Allow applications to control audio send bitrate through RtpParameters.

webrtc/api/webrtcsession_unittest.cc

 /*
  *  Copyright 2012 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.
  */
 
@@ skipping 3367 matching lines @@
   EXPECT_TRUE(channel->GetOutputVolume(receive_ssrc, &volume));
   EXPECT_EQ(1, volume);
   session_->SetAudioPlayout(receive_ssrc, false);
   EXPECT_TRUE(channel->GetOutputVolume(receive_ssrc, &volume));
   EXPECT_EQ(0, volume);
   session_->SetAudioPlayout(receive_ssrc, true);
   EXPECT_TRUE(channel->GetOutputVolume(receive_ssrc, &volume));
   EXPECT_EQ(1, volume);
 }
 
-TEST_F(WebRtcSessionTest, AudioMaxSendBitrateNotImplemented) {
-  // This test verifies that RtpParameters for audio RtpSenders cannot be
-  // changed.
-  // TODO(skvlad): Update the test after adding support for bitrate limiting in
-  // WebRtcAudioSendStream.
-
+TEST_F(WebRtcSessionTest, SetAudioMaxSendBitrate) {
   Init();
   SendAudioVideoStream1();
   CreateAndSetRemoteOfferAndLocalAnswer();
   cricket::FakeVoiceMediaChannel* channel = media_engine_->GetVoiceChannel(0);
   ASSERT_TRUE(channel != NULL);
   uint32_t send_ssrc = channel->send_streams()[0].first_ssrc();
+  EXPECT_EQ(-1, channel->max_bps());
   webrtc::RtpParameters params = session_->GetAudioRtpParameters(send_ssrc);
+  EXPECT_EQ(1, params.encodings.size());

[pthatcher1, 2016/04/12 18:04:23]

This should be ASSERT_EQ since the next line may crash if it's false.

+  EXPECT_EQ(-1, params.encodings[0].max_bitrate_bps);
+  params.encodings[0].max_bitrate_bps = 1000;
+  EXPECT_TRUE(session_->SetAudioRtpParameters(send_ssrc, params));
 
-  EXPECT_EQ(0, params.encodings.size());
-  params.encodings.push_back(webrtc::RtpEncodingParameters());
-  EXPECT_FALSE(session_->SetAudioRtpParameters(send_ssrc, params));
+  // Read back the parameters and verify they have been changed.
+  params = session_->GetAudioRtpParameters(send_ssrc);
+  EXPECT_EQ(1, params.encodings.size());
+  EXPECT_EQ(1000, params.encodings[0].max_bitrate_bps);
+
+  // Verify that the audio channel received the new parameters.
+  params = channel->GetRtpParameters(send_ssrc);
+  EXPECT_EQ(1, params.encodings.size());
+  EXPECT_EQ(1000, params.encodings[0].max_bitrate_bps);
+
+  // Verify that the global bitrate limit has not been changed.
+  EXPECT_EQ(-1, channel->max_bps());
 }
 
 TEST_F(WebRtcSessionTest, SetAudioSend) {
   Init();
   SendAudioVideoStream1();
   CreateAndSetRemoteOfferAndLocalAnswer();
   cricket::FakeVoiceMediaChannel* channel = media_engine_->GetVoiceChannel(0);
   ASSERT_TRUE(channel != NULL);
   ASSERT_EQ(1u, channel->send_streams().size());
   uint32_t send_ssrc = channel->send_streams()[0].first_ssrc();
@@ skipping 979 matching lines @@
 }
 
 // TODO(bemasc): Add a TestIceStatesBundle with BUNDLE enabled.  That test
 // currently fails because upon disconnection and reconnection OnIceComplete is
 // called more than once without returning to IceGatheringGathering.
 
 INSTANTIATE_TEST_CASE_P(WebRtcSessionTests,
                         WebRtcSessionTest,
                         testing::Values(ALREADY_GENERATED,
                                         DTLS_IDENTITY_STORE));