Enable setting the maximum bitrate limit in RtpSender.

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 8062fe5db75f7cbc56f58741114da65f9ece5f87..269d9425d25f8ba035f73dc4e11c45f7e617a591 100644
--- a/webrtc/api/webrtcsession_unittest.cc
+++ b/webrtc/api/webrtcsession_unittest.cc
@@ -3333,6 +3333,36 @@ TEST_F(WebRtcSessionTest, SetAudioPlayout) {
   EXPECT_EQ(1, volume);
 }
 
+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::RTCRtpParameters params = session_->GetAudioRtpParameters(send_ssrc);
+  EXPECT_EQ(1, params.encodings.size());
+  EXPECT_EQ(-1, params.encodings[0].max_bitrate_bps);
+  params.encodings[0].max_bitrate_bps = 1000;
+  EXPECT_TRUE(session_->SetAudioRtpParameters(send_ssrc, params));
+
+  // Verify that the media channel received the new parameters

[Taylor Brandstetter, 2016/03/12 01:57:06]

nit: Add periods at the end of comments like this.

[skvlad, 2016/03/15 21:18:18]

Done.

+  params = channel->GetRtpParameters(send_ssrc);
+  EXPECT_EQ(1, params.encodings.size());
+  EXPECT_EQ(1000, params.encodings[0].max_bitrate_bps);
+
+  // Verify that reading back the parameters returns the same
+  // value as what has been set previously
+  params = session_->GetAudioRtpParameters(send_ssrc);
+  EXPECT_EQ(1, params.encodings.size());
+  EXPECT_EQ(1000, params.encodings[0].max_bitrate_bps);
+
+  // Verify that the global bitrate limit (coming from SDP) has
+  // not been changed
+  EXPECT_EQ(-1, channel->max_bps());
+}
+
 TEST_F(WebRtcSessionTest, SetAudioSend) {
   Init();
   SendAudioVideoStream1();
@@ -3399,6 +3429,34 @@ TEST_F(WebRtcSessionTest, SetVideoPlayout) {
   EXPECT_TRUE(channel->sinks().begin()->second == NULL);
 }
 
+TEST_F(WebRtcSessionTest, SetVideoMaxSendBitrate) {
+  Init();
+  SendAudioVideoStream1();
+  CreateAndSetRemoteOfferAndLocalAnswer();
+  cricket::FakeVideoMediaChannel* channel = media_engine_->GetVideoChannel(0);
+  ASSERT_TRUE(channel != NULL);
+  uint32_t send_ssrc = channel->send_streams()[0].first_ssrc();
+  EXPECT_EQ(-1, channel->max_bps());
+  webrtc::RTCRtpParameters params = session_->GetVideoRtpParameters(send_ssrc);
+  EXPECT_EQ(1, params.encodings.size());
+  EXPECT_EQ(-1, params.encodings[0].max_bitrate_bps);
+  params.encodings[0].max_bitrate_bps = 1000;
+  EXPECT_TRUE(session_->SetVideoRtpParameters(send_ssrc, params));
+
+  // Read back the parameters and verify they have been changed
+  params = session_->GetVideoRtpParameters(send_ssrc);
+  EXPECT_EQ(1, params.encodings.size());
+  EXPECT_EQ(1000, params.encodings[0].max_bitrate_bps);
+
+  // Verify that the video 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, SetVideoSend) {
   Init();
   SendAudioVideoStream1();