1 files changed, 40 insertions, 7 deletions

diff --git a/media/webrtc/fakewebrtcvoiceengine.h b/media/webrtc/fakewebrtcvoiceengine.h
index f731b8d..52a50ff 100644
--- a/media/webrtc/fakewebrtcvoiceengine.h
+++ b/media/webrtc/fakewebrtcvoiceengine.h
@@ -43,10 +43,7 @@
 #ifdef USE_WEBRTC_DEV_BRANCH
 #include "webrtc/modules/audio_processing/include/audio_processing.h"
 #endif
-
-namespace webrtc {
-class ViENetwork;
-}
+#include "webrtc/video_engine/include/vie_network.h"
 
 namespace cricket {
 
@@ -64,6 +61,12 @@ static const int kFakeDeviceId = 0;
 static const int kFakeDeviceId = 1;
 #endif
 
+static const int kOpusBandwidthNb = 4000;
+static const int kOpusBandwidthMb = 6000;
+static const int kOpusBandwidthWb = 8000;
+static const int kOpusBandwidthSwb = 12000;
+static const int kOpusBandwidthFb = 20000;
+
 // Verify the header extension ID, if enabled, is within the bounds specified in
 // [RFC5285]: 1-14 inclusive.
 #define WEBRTC_CHECK_HEADER_EXTENSION_ID(enable, id) \
@@ -183,6 +186,7 @@ class FakeWebRtcVoiceEngine
           file(false),
           vad(false),
           codec_fec(false),
+          max_encoding_bandwidth(0),
           red(false),
           nack(false),
           media_processor_registered(false),
@@ -212,6 +216,7 @@ class FakeWebRtcVoiceEngine
     bool file;
     bool vad;
     bool codec_fec;
+    int max_encoding_bandwidth;
     bool red;
     bool nack;
     bool media_processor_registered;
@@ -308,6 +313,9 @@ class FakeWebRtcVoiceEngine
   bool GetCodecFEC(int channel) {
     return channels_[channel]->codec_fec;
   }
+  int GetMaxEncodingBandwidth(int channel) {
+    return channels_[channel]->max_encoding_bandwidth;
+  }
   bool GetNACK(int channel) {
     return channels_[channel]->nack;
   }
@@ -316,6 +324,8 @@ class FakeWebRtcVoiceEngine
   }
   webrtc::ViENetwork* GetViENetwork(int channel) {
     WEBRTC_ASSERT_CHANNEL(channel);
+    // WARNING: This pointer is for verification purposes only. Calling
+    // functions on it may result in undefined behavior!
     return channels_[channel]->vie_network;
   }
   int GetVideoChannel(int channel) {
@@ -488,8 +498,6 @@ class FakeWebRtcVoiceEngine
   WEBRTC_STUB(LastError, ());
   WEBRTC_STUB(SetOnHoldStatus, (int, bool, webrtc::OnHoldModes));
   WEBRTC_STUB(GetOnHoldStatus, (int, bool&, webrtc::OnHoldModes&));
-  WEBRTC_STUB(SetNetEQPlayoutMode, (int, webrtc::NetEqModes));
-  WEBRTC_STUB(GetNetEQPlayoutMode, (int, webrtc::NetEqModes&));
 
   // webrtc::VoECodec
   WEBRTC_FUNC(NumOfCodecs, ()) {
@@ -625,10 +633,11 @@ class FakeWebRtcVoiceEngine
   }
   WEBRTC_STUB(GetVADStatus, (int channel, bool& enabled,
                              webrtc::VadModes& mode, bool& disabledDTX));
+
 #ifdef USE_WEBRTC_DEV_BRANCH
   WEBRTC_FUNC(SetFECStatus, (int channel, bool enable)) {
     WEBRTC_CHECK_CHANNEL(channel);
-    if (strcmp(channels_[channel]->send_codec.plname, "opus")) {
+    if (_stricmp(channels_[channel]->send_codec.plname, "opus") != 0) {
       // Return -1 if current send codec is not Opus.
       // TODO(minyue): Excludes other codecs if they support inband FEC.
       return -1;
@@ -641,6 +650,25 @@ class FakeWebRtcVoiceEngine
     enable = channels_[channel]->codec_fec;
     return 0;
   }
+
+  WEBRTC_FUNC(SetOpusMaxPlaybackRate, (int channel, int frequency_hz)) {
+    WEBRTC_CHECK_CHANNEL(channel);
+    if (_stricmp(channels_[channel]->send_codec.plname, "opus") != 0) {
+      // Return -1 if current send codec is not Opus.
+      return -1;
+    }
+    if (frequency_hz <= 8000)
+      channels_[channel]->max_encoding_bandwidth = kOpusBandwidthNb;
+    else if (frequency_hz <= 12000)
+      channels_[channel]->max_encoding_bandwidth = kOpusBandwidthMb;
+    else if (frequency_hz <= 16000)
+      channels_[channel]->max_encoding_bandwidth = kOpusBandwidthWb;
+    else if (frequency_hz <= 24000)
+      channels_[channel]->max_encoding_bandwidth = kOpusBandwidthSwb;
+    else
+      channels_[channel]->max_encoding_bandwidth = kOpusBandwidthFb;
+    return 0;
+  }
 #endif  // USE_WEBRTC_DEV_BRANCH
 
   // webrtc::VoEDtmf
@@ -999,6 +1027,11 @@ class FakeWebRtcVoiceEngine
     WEBRTC_CHECK_CHANNEL(channel);
     channels_[channel]->vie_network = vie_network;
     channels_[channel]->video_channel = video_channel;
+    if (vie_network) {
+      // The interface is released here to avoid leaks. A test should not
+      // attempt to call functions on the interface stored in the channel.
+      vie_network->Release();
+    }
     return 0;
   }