commit db0cd9e774dbe91f41fea349fef995c2b5887c25
author Taylor Brandstetter <deadbeef@webrtc.org> Mon May 16 11:40:30 2016 -0700
committer Taylor Brandstetter <deadbeef@webrtc.org> Mon May 16 18:40:38 2016 +0000
tree 102260b5a7cf953fb1424e321b4067192017a63d
parent 8bce67b7451d2d75390410dd7bbc82078ccfb2c9

Adding getParameters/setParameters APIs to RtpReceiver.

This is similar to how a "receive" method is used to apply
RtpParameters to an RtpReceiver in ORTC. Currently, SetParameters
doesn't allow changing the parameters, so the main use of the API is
to retrieve the set of configured codecs. But other uses will likely
be made possible in the future.

R=glaznev@webrtc.org, pthatcher@webrtc.org, tkchin@webrtc.org

Review URL: https://codereview.webrtc.org/1917193008 .

Cr-Commit-Position: refs/heads/master@{#12761}

webrtc/api/rtpsenderreceiver_unittest.cc

diff --git a/webrtc/api/rtpsenderreceiver_unittest.cc b/webrtc/api/rtpsenderreceiver_unittest.cc
index f8b968f..3db9d5e 100644
--- a/webrtc/api/rtpsenderreceiver_unittest.cc
+++ b/webrtc/api/rtpsenderreceiver_unittest.cc
@@ -59,8 +59,12 @@
                     const cricket::AudioOptions& options,
                     cricket::AudioSource* source));
   MOCK_METHOD2(SetAudioPlayoutVolume, void(uint32_t ssrc, double volume));
-  MOCK_CONST_METHOD1(GetAudioRtpParameters, RtpParameters(uint32_t ssrc));
-  MOCK_METHOD2(SetAudioRtpParameters,
+  MOCK_CONST_METHOD1(GetAudioRtpSendParameters, RtpParameters(uint32_t ssrc));
+  MOCK_METHOD2(SetAudioRtpSendParameters,
+               bool(uint32_t ssrc, const RtpParameters&));
+  MOCK_CONST_METHOD1(GetAudioRtpReceiveParameters,
+                     RtpParameters(uint32_t ssrc));
+  MOCK_METHOD2(SetAudioRtpReceiveParameters,
                bool(uint32_t ssrc, const RtpParameters&));
 
   void SetRawAudioSink(
@@ -88,8 +92,12 @@
                     bool enable,
                     const cricket::VideoOptions* options));
 
-  MOCK_CONST_METHOD1(GetVideoRtpParameters, RtpParameters(uint32_t ssrc));
-  MOCK_METHOD2(SetVideoRtpParameters,
+  MOCK_CONST_METHOD1(GetVideoRtpSendParameters, RtpParameters(uint32_t ssrc));
+  MOCK_METHOD2(SetVideoRtpSendParameters,
+               bool(uint32_t ssrc, const RtpParameters&));
+  MOCK_CONST_METHOD1(GetVideoRtpReceiveParameters,
+                     RtpParameters(uint32_t ssrc));
+  MOCK_METHOD2(SetVideoRtpReceiveParameters,
                bool(uint32_t ssrc, const RtpParameters&));
 };
 
@@ -504,9 +512,9 @@
 TEST_F(RtpSenderReceiverTest, AudioSenderCanSetParameters) {
   CreateAudioRtpSender();
 
-  EXPECT_CALL(audio_provider_, GetAudioRtpParameters(kAudioSsrc))
+  EXPECT_CALL(audio_provider_, GetAudioRtpSendParameters(kAudioSsrc))
       .WillOnce(Return(RtpParameters()));
-  EXPECT_CALL(audio_provider_, SetAudioRtpParameters(kAudioSsrc, _))
+  EXPECT_CALL(audio_provider_, SetAudioRtpSendParameters(kAudioSsrc, _))
       .WillOnce(Return(true));
   RtpParameters params = audio_rtp_sender_->GetParameters();
   EXPECT_TRUE(audio_rtp_sender_->SetParameters(params));
@@ -517,9 +525,9 @@
 TEST_F(RtpSenderReceiverTest, VideoSenderCanSetParameters) {
   CreateVideoRtpSender();
 
-  EXPECT_CALL(video_provider_, GetVideoRtpParameters(kVideoSsrc))
+  EXPECT_CALL(video_provider_, GetVideoRtpSendParameters(kVideoSsrc))
       .WillOnce(Return(RtpParameters()));
-  EXPECT_CALL(video_provider_, SetVideoRtpParameters(kVideoSsrc, _))
+  EXPECT_CALL(video_provider_, SetVideoRtpSendParameters(kVideoSsrc, _))
       .WillOnce(Return(true));
   RtpParameters params = video_rtp_sender_->GetParameters();
   EXPECT_TRUE(video_rtp_sender_->SetParameters(params));
@@ -527,4 +535,30 @@
   DestroyVideoRtpSender();
 }
 
+TEST_F(RtpSenderReceiverTest, AudioReceiverCanSetParameters) {
+  CreateAudioRtpReceiver();
+
+  EXPECT_CALL(audio_provider_, GetAudioRtpReceiveParameters(kAudioSsrc))
+      .WillOnce(Return(RtpParameters()));
+  EXPECT_CALL(audio_provider_, SetAudioRtpReceiveParameters(kAudioSsrc, _))
+      .WillOnce(Return(true));
+  RtpParameters params = audio_rtp_receiver_->GetParameters();
+  EXPECT_TRUE(audio_rtp_receiver_->SetParameters(params));
+
+  DestroyAudioRtpReceiver();
+}
+
+TEST_F(RtpSenderReceiverTest, VideoReceiverCanSetParameters) {
+  CreateVideoRtpReceiver();
+
+  EXPECT_CALL(video_provider_, GetVideoRtpReceiveParameters(kVideoSsrc))
+      .WillOnce(Return(RtpParameters()));
+  EXPECT_CALL(video_provider_, SetVideoRtpReceiveParameters(kVideoSsrc, _))
+      .WillOnce(Return(true));
+  RtpParameters params = video_rtp_receiver_->GetParameters();
+  EXPECT_TRUE(video_rtp_receiver_->SetParameters(params));
+
+  DestroyVideoRtpReceiver();
+}
+
 }  // namespace webrtc