Allow applications to control audio send bitrate through RtpParameters.

webrtc/api/webrtcsession_unittest.cc

Index: webrtc/api/webrtcsession_unittest.cc
diff --git a/webrtc/api/webrtcsession_unittest.cc b/webrtc/api/webrtcsession_unittest.cc
index 18c1a95116e5e812c0f784137c5fb28a463e895c..f4b41d355a52b10653201199d884719e75df599e 100644
--- a/webrtc/api/webrtcsession_unittest.cc
+++ b/webrtc/api/webrtcsession_unittest.cc
@@ -3385,23 +3385,32 @@ TEST_F(WebRtcSessionTest, SetAudioPlayout) {
   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) {