUE(SetupEngine()); - int channel_num = voe_.GetLastChannel(); - std::vector<cricket::AudioCodec> codecs; - codecs.push_back(kG722CodecSdp); - EXPECT_TRUE(channel_->SetSendCodecs(codecs)); - webrtc::CodecInst gcodec; - EXPECT_EQ(0, voe_.GetSendCodec(channel_num, gcodec)); - EXPECT_STREQ("G722", gcodec.plname); - EXPECT_EQ(1, gcodec.channels); - EXPECT_EQ(16000, gcodec.plfreq); -} - // Test that if clockrate is not 48000 for opus, we fail. TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecOpusBadClockrate) { EXPECT_TRUE(SetupEngine()); @@ -3224,7 +3208,7 @@ TEST(WebRtcVoiceEngineTest, HasCorrectCodecs) { EXPECT_TRUE(engine.FindCodec( cricket::AudioCodec(96, "PCMA", 8000, 0, 1, 0))); EXPECT_TRUE(engine.FindCodec( - cricket::AudioCodec(96, "G722", 8000, 0, 1, 0))); + cricket::AudioCodec(96, "G722", 16000, 0, 1, 0))); EXPECT_TRUE(engine.FindCodec( cricket::AudioCodec(96, "red", 8000, 0, 1, 0))); EXPECT_TRUE(engine.FindCodec( @@ -3241,7 +3225,7 @@ TEST(WebRtcVoiceEngineTest, HasCorrectCodecs) { EXPECT_TRUE(engine.FindCodec( cricket::AudioCodec(8, "", 8000, 0, 1, 0))); // PCMA EXPECT_TRUE(engine.FindCodec( - cricket::AudioCodec(9, "", 8000, 0, 1, 0))); // G722 + cricket::AudioCodec(9, "", 16000, 0, 1, 0))); // G722 EXPECT_TRUE(engine.FindCodec( cricket::AudioCodec(13, "", 8000, 0, 1, 0))); // CN // Check sample/bitrate matching. @@ -3264,7 +3248,7 @@ TEST(WebRtcVoiceEngineTest, HasCorrectCodecs) { EXPECT_EQ(103, it->id); } else if (it->name == "ISAC" && it->clockrate == 32000) { EXPECT_EQ(104, it->id); - } else if (it->name == "G722" && it->clockrate == 8000) { + } else if (it->name == "G722" && it->clockrate == 16000) { EXPECT_EQ(9, it->id); } else if (it->name == "telephone-event") { EXPECT_EQ(126, it->id); |