Deprecate RtpReceiver's SetParameters method

This removes the SetParameters method from AudioRtpReceiver and Video
RtpReceiver, which is currently not used and is not part of the
specifications.


Bug: webrtc:11111
Change-Id: I6f67773bfef2d4b51e9ab670bde17b5fbf5f94c3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159307
Reviewed-by: Patrik Höglund <phoglund@google.com>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Daniela Jovanoska Petrenko <denicija@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Saurav Das <dinosaurav@chromium.org>
Cr-Commit-Position: refs/heads/master@{#29995}

api/rtp_receiver_interface.h

@@ -76,8 +76,9 @@ class RTC_EXPORT RtpReceiverInterface : public rtc::RefCountInterface {
   // but this API also applies them to receivers, similar to ORTC:
   // http://ortc.org/wp-content/uploads/2016/03/ortc.html#rtcrtpparameters*.
   virtual RtpParameters GetParameters() const = 0;
-  // Currently, doesn't support changing any parameters, but may in the future.
-  virtual bool SetParameters(const RtpParameters& parameters) = 0;
+  // TODO(dinosaurav): Delete SetParameters entirely after rolling to Chromium.
+  // Currently, doesn't support changing any parameters.
+  virtual bool SetParameters(const RtpParameters& parameters) { return false; }
 
   // Does not take ownership of observer.
   // Must call SetObserver(nullptr) before the observer is destroyed.
@@ -123,7 +124,6 @@ PROXY_CONSTMETHOD0(std::vector<rtc::scoped_refptr<MediaStreamInterface>>,
 PROXY_CONSTMETHOD0(cricket::MediaType, media_type)
 PROXY_CONSTMETHOD0(std::string, id)
 PROXY_CONSTMETHOD0(RtpParameters, GetParameters)
-PROXY_METHOD1(bool, SetParameters, const RtpParameters&)
 PROXY_METHOD1(void, SetObserver, RtpReceiverObserverInterface*)
 PROXY_METHOD1(void, SetJitterBufferMinimumDelay, absl::optional<double>)
 PROXY_CONSTMETHOD0(std::vector<RtpSource>, GetSources)

api/test/mock_rtpreceiver.h

@@ -28,7 +28,6 @@ class MockRtpReceiver : public rtc::RefCountedObject<RtpReceiverInterface> {
   MOCK_CONST_METHOD0(media_type, cricket::MediaType());
   MOCK_CONST_METHOD0(id, std::string());
   MOCK_CONST_METHOD0(GetParameters, RtpParameters());
-  MOCK_METHOD1(SetParameters, bool(const RtpParameters&));
   MOCK_METHOD1(SetObserver, void(RtpReceiverObserverInterface*));
   MOCK_METHOD1(SetJitterBufferMinimumDelay, void(absl::optional<double>));
   MOCK_CONST_METHOD0(GetSources, std::vector<RtpSource>());

media/base/fake_media_engine.h

@@ -168,18 +168,6 @@ class RtpHelper : public Base {
     }
     return webrtc::RtpParameters();
   }
-  virtual bool SetRtpReceiveParameters(
-      uint32_t ssrc,
-      const webrtc::RtpParameters& parameters) {
-    auto parameters_iterator = rtp_receive_parameters_.find(ssrc);
-    if (parameters_iterator != rtp_receive_parameters_.end()) {
-      parameters_iterator->second = parameters;
-      return true;
-    }
-    // Replicate the behavior of the real media channel: return false
-    // when setting parameters for unknown SSRCs.
-    return false;
-  }
 
   bool IsStreamMuted(uint32_t ssrc) const {
     bool ret = muted_streams_.find(ssrc) != muted_streams_.end();

media/base/media_channel.h

@@ -806,9 +806,6 @@ class VoiceMediaChannel : public MediaChannel, public Delayable {
   // member.
   virtual webrtc::RtpParameters GetRtpReceiveParameters(
       uint32_t ssrc) const = 0;
-  virtual bool SetRtpReceiveParameters(
-      uint32_t ssrc,
-      const webrtc::RtpParameters& parameters) = 0;
   // Starts or stops playout of received audio.
   virtual void SetPlayout(bool playout) = 0;
   // Starts or stops sending (and potentially capture) of local audio.
@@ -875,9 +872,6 @@ class VideoMediaChannel : public MediaChannel, public Delayable {
   // member.
   virtual webrtc::RtpParameters GetRtpReceiveParameters(
       uint32_t ssrc) const = 0;
-  virtual bool SetRtpReceiveParameters(
-      uint32_t ssrc,
-      const webrtc::RtpParameters& parameters) = 0;
   // Gets the currently set codecs/payload types to be used for outgoing media.
   virtual bool GetSendCodec(VideoCodec* send_codec) = 0;
   // Starts or stops transmission (and potentially capture) of local video.

media/engine/webrtc_video_engine.cc

@@ -952,40 +952,6 @@ webrtc::RtpParameters WebRtcVideoChannel::GetRtpReceiveParameters(
   return rtp_params;
 }
 
-bool WebRtcVideoChannel::SetRtpReceiveParameters(
-    uint32_t ssrc,
-    const webrtc::RtpParameters& parameters) {
-  RTC_DCHECK_RUN_ON(&thread_checker_);
-  TRACE_EVENT0("webrtc", "WebRtcVideoChannel::SetRtpReceiveParameters");
-
-  // SSRC of 0 represents an unsignaled receive stream.
-  if (ssrc == 0) {
-    if (!default_unsignalled_ssrc_handler_.GetDefaultSink()) {
-      RTC_LOG(LS_WARNING)
-          << "Attempting to set RTP parameters for the default, "
-             "unsignaled video receive stream, but not yet "
-             "configured to receive such a stream.";
-      return false;
-    }
-  } else {
-    auto it = receive_streams_.find(ssrc);
-    if (it == receive_streams_.end()) {
-      RTC_LOG(LS_WARNING)
-          << "Attempting to set RTP receive parameters for stream "
-          << "with SSRC " << ssrc << " which doesn't exist.";
-      return false;
-    }
-  }
-
-  webrtc::RtpParameters current_parameters = GetRtpReceiveParameters(ssrc);
-  if (current_parameters != parameters) {
-    RTC_DLOG(LS_ERROR) << "Changing the RTP receive parameters is currently "
-                       << "unsupported.";
-    return false;
-  }
-  return true;
-}
-
 bool WebRtcVideoChannel::GetChangedRecvParameters(
     const VideoRecvParameters& params,
     ChangedRecvParameters* changed_params) const {

media/engine/webrtc_video_engine.h

@@ -129,9 +129,6 @@ class WebRtcVideoChannel : public VideoMediaChannel,
       uint32_t ssrc,
       const webrtc::RtpParameters& parameters) override;
   webrtc::RtpParameters GetRtpReceiveParameters(uint32_t ssrc) const override;
-  bool SetRtpReceiveParameters(
-      uint32_t ssrc,
-      const webrtc::RtpParameters& parameters) override;
   bool GetSendCodec(VideoCodec* send_codec) override;
   bool SetSend(bool send) override;
   bool SetVideoSend(

media/engine/webrtc_video_engine_unittest.cc

@@ -7328,9 +7328,6 @@ TEST_F(WebRtcVideoChannelTest, SetAndGetRtpReceiveParameters) {
   webrtc::RtpParameters initial_params =
       channel_->GetRtpReceiveParameters(last_ssrc_);
 
-  // We should be able to set the params we just got.
-  EXPECT_TRUE(channel_->SetRtpReceiveParameters(last_ssrc_, initial_params));
-
   // ... And this shouldn't change the params returned by
   // GetRtpReceiveParameters.
   EXPECT_EQ(initial_params, channel_->GetRtpReceiveParameters(last_ssrc_));

media/engine/webrtc_voice_engine.cc

@@ -1450,38 +1450,6 @@ webrtc::RtpParameters WebRtcVoiceMediaChannel::GetRtpReceiveParameters(
   return rtp_params;
 }
 
-bool WebRtcVoiceMediaChannel::SetRtpReceiveParameters(
-    uint32_t ssrc,
-    const webrtc::RtpParameters& parameters) {
-  RTC_DCHECK(worker_thread_checker_.IsCurrent());
-  // SSRC of 0 represents the default receive stream.
-  if (ssrc == 0) {
-    if (!default_sink_) {
-      RTC_LOG(LS_WARNING)
-          << "Attempting to set RTP parameters for the default, "
-             "unsignaled audio receive stream, but not yet "
-             "configured to receive such a stream.";
-      return false;
-    }
-  } else {
-    auto it = recv_streams_.find(ssrc);
-    if (it == recv_streams_.end()) {
-      RTC_LOG(LS_WARNING)
-          << "Attempting to set RTP receive parameters for stream "
-          << "with ssrc " << ssrc << " which doesn't exist.";
-      return false;
-    }
-  }
-
-  webrtc::RtpParameters current_parameters = GetRtpReceiveParameters(ssrc);
-  if (current_parameters != parameters) {
-    RTC_DLOG(LS_ERROR) << "Changing the RTP receive parameters is currently "
-                       << "unsupported.";
-    return false;
-  }
-  return true;
-}
-
 bool WebRtcVoiceMediaChannel::SetOptions(const AudioOptions& options) {
   RTC_DCHECK(worker_thread_checker_.IsCurrent());
   RTC_LOG(LS_INFO) << "Setting voice channel options: " << options.ToString();

media/engine/webrtc_voice_engine.h

@@ -151,9 +151,6 @@ class WebRtcVoiceMediaChannel final : public VoiceMediaChannel,
       uint32_t ssrc,
       const webrtc::RtpParameters& parameters) override;
   webrtc::RtpParameters GetRtpReceiveParameters(uint32_t ssrc) const override;
-  bool SetRtpReceiveParameters(
-      uint32_t ssrc,
-      const webrtc::RtpParameters& parameters) override;
 
   void SetPlayout(bool playout) override;
   void SetSend(bool send) override;

media/engine/webrtc_voice_engine_unittest.cc

@@ -1373,9 +1373,6 @@ TEST_F(WebRtcVoiceEngineTestFake, SetAndGetRtpReceiveParameters) {
   webrtc::RtpParameters initial_params =
       channel_->GetRtpReceiveParameters(kSsrcX);
 
-  // We should be able to set the params we just got.
-  EXPECT_TRUE(channel_->SetRtpReceiveParameters(kSsrcX, initial_params));
-
   // ... And this shouldn't change the params returned by
   // GetRtpReceiveParameters.
   webrtc::RtpParameters new_params = channel_->GetRtpReceiveParameters(kSsrcX);

pc/audio_rtp_receiver.cc

@@ -117,17 +117,6 @@ RtpParameters AudioRtpReceiver::GetParameters() const {
   });
 }
 
-bool AudioRtpReceiver::SetParameters(const RtpParameters& parameters) {
-  TRACE_EVENT0("webrtc", "AudioRtpReceiver::SetParameters");
-  if (!media_channel_ || stopped_) {
-    return false;
-  }
-  return worker_thread_->Invoke<bool>(RTC_FROM_HERE, [&] {
-    return media_channel_->SetRtpReceiveParameters(ssrc_.value_or(0),
-                                                   parameters);
-  });
-}
-
 void AudioRtpReceiver::SetFrameDecryptor(
     rtc::scoped_refptr<FrameDecryptorInterface> frame_decryptor) {
   frame_decryptor_ = std::move(frame_decryptor);

pc/audio_rtp_receiver.h

@@ -75,7 +75,6 @@ class AudioRtpReceiver : public ObserverInterface,
   std::string id() const override { return id_; }
 
   RtpParameters GetParameters() const override;
-  bool SetParameters(const RtpParameters& parameters) override;
 
   void SetFrameDecryptor(
       rtc::scoped_refptr<FrameDecryptorInterface> frame_decryptor) override;

pc/rtp_sender_receiver_unittest.cc

@@ -1339,26 +1339,6 @@ TEST_F(RtpSenderReceiverTest, SetVideoBitratePriority) {
   DestroyVideoRtpSender();
 }
 
-TEST_F(RtpSenderReceiverTest, AudioReceiverCanSetParameters) {
-  CreateAudioRtpReceiver();
-
-  RtpParameters params = audio_rtp_receiver_->GetParameters();
-  EXPECT_EQ(1u, params.encodings.size());
-  EXPECT_TRUE(audio_rtp_receiver_->SetParameters(params));
-
-  DestroyAudioRtpReceiver();
-}
-
-TEST_F(RtpSenderReceiverTest, VideoReceiverCanSetParameters) {
-  CreateVideoRtpReceiver();
-
-  RtpParameters params = video_rtp_receiver_->GetParameters();
-  EXPECT_EQ(1u, params.encodings.size());
-  EXPECT_TRUE(video_rtp_receiver_->SetParameters(params));
-
-  DestroyVideoRtpReceiver();
-}
-
 TEST_F(RtpSenderReceiverTest, VideoReceiverCanGetParametersWithSimulcast) {
   CreateVideoRtpReceiverWithSimulcast({}, 2);
 

pc/test/mock_rtp_receiver_internal.h

@@ -34,7 +34,6 @@ class MockRtpReceiverInternal : public RtpReceiverInternal {
   MOCK_CONST_METHOD0(media_type, cricket::MediaType());
   MOCK_CONST_METHOD0(id, std::string());
   MOCK_CONST_METHOD0(GetParameters, RtpParameters());
-  MOCK_METHOD1(SetParameters, bool(const RtpParameters&));
   MOCK_METHOD1(SetObserver, void(RtpReceiverObserverInterface*));
   MOCK_METHOD1(SetJitterBufferMinimumDelay, void(absl::optional<double>));
   MOCK_CONST_METHOD0(GetSources, std::vector<RtpSource>());

pc/video_rtp_receiver.cc

@@ -97,18 +97,6 @@ RtpParameters VideoRtpReceiver::GetParameters() const {
   });
 }
 
-bool VideoRtpReceiver::SetParameters(const RtpParameters& parameters) {
-  TRACE_EVENT0("webrtc", "VideoRtpReceiver::SetParameters");
-  if (!media_channel_ || stopped_) {
-    return false;
-  }
-  return worker_thread_->Invoke<bool>(RTC_FROM_HERE, [&] {
-    // TODO(bugs.webrtc.org/8694): Stop using 0 to mean unsignalled SSRC
-    return media_channel_->SetRtpReceiveParameters(ssrc_.value_or(0),
-                                                   parameters);
-  });
-}
-
 void VideoRtpReceiver::SetFrameDecryptor(
     rtc::scoped_refptr<FrameDecryptorInterface> frame_decryptor) {
   frame_decryptor_ = std::move(frame_decryptor);

pc/video_rtp_receiver.h

@@ -76,7 +76,6 @@ class VideoRtpReceiver : public rtc::RefCountedObject<RtpReceiverInternal>,
   std::string id() const override { return id_; }
 
   RtpParameters GetParameters() const override;
-  bool SetParameters(const RtpParameters& parameters) override;
 
   void SetFrameDecryptor(
       rtc::scoped_refptr<FrameDecryptorInterface> frame_decryptor) override;

sdk/android/api/org/webrtc/RtpReceiver.java

@@ -39,11 +39,6 @@ public class RtpReceiver {
     return cachedTrack;
   }
 
-  public boolean setParameters(@Nullable RtpParameters parameters) {
-    checkRtpReceiverExists();
-    return parameters == null ? false : nativeSetParameters(nativeRtpReceiver, parameters);
-  }
-
   public RtpParameters getParameters() {
     checkRtpReceiverExists();
     return nativeGetParameters(nativeRtpReceiver);
@@ -89,7 +84,6 @@ public class RtpReceiver {
   // This should increment the reference count of the track.
   // Will be released in dispose().
   private static native long nativeGetTrack(long rtpReceiver);
-  private static native boolean nativeSetParameters(long rtpReceiver, RtpParameters parameters);
   private static native RtpParameters nativeGetParameters(long rtpReceiver);
   private static native String nativeGetId(long rtpReceiver);
   private static native long nativeSetObserver(long rtpReceiver, Observer observer);

sdk/android/src/jni/pc/rtp_receiver.cc

@@ -74,15 +74,6 @@ static jlong JNI_RtpReceiver_GetTrack(JNIEnv* jni,
           .release());
 }
 
-static jboolean JNI_RtpReceiver_SetParameters(
-    JNIEnv* jni,
-    jlong j_rtp_receiver_pointer,
-    const JavaParamRef<jobject>& j_parameters) {
-  RtpParameters parameters = JavaToNativeRtpParameters(jni, j_parameters);
-  return reinterpret_cast<RtpReceiverInterface*>(j_rtp_receiver_pointer)
-      ->SetParameters(parameters);
-}
-
 static ScopedJavaLocalRef<jobject> JNI_RtpReceiver_GetParameters(
     JNIEnv* jni,
     jlong j_rtp_receiver_pointer) {

sdk/objc/api/peerconnection/RTCRtpReceiver.mm

@@ -53,13 +53,6 @@ void RtpReceiverDelegateAdapter::OnFirstPacketReceived(
       initWithNativeParameters:_nativeRtpReceiver->GetParameters()];
 }
 
-- (void)setParameters:(RTCRtpParameters *)parameters {
-  if (!_nativeRtpReceiver->SetParameters(parameters.nativeParameters)) {
-    RTCLogError(@"RTCRtpReceiver(%p): Failed to set parameters: %@", self,
-        parameters);
-  }
-}
-
 - (nullable RTCMediaStreamTrack *)track {
   rtc::scoped_refptr<webrtc::MediaStreamTrackInterface> nativeTrack(
     _nativeRtpReceiver->track());