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}
diff --git a/api/rtp_receiver_interface.h b/api/rtp_receiver_interface.h
index b01e07d..6052763 100644
--- a/api/rtp_receiver_interface.h
+++ b/api/rtp_receiver_interface.h
@@ -76,8 +76,9 @@
// 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(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)
diff --git a/api/test/mock_rtpreceiver.h b/api/test/mock_rtpreceiver.h
index 710f8c5..d4da908 100644
--- a/api/test/mock_rtpreceiver.h
+++ b/api/test/mock_rtpreceiver.h
@@ -28,7 +28,6 @@
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>());
diff --git a/media/base/fake_media_engine.h b/media/base/fake_media_engine.h
index ac303e6..99a9b49 100644
--- a/media/base/fake_media_engine.h
+++ b/media/base/fake_media_engine.h
@@ -168,18 +168,6 @@
}
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();
diff --git a/media/base/media_channel.h b/media/base/media_channel.h
index 185c883..696e5f7 100644
--- a/media/base/media_channel.h
+++ b/media/base/media_channel.h
@@ -806,9 +806,6 @@
// 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 @@
// 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.
diff --git a/media/engine/webrtc_video_engine.cc b/media/engine/webrtc_video_engine.cc
index cab8e29..71d0c9b 100644
--- a/media/engine/webrtc_video_engine.cc
+++ b/media/engine/webrtc_video_engine.cc
@@ -952,40 +952,6 @@
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 {
diff --git a/media/engine/webrtc_video_engine.h b/media/engine/webrtc_video_engine.h
index 2493edb..20461ba 100644
--- a/media/engine/webrtc_video_engine.h
+++ b/media/engine/webrtc_video_engine.h
@@ -129,9 +129,6 @@
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(
diff --git a/media/engine/webrtc_video_engine_unittest.cc b/media/engine/webrtc_video_engine_unittest.cc
index 2c49c87..d07042c 100644
--- a/media/engine/webrtc_video_engine_unittest.cc
+++ b/media/engine/webrtc_video_engine_unittest.cc
@@ -7328,9 +7328,6 @@
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_));
diff --git a/media/engine/webrtc_voice_engine.cc b/media/engine/webrtc_voice_engine.cc
index eab2bc8..cfbd774 100644
--- a/media/engine/webrtc_voice_engine.cc
+++ b/media/engine/webrtc_voice_engine.cc
@@ -1450,38 +1450,6 @@
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();
diff --git a/media/engine/webrtc_voice_engine.h b/media/engine/webrtc_voice_engine.h
index add587f..3c46980 100644
--- a/media/engine/webrtc_voice_engine.h
+++ b/media/engine/webrtc_voice_engine.h
@@ -151,9 +151,6 @@
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;
diff --git a/media/engine/webrtc_voice_engine_unittest.cc b/media/engine/webrtc_voice_engine_unittest.cc
index 775d586..d509831 100644
--- a/media/engine/webrtc_voice_engine_unittest.cc
+++ b/media/engine/webrtc_voice_engine_unittest.cc
@@ -1373,9 +1373,6 @@
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);
diff --git a/pc/audio_rtp_receiver.cc b/pc/audio_rtp_receiver.cc
index d67b249..6c0445a 100644
--- a/pc/audio_rtp_receiver.cc
+++ b/pc/audio_rtp_receiver.cc
@@ -117,17 +117,6 @@
});
}
-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);
diff --git a/pc/audio_rtp_receiver.h b/pc/audio_rtp_receiver.h
index e1b1888..908cb64 100644
--- a/pc/audio_rtp_receiver.h
+++ b/pc/audio_rtp_receiver.h
@@ -75,7 +75,6 @@
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;
diff --git a/pc/rtp_sender_receiver_unittest.cc b/pc/rtp_sender_receiver_unittest.cc
index b9c07ef..2795e6b 100644
--- a/pc/rtp_sender_receiver_unittest.cc
+++ b/pc/rtp_sender_receiver_unittest.cc
@@ -1339,26 +1339,6 @@
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);
diff --git a/pc/test/mock_rtp_receiver_internal.h b/pc/test/mock_rtp_receiver_internal.h
index f854e33..ffe78b5 100644
--- a/pc/test/mock_rtp_receiver_internal.h
+++ b/pc/test/mock_rtp_receiver_internal.h
@@ -34,7 +34,6 @@
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>());
diff --git a/pc/video_rtp_receiver.cc b/pc/video_rtp_receiver.cc
index d9d2a2e..34e03b4 100644
--- a/pc/video_rtp_receiver.cc
+++ b/pc/video_rtp_receiver.cc
@@ -97,18 +97,6 @@
});
}
-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);
diff --git a/pc/video_rtp_receiver.h b/pc/video_rtp_receiver.h
index 0bb54e7..16b94b5 100644
--- a/pc/video_rtp_receiver.h
+++ b/pc/video_rtp_receiver.h
@@ -76,7 +76,6 @@
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;
diff --git a/sdk/android/api/org/webrtc/RtpReceiver.java b/sdk/android/api/org/webrtc/RtpReceiver.java
index 8564d47..015d35a 100644
--- a/sdk/android/api/org/webrtc/RtpReceiver.java
+++ b/sdk/android/api/org/webrtc/RtpReceiver.java
@@ -39,11 +39,6 @@
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 @@
// 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);
diff --git a/sdk/android/src/jni/pc/rtp_receiver.cc b/sdk/android/src/jni/pc/rtp_receiver.cc
index 15abe8d..4d7e954 100644
--- a/sdk/android/src/jni/pc/rtp_receiver.cc
+++ b/sdk/android/src/jni/pc/rtp_receiver.cc
@@ -74,15 +74,6 @@
.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) {
diff --git a/sdk/objc/api/peerconnection/RTCRtpReceiver.mm b/sdk/objc/api/peerconnection/RTCRtpReceiver.mm
index 5d44478..deeb4cb 100644
--- a/sdk/objc/api/peerconnection/RTCRtpReceiver.mm
+++ b/sdk/objc/api/peerconnection/RTCRtpReceiver.mm
@@ -53,13 +53,6 @@
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());
