Enable SSRC 0 in MediaChannel methods
Refactor voice engine and video engine to use default methods instead of
treating 0 as a special value.
Bug: webrtc:8694
Change-Id: I47c211c6e870cdec737d6b0d05df29a9b534a011
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158600
Reviewed-by: Markus Handell <handellm@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Saurav Das <dinosaurav@chromium.org>
Cr-Commit-Position: refs/heads/master@{#30010}
diff --git a/media/base/fake_media_engine.cc b/media/base/fake_media_engine.cc
index d130234..c31ef97 100644
--- a/media/base/fake_media_engine.cc
+++ b/media/base/fake_media_engine.cc
@@ -147,18 +147,18 @@
return true;
}
bool FakeVoiceMediaChannel::SetOutputVolume(uint32_t ssrc, double volume) {
- if (0 == ssrc) {
- std::map<uint32_t, double>::iterator it;
- for (it = output_scalings_.begin(); it != output_scalings_.end(); ++it) {
- it->second = volume;
- }
- return true;
- } else if (output_scalings_.find(ssrc) != output_scalings_.end()) {
+ if (output_scalings_.find(ssrc) != output_scalings_.end()) {
output_scalings_[ssrc] = volume;
return true;
}
return false;
}
+bool FakeVoiceMediaChannel::SetDefaultOutputVolume(double volume) {
+ for (auto& entry : output_scalings_) {
+ entry.second = volume;
+ }
+ return true;
+}
bool FakeVoiceMediaChannel::GetOutputVolume(uint32_t ssrc, double* volume) {
if (output_scalings_.find(ssrc) == output_scalings_.end())
return false;
@@ -190,6 +190,10 @@
std::unique_ptr<webrtc::AudioSinkInterface> sink) {
sink_ = std::move(sink);
}
+void FakeVoiceMediaChannel::SetDefaultRawAudioSink(
+ std::unique_ptr<webrtc::AudioSinkInterface> sink) {
+ sink_ = std::move(sink);
+}
std::vector<webrtc::RtpSource> FakeVoiceMediaChannel::GetSources(
uint32_t ssrc) const {
return std::vector<webrtc::RtpSource>();
@@ -308,14 +312,15 @@
bool FakeVideoMediaChannel::SetSink(
uint32_t ssrc,
rtc::VideoSinkInterface<webrtc::VideoFrame>* sink) {
- if (ssrc != 0 && sinks_.find(ssrc) == sinks_.end()) {
+ auto it = sinks_.find(ssrc);
+ if (it == sinks_.end()) {
return false;
}
- if (ssrc != 0) {
- sinks_[ssrc] = sink;
- }
+ it->second = sink;
return true;
}
+void FakeVideoMediaChannel::SetDefaultSink(
+ rtc::VideoSinkInterface<webrtc::VideoFrame>* sink) {}
bool FakeVideoMediaChannel::HasSink(uint32_t ssrc) const {
return sinks_.find(ssrc) != sinks_.end() && sinks_.at(ssrc) != nullptr;
}
diff --git a/media/base/fake_media_engine.h b/media/base/fake_media_engine.h
index aa713d0..3df8f85 100644
--- a/media/base/fake_media_engine.h
+++ b/media/base/fake_media_engine.h
@@ -168,6 +168,9 @@
}
return webrtc::RtpParameters();
}
+ virtual webrtc::RtpParameters GetDefaultRtpReceiveParameters() const {
+ return webrtc::RtpParameters();
+ }
bool IsStreamMuted(uint32_t ssrc) const {
bool ret = muted_streams_.find(ssrc) != muted_streams_.end();
@@ -338,6 +341,8 @@
bool InsertDtmf(uint32_t ssrc, int event_code, int duration) override;
bool SetOutputVolume(uint32_t ssrc, double volume) override;
+ bool SetDefaultOutputVolume(double volume) override;
+
bool GetOutputVolume(uint32_t ssrc, double* volume);
bool SetBaseMinimumPlayoutDelayMs(uint32_t ssrc, int delay_ms) override;
@@ -349,6 +354,8 @@
void SetRawAudioSink(
uint32_t ssrc,
std::unique_ptr<webrtc::AudioSinkInterface> sink) override;
+ void SetDefaultRawAudioSink(
+ std::unique_ptr<webrtc::AudioSinkInterface> sink) override;
std::vector<webrtc::RtpSource> GetSources(uint32_t ssrc) const override;
@@ -415,6 +422,8 @@
bool GetSendCodec(VideoCodec* send_codec) override;
bool SetSink(uint32_t ssrc,
rtc::VideoSinkInterface<webrtc::VideoFrame>* sink) override;
+ void SetDefaultSink(
+ rtc::VideoSinkInterface<webrtc::VideoFrame>* sink) override;
bool HasSink(uint32_t ssrc) const;
bool SetSend(bool send) override;
diff --git a/media/base/media_channel.h b/media/base/media_channel.h
index be4730e..90c33bd 100644
--- a/media/base/media_channel.h
+++ b/media/base/media_channel.h
@@ -802,12 +802,11 @@
virtual bool SetSendParameters(const AudioSendParameters& params) = 0;
virtual bool SetRecvParameters(const AudioRecvParameters& params) = 0;
// Get the receive parameters for the incoming stream identified by |ssrc|.
- // If |ssrc| is 0, retrieve the receive parameters for the default receive
- // stream, which is used when SSRCs are not signaled. Note that calling with
- // an |ssrc| of 0 will return encoding parameters with an unset |ssrc|
- // member.
virtual webrtc::RtpParameters GetRtpReceiveParameters(
uint32_t ssrc) const = 0;
+ // Retrieve the receive parameters for the default receive
+ // stream, which is used when SSRCs are not signaled.
+ virtual webrtc::RtpParameters GetDefaultRtpReceiveParameters() const = 0;
// Starts or stops playout of received audio.
virtual void SetPlayout(bool playout) = 0;
// Starts or stops sending (and potentially capture) of local audio.
@@ -819,6 +818,8 @@
AudioSource* source) = 0;
// Set speaker output volume of the specified ssrc.
virtual bool SetOutputVolume(uint32_t ssrc, double volume) = 0;
+ // Set speaker output volume for future unsignaled streams.
+ virtual bool SetDefaultOutputVolume(double volume) = 0;
// Returns if the telephone-event has been negotiated.
virtual bool CanInsertDtmf() = 0;
// Send a DTMF |event|. The DTMF out-of-band signal will be used.
@@ -832,6 +833,8 @@
virtual void SetRawAudioSink(
uint32_t ssrc,
std::unique_ptr<webrtc::AudioSinkInterface> sink) = 0;
+ virtual void SetDefaultRawAudioSink(
+ std::unique_ptr<webrtc::AudioSinkInterface> sink) = 0;
virtual std::vector<webrtc::RtpSource> GetSources(uint32_t ssrc) const = 0;
};
@@ -868,12 +871,11 @@
virtual bool SetSendParameters(const VideoSendParameters& params) = 0;
virtual bool SetRecvParameters(const VideoRecvParameters& params) = 0;
// Get the receive parameters for the incoming stream identified by |ssrc|.
- // If |ssrc| is 0, retrieve the receive parameters for the default receive
- // stream, which is used when SSRCs are not signaled. Note that calling with
- // an |ssrc| of 0 will return encoding parameters with an unset |ssrc|
- // member.
virtual webrtc::RtpParameters GetRtpReceiveParameters(
uint32_t ssrc) const = 0;
+ // Retrieve the receive parameters for the default receive
+ // stream, which is used when SSRCs are not signaled.
+ virtual webrtc::RtpParameters GetDefaultRtpReceiveParameters() const = 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.
@@ -885,9 +887,11 @@
const VideoOptions* options,
rtc::VideoSourceInterface<webrtc::VideoFrame>* source) = 0;
// Sets the sink object to be used for the specified stream.
- // If SSRC is 0, the sink is used for the 'default' stream.
virtual bool SetSink(uint32_t ssrc,
rtc::VideoSinkInterface<webrtc::VideoFrame>* sink) = 0;
+ // The sink is used for the 'default' stream.
+ virtual void SetDefaultSink(
+ rtc::VideoSinkInterface<webrtc::VideoFrame>* sink) = 0;
// This fills the "bitrate parts" (rtx, video bitrate) of the
// BandwidthEstimationInfo, since that part that isn't possible to get
// through webrtc::Call::GetStats, as they are statistics of the send
diff --git a/media/engine/webrtc_video_engine.cc b/media/engine/webrtc_video_engine.cc
index 7f241c8..a5afcb3 100644
--- a/media/engine/webrtc_video_engine.cc
+++ b/media/engine/webrtc_video_engine.cc
@@ -923,26 +923,34 @@
uint32_t ssrc) const {
RTC_DCHECK_RUN_ON(&thread_checker_);
webrtc::RtpParameters rtp_params;
- // SSRC of 0 represents an unsignaled receive stream.
- if (ssrc == 0) {
- if (!default_unsignalled_ssrc_handler_.GetDefaultSink()) {
- RTC_LOG(LS_WARNING)
- << "Attempting to get RTP parameters for the default, "
- "unsignaled video receive stream, but not yet "
- "configured to receive such a stream.";
- return rtp_params;
- }
- rtp_params.encodings.emplace_back();
- } else {
- auto it = receive_streams_.find(ssrc);
- if (it == receive_streams_.end()) {
- RTC_LOG(LS_WARNING)
- << "Attempting to get RTP receive parameters for stream "
- << "with SSRC " << ssrc << " which doesn't exist.";
- return webrtc::RtpParameters();
- }
- rtp_params = it->second->GetRtpParameters();
+ auto it = receive_streams_.find(ssrc);
+ if (it == receive_streams_.end()) {
+ RTC_LOG(LS_WARNING)
+ << "Attempting to get RTP receive parameters for stream "
+ << "with SSRC " << ssrc << " which doesn't exist.";
+ return webrtc::RtpParameters();
}
+ rtp_params = it->second->GetRtpParameters();
+
+ // Add codecs, which any stream is prepared to receive.
+ for (const VideoCodec& codec : recv_params_.codecs) {
+ rtp_params.codecs.push_back(codec.ToCodecParameters());
+ }
+
+ return rtp_params;
+}
+
+webrtc::RtpParameters WebRtcVideoChannel::GetDefaultRtpReceiveParameters()
+ const {
+ RTC_DCHECK_RUN_ON(&thread_checker_);
+ webrtc::RtpParameters rtp_params;
+ if (!default_unsignalled_ssrc_handler_.GetDefaultSink()) {
+ RTC_LOG(LS_WARNING) << "Attempting to get RTP parameters for the default, "
+ "unsignaled video receive stream, but not yet "
+ "configured to receive such a stream.";
+ return rtp_params;
+ }
+ rtp_params.encodings.emplace_back();
// Add codecs, which any stream is prepared to receive.
for (const VideoCodec& codec : recv_params_.codecs) {
@@ -1360,10 +1368,6 @@
RTC_DCHECK_RUN_ON(&thread_checker_);
RTC_LOG(LS_INFO) << "SetSink: ssrc:" << ssrc << " "
<< (sink ? "(ptr)" : "nullptr");
- if (ssrc == 0) {
- default_unsignalled_ssrc_handler_.SetDefaultSink(this, sink);
- return true;
- }
std::map<uint32_t, WebRtcVideoReceiveStream*>::iterator it =
receive_streams_.find(ssrc);
@@ -1375,6 +1379,13 @@
return true;
}
+void WebRtcVideoChannel::SetDefaultSink(
+ rtc::VideoSinkInterface<webrtc::VideoFrame>* sink) {
+ RTC_DCHECK_RUN_ON(&thread_checker_);
+ RTC_LOG(LS_INFO) << "SetDefaultSink: " << (sink ? "(ptr)" : "nullptr");
+ default_unsignalled_ssrc_handler_.SetDefaultSink(this, sink);
+}
+
bool WebRtcVideoChannel::GetStats(VideoMediaInfo* info) {
RTC_DCHECK_RUN_ON(&thread_checker_);
TRACE_EVENT0("webrtc", "WebRtcVideoChannel::GetStats");
diff --git a/media/engine/webrtc_video_engine.h b/media/engine/webrtc_video_engine.h
index 4b42324..3a0a156 100644
--- a/media/engine/webrtc_video_engine.h
+++ b/media/engine/webrtc_video_engine.h
@@ -129,6 +129,7 @@
uint32_t ssrc,
const webrtc::RtpParameters& parameters) override;
webrtc::RtpParameters GetRtpReceiveParameters(uint32_t ssrc) const override;
+ webrtc::RtpParameters GetDefaultRtpReceiveParameters() const override;
bool GetSendCodec(VideoCodec* send_codec) override;
bool SetSend(bool send) override;
bool SetVideoSend(
@@ -143,6 +144,8 @@
void ResetUnsignaledRecvStream() override;
bool SetSink(uint32_t ssrc,
rtc::VideoSinkInterface<webrtc::VideoFrame>* sink) override;
+ void SetDefaultSink(
+ rtc::VideoSinkInterface<webrtc::VideoFrame>* sink) override;
void FillBitrateInfo(BandwidthEstimationInfo* bwe_info) override;
bool GetStats(VideoMediaInfo* info) override;
@@ -210,6 +213,7 @@
void RequestEncoderFallback() override;
void RequestEncoderSwitch(
const EncoderSwitchRequestCallback::Config& conf) override;
+
void SetRecordableEncodedFrameCallback(
uint32_t ssrc,
std::function<void(const webrtc::RecordableEncodedFrame&)> callback)
@@ -220,8 +224,8 @@
private:
class WebRtcVideoReceiveStream;
- // Finds VideoReceiveStream corresponding to ssrc. Aware of unsignalled
- // ssrc handling.
+ // Finds VideoReceiveStream corresponding to ssrc. Aware of unsignalled ssrc
+ // handling.
WebRtcVideoReceiveStream* FindReceiveStream(uint32_t ssrc)
RTC_EXCLUSIVE_LOCKS_REQUIRED(thread_checker_);
diff --git a/media/engine/webrtc_video_engine_unittest.cc b/media/engine/webrtc_video_engine_unittest.cc
index b7f9266..0270355 100644
--- a/media/engine/webrtc_video_engine_unittest.cc
+++ b/media/engine/webrtc_video_engine_unittest.cc
@@ -79,7 +79,6 @@
static const uint8_t kRedRtxPayloadType = 125;
static const uint32_t kTimeout = 5000U;
-static const uint32_t kDefaultReceiveSsrc = 0;
static const uint32_t kSsrc = 1234u;
static const uint32_t kSsrcs4[] = {1, 2, 3, 4};
static const int kVideoWidth = 640;
@@ -91,7 +90,6 @@
static const uint32_t kRtxSsrcs1[] = {4};
static const uint32_t kFlexfecSsrc = 5;
static const uint32_t kIncomingUnsignalledSsrc = 0xC0FFEE;
-static const uint32_t kDefaultRecvSsrc = 0;
constexpr uint32_t kRtpHeaderSize = 12;
@@ -1579,7 +1577,7 @@
void SendAndReceive(const cricket::VideoCodec& codec) {
EXPECT_TRUE(SetOneCodec(codec));
EXPECT_TRUE(SetSend(true));
- EXPECT_TRUE(channel_->SetSink(kDefaultReceiveSsrc, &renderer_));
+ channel_->SetDefaultSink(&renderer_);
EXPECT_EQ(0, renderer_.num_rendered_frames());
SendFrame();
EXPECT_FRAME_WAIT(1, kVideoWidth, kVideoHeight, kTimeout);
@@ -1592,7 +1590,7 @@
int fps) {
EXPECT_TRUE(SetOneCodec(codec));
EXPECT_TRUE(SetSend(true));
- EXPECT_TRUE(channel_->SetSink(kDefaultReceiveSsrc, &renderer_));
+ channel_->SetDefaultSink(&renderer_);
EXPECT_EQ(0, renderer_.num_rendered_frames());
for (int i = 0; i < duration_sec; ++i) {
for (int frame = 1; frame <= fps; ++frame) {
@@ -1984,12 +1982,12 @@
rtc::CopyOnWriteBuffer packet1(data1, sizeof(data1));
rtc::SetBE32(packet1.data() + 8, kSsrc);
- channel_->SetSink(kDefaultReceiveSsrc, NULL);
+ channel_->SetDefaultSink(NULL);
EXPECT_TRUE(SetDefaultCodec());
EXPECT_TRUE(SetSend(true));
EXPECT_EQ(0, renderer_.num_rendered_frames());
channel_->OnPacketReceived(packet1, /* packet_time_us */ -1);
- EXPECT_TRUE(channel_->SetSink(kDefaultReceiveSsrc, &renderer_));
+ channel_->SetDefaultSink(&renderer_);
SendFrame();
EXPECT_FRAME_WAIT(1, kVideoWidth, kVideoHeight, kTimeout);
}
@@ -1998,7 +1996,7 @@
TEST_F(WebRtcVideoChannelBaseTest, AddRemoveSendStreams) {
EXPECT_TRUE(SetOneCodec(DefaultCodec()));
EXPECT_TRUE(SetSend(true));
- EXPECT_TRUE(channel_->SetSink(kDefaultReceiveSsrc, &renderer_));
+ channel_->SetDefaultSink(&renderer_);
SendFrame();
EXPECT_FRAME_WAIT(1, kVideoWidth, kVideoHeight, kTimeout);
EXPECT_GT(NumRtpPackets(), 0);
@@ -2073,7 +2071,7 @@
const int time_between_send_ms = VideoFormat::FpsToInterval(kFramerate);
EXPECT_TRUE(SetOneCodec(codec));
EXPECT_TRUE(SetSend(true));
- EXPECT_TRUE(channel_->SetSink(kDefaultReceiveSsrc, &renderer_));
+ channel_->SetDefaultSink(&renderer_);
EXPECT_EQ(0, renderer_.num_rendered_frames());
SendFrame();
EXPECT_FRAME_WAIT(1, kVideoWidth, kVideoHeight, kTimeout);
@@ -2133,7 +2131,7 @@
TEST_F(WebRtcVideoChannelBaseTest, RemoveCapturerWithoutAdd) {
EXPECT_TRUE(SetOneCodec(DefaultCodec()));
EXPECT_TRUE(SetSend(true));
- EXPECT_TRUE(channel_->SetSink(kDefaultReceiveSsrc, &renderer_));
+ channel_->SetDefaultSink(&renderer_);
EXPECT_EQ(0, renderer_.num_rendered_frames());
SendFrame();
EXPECT_FRAME_WAIT(1, kVideoWidth, kVideoHeight, kTimeout);
@@ -5781,7 +5779,7 @@
// No receive streams yet.
ASSERT_EQ(0u, fake_call_->GetVideoReceiveStreams().size());
cricket::FakeVideoRenderer renderer;
- EXPECT_TRUE(channel_->SetSink(kDefaultRecvSsrc, &renderer));
+ channel_->SetDefaultSink(&renderer);
// Receive VP8 packet on first SSRC.
uint8_t data[kMinRtpPacketLen];
@@ -7478,10 +7476,12 @@
EXPECT_EQ(initial_params, channel_->GetRtpReceiveParameters(last_ssrc_));
}
-// Test that GetRtpReceiveParameters returns parameters correctly when SSRCs
-// aren't signaled. It should always return an empty "RtpEncodingParameters",
-// even after a packet is received and the unsignaled SSRC is known.
-TEST_F(WebRtcVideoChannelTest, GetRtpReceiveParametersWithUnsignaledSsrc) {
+// Test that GetDefaultRtpReceiveParameters returns parameters correctly when
+// SSRCs aren't signaled. It should always return an empty
+// "RtpEncodingParameters", even after a packet is received and the unsignaled
+// SSRC is known.
+TEST_F(WebRtcVideoChannelTest,
+ GetDefaultRtpReceiveParametersWithUnsignaledSsrc) {
// Call necessary methods to configure receiving a default stream as
// soon as it arrives.
cricket::VideoRecvParameters parameters;
@@ -7491,16 +7491,16 @@
// Call GetRtpReceiveParameters before configured to receive an unsignaled
// stream. Should return nothing.
- EXPECT_EQ(webrtc::RtpParameters(), channel_->GetRtpReceiveParameters(0));
+ EXPECT_EQ(webrtc::RtpParameters(),
+ channel_->GetDefaultRtpReceiveParameters());
// Set a sink for an unsignaled stream.
cricket::FakeVideoRenderer renderer;
- // Value of "0" means "unsignaled stream".
- EXPECT_TRUE(channel_->SetSink(0, &renderer));
+ channel_->SetDefaultSink(&renderer);
- // Call GetRtpReceiveParameters before the SSRC is known. Value of "0"
- // in this method means "unsignaled stream".
- webrtc::RtpParameters rtp_parameters = channel_->GetRtpReceiveParameters(0);
+ // Call GetDefaultRtpReceiveParameters before the SSRC is known.
+ webrtc::RtpParameters rtp_parameters =
+ channel_->GetDefaultRtpReceiveParameters();
ASSERT_EQ(1u, rtp_parameters.encodings.size());
EXPECT_FALSE(rtp_parameters.encodings[0].ssrc);
@@ -7515,7 +7515,7 @@
channel_->OnPacketReceived(packet, /* packet_time_us */ -1);
// The |ssrc| member should still be unset.
- rtp_parameters = channel_->GetRtpReceiveParameters(0);
+ rtp_parameters = channel_->GetDefaultRtpReceiveParameters();
ASSERT_EQ(1u, rtp_parameters.encodings.size());
EXPECT_FALSE(rtp_parameters.encodings[0].ssrc);
}
@@ -7780,7 +7780,7 @@
TEST_F(WebRtcVideoChannelBaseTest, GetSources) {
EXPECT_THAT(channel_->GetSources(kSsrc), IsEmpty());
- EXPECT_TRUE(channel_->SetSink(kDefaultReceiveSsrc, &renderer_));
+ channel_->SetDefaultSink(&renderer_);
EXPECT_TRUE(SetDefaultCodec());
EXPECT_TRUE(SetSend(true));
EXPECT_EQ(renderer_.num_rendered_frames(), 0);
diff --git a/media/engine/webrtc_voice_engine.cc b/media/engine/webrtc_voice_engine.cc
index cfbd774..97e12ba 100644
--- a/media/engine/webrtc_voice_engine.cc
+++ b/media/engine/webrtc_voice_engine.cc
@@ -1423,26 +1423,32 @@
uint32_t ssrc) const {
RTC_DCHECK(worker_thread_checker_.IsCurrent());
webrtc::RtpParameters rtp_params;
- // SSRC of 0 represents the default receive stream.
- if (ssrc == 0) {
- if (!default_sink_) {
- RTC_LOG(LS_WARNING)
- << "Attempting to get RTP parameters for the default, "
- "unsignaled audio receive stream, but not yet "
- "configured to receive such a stream.";
- return rtp_params;
- }
- rtp_params.encodings.emplace_back();
- } else {
- auto it = recv_streams_.find(ssrc);
- if (it == recv_streams_.end()) {
- RTC_LOG(LS_WARNING)
- << "Attempting to get RTP receive parameters for stream "
- << "with ssrc " << ssrc << " which doesn't exist.";
- return webrtc::RtpParameters();
- }
- rtp_params = it->second->GetRtpParameters();
+ auto it = recv_streams_.find(ssrc);
+ if (it == recv_streams_.end()) {
+ RTC_LOG(LS_WARNING)
+ << "Attempting to get RTP receive parameters for stream "
+ << "with ssrc " << ssrc << " which doesn't exist.";
+ return webrtc::RtpParameters();
}
+ rtp_params = it->second->GetRtpParameters();
+
+ for (const AudioCodec& codec : recv_codecs_) {
+ rtp_params.codecs.push_back(codec.ToCodecParameters());
+ }
+ return rtp_params;
+}
+
+webrtc::RtpParameters WebRtcVoiceMediaChannel::GetDefaultRtpReceiveParameters()
+ const {
+ RTC_DCHECK(worker_thread_checker_.IsCurrent());
+ webrtc::RtpParameters rtp_params;
+ if (!default_sink_) {
+ RTC_LOG(LS_WARNING) << "Attempting to get RTP parameters for the default, "
+ "unsignaled audio receive stream, but not yet "
+ "configured to receive such a stream.";
+ return rtp_params;
+ }
+ rtp_params.encodings.emplace_back();
for (const AudioCodec& codec : recv_codecs_) {
rtp_params.codecs.push_back(codec.ToCodecParameters());
@@ -1910,20 +1916,28 @@
bool WebRtcVoiceMediaChannel::SetOutputVolume(uint32_t ssrc, double volume) {
RTC_DCHECK(worker_thread_checker_.IsCurrent());
- std::vector<uint32_t> ssrcs(1, ssrc);
- // SSRC of 0 represents the default receive stream.
- if (ssrc == 0) {
- default_recv_volume_ = volume;
- ssrcs = unsignaled_recv_ssrcs_;
+ const auto it = recv_streams_.find(ssrc);
+ if (it == recv_streams_.end()) {
+ RTC_LOG(LS_WARNING) << "SetOutputVolume: no recv stream " << ssrc;
+ return false;
}
- for (uint32_t ssrc : ssrcs) {
+ it->second->SetOutputVolume(volume);
+ RTC_LOG(LS_INFO) << "SetOutputVolume() to " << volume
+ << " for recv stream with ssrc " << ssrc;
+ return true;
+}
+
+bool WebRtcVoiceMediaChannel::SetDefaultOutputVolume(double volume) {
+ RTC_DCHECK(worker_thread_checker_.IsCurrent());
+ default_recv_volume_ = volume;
+ for (uint32_t ssrc : unsignaled_recv_ssrcs_) {
const auto it = recv_streams_.find(ssrc);
if (it == recv_streams_.end()) {
- RTC_LOG(LS_WARNING) << "SetOutputVolume: no recv stream " << ssrc;
+ RTC_LOG(LS_WARNING) << "SetDefaultOutputVolume: no recv stream " << ssrc;
return false;
}
it->second->SetOutputVolume(volume);
- RTC_LOG(LS_INFO) << "SetOutputVolume() to " << volume
+ RTC_LOG(LS_INFO) << "SetDefaultOutputVolume() to " << volume
<< " for recv stream with ssrc " << ssrc;
}
return true;
@@ -2267,15 +2281,6 @@
RTC_DCHECK(worker_thread_checker_.IsCurrent());
RTC_LOG(LS_VERBOSE) << "WebRtcVoiceMediaChannel::SetRawAudioSink: ssrc:"
<< ssrc << " " << (sink ? "(ptr)" : "NULL");
- if (ssrc == 0) {
- if (!unsignaled_recv_ssrcs_.empty()) {
- std::unique_ptr<webrtc::AudioSinkInterface> proxy_sink(
- sink ? new ProxySink(sink.get()) : nullptr);
- SetRawAudioSink(unsignaled_recv_ssrcs_.back(), std::move(proxy_sink));
- }
- default_sink_ = std::move(sink);
- return;
- }
const auto it = recv_streams_.find(ssrc);
if (it == recv_streams_.end()) {
RTC_LOG(LS_WARNING) << "SetRawAudioSink: no recv stream " << ssrc;
@@ -2284,6 +2289,18 @@
it->second->SetRawAudioSink(std::move(sink));
}
+void WebRtcVoiceMediaChannel::SetDefaultRawAudioSink(
+ std::unique_ptr<webrtc::AudioSinkInterface> sink) {
+ RTC_DCHECK(worker_thread_checker_.IsCurrent());
+ RTC_LOG(LS_VERBOSE) << "WebRtcVoiceMediaChannel::SetDefaultRawAudioSink:";
+ if (!unsignaled_recv_ssrcs_.empty()) {
+ std::unique_ptr<webrtc::AudioSinkInterface> proxy_sink(
+ sink ? new ProxySink(sink.get()) : nullptr);
+ SetRawAudioSink(unsignaled_recv_ssrcs_.back(), std::move(proxy_sink));
+ }
+ default_sink_ = std::move(sink);
+}
+
std::vector<webrtc::RtpSource> WebRtcVoiceMediaChannel::GetSources(
uint32_t ssrc) const {
auto it = recv_streams_.find(ssrc);
diff --git a/media/engine/webrtc_voice_engine.h b/media/engine/webrtc_voice_engine.h
index 3c46980..7ff8690 100644
--- a/media/engine/webrtc_voice_engine.h
+++ b/media/engine/webrtc_voice_engine.h
@@ -151,6 +151,7 @@
uint32_t ssrc,
const webrtc::RtpParameters& parameters) override;
webrtc::RtpParameters GetRtpReceiveParameters(uint32_t ssrc) const override;
+ webrtc::RtpParameters GetDefaultRtpReceiveParameters() const override;
void SetPlayout(bool playout) override;
void SetSend(bool send) override;
@@ -178,8 +179,9 @@
rtc::scoped_refptr<webrtc::FrameEncryptorInterface>
frame_encryptor) override;
- // SSRC=0 will apply the new volume to current and future unsignaled streams.
bool SetOutputVolume(uint32_t ssrc, double volume) override;
+ // Applies the new volume to current and future unsignaled streams.
+ bool SetDefaultOutputVolume(double volume) override;
bool SetBaseMinimumPlayoutDelayMs(uint32_t ssrc, int delay_ms) override;
absl::optional<int> GetBaseMinimumPlayoutDelayMs(
@@ -195,11 +197,14 @@
void OnReadyToSend(bool ready) override;
bool GetStats(VoiceMediaInfo* info) override;
- // SSRC=0 will set the audio sink on the latest unsignaled stream, future or
- // current. Only one stream at a time will use the sink.
+ // Set the audio sink for an existing stream.
void SetRawAudioSink(
uint32_t ssrc,
std::unique_ptr<webrtc::AudioSinkInterface> sink) override;
+ // Will set the audio sink on the latest unsignaled stream, future or
+ // current. Only one stream at a time will use the sink.
+ void SetDefaultRawAudioSink(
+ std::unique_ptr<webrtc::AudioSinkInterface> sink) override;
std::vector<webrtc::RtpSource> GetSources(uint32_t ssrc) const override;
diff --git a/media/engine/webrtc_voice_engine_unittest.cc b/media/engine/webrtc_voice_engine_unittest.cc
index d509831..d34383c 100644
--- a/media/engine/webrtc_voice_engine_unittest.cc
+++ b/media/engine/webrtc_voice_engine_unittest.cc
@@ -1392,18 +1392,18 @@
parameters.codecs.push_back(kPcmuCodec);
EXPECT_TRUE(channel_->SetRecvParameters(parameters));
- // Call GetRtpReceiveParameters before configured to receive an unsignaled
- // stream. Should return nothing.
- EXPECT_EQ(webrtc::RtpParameters(), channel_->GetRtpReceiveParameters(0));
+ // Call GetDefaultRtpReceiveParameters before configured to receive an
+ // unsignaled stream. Should return nothing.
+ EXPECT_EQ(webrtc::RtpParameters(),
+ channel_->GetDefaultRtpReceiveParameters());
// Set a sink for an unsignaled stream.
std::unique_ptr<FakeAudioSink> fake_sink(new FakeAudioSink());
- // Value of "0" means "unsignaled stream".
- channel_->SetRawAudioSink(0, std::move(fake_sink));
+ channel_->SetDefaultRawAudioSink(std::move(fake_sink));
- // Call GetRtpReceiveParameters before the SSRC is known. Value of "0"
- // in this method means "unsignaled stream".
- webrtc::RtpParameters rtp_parameters = channel_->GetRtpReceiveParameters(0);
+ // Call GetDefaultRtpReceiveParameters before the SSRC is known.
+ webrtc::RtpParameters rtp_parameters =
+ channel_->GetDefaultRtpReceiveParameters();
ASSERT_EQ(1u, rtp_parameters.encodings.size());
EXPECT_FALSE(rtp_parameters.encodings[0].ssrc);
@@ -1411,7 +1411,7 @@
DeliverPacket(kPcmuFrame, sizeof(kPcmuFrame));
// The |ssrc| member should still be unset.
- rtp_parameters = channel_->GetRtpReceiveParameters(0);
+ rtp_parameters = channel_->GetDefaultRtpReceiveParameters();
ASSERT_EQ(1u, rtp_parameters.encodings.size());
EXPECT_FALSE(rtp_parameters.encodings[0].ssrc);
}
@@ -3104,7 +3104,7 @@
// Should remember the volume "2" which will be set on new unsignaled streams,
// and also set the gain to 2 on existing unsignaled streams.
- EXPECT_TRUE(channel_->SetOutputVolume(kSsrc0, 2));
+ EXPECT_TRUE(channel_->SetDefaultOutputVolume(2));
EXPECT_DOUBLE_EQ(2, GetRecvStream(kSsrc1).gain());
// Spawn an unsignaled stream by sending a packet - gain should be 2.
@@ -3114,8 +3114,8 @@
DeliverPacket(pcmuFrame2, sizeof(pcmuFrame2));
EXPECT_DOUBLE_EQ(2, GetRecvStream(kSsrcX).gain());
- // Setting gain with SSRC=0 should affect all unsignaled streams.
- EXPECT_TRUE(channel_->SetOutputVolume(kSsrc0, 3));
+ // Setting gain for all unsignaled streams.
+ EXPECT_TRUE(channel_->SetDefaultOutputVolume(3));
if (kMaxUnsignaledRecvStreams > 1) {
EXPECT_DOUBLE_EQ(3, GetRecvStream(kSsrc1).gain());
}
@@ -3341,7 +3341,7 @@
std::unique_ptr<FakeAudioSink> fake_sink_4(new FakeAudioSink());
// Should be able to set a default sink even when no stream exists.
- channel_->SetRawAudioSink(0, std::move(fake_sink_1));
+ channel_->SetDefaultRawAudioSink(std::move(fake_sink_1));
// Spawn an unsignaled stream by sending a packet - it should be assigned the
// default sink.
@@ -3349,11 +3349,11 @@
EXPECT_NE(nullptr, GetRecvStream(kSsrc1).sink());
// Try resetting the default sink.
- channel_->SetRawAudioSink(kSsrc0, nullptr);
+ channel_->SetDefaultRawAudioSink(nullptr);
EXPECT_EQ(nullptr, GetRecvStream(kSsrc1).sink());
// Try setting the default sink while the default stream exists.
- channel_->SetRawAudioSink(kSsrc0, std::move(fake_sink_2));
+ channel_->SetDefaultRawAudioSink(std::move(fake_sink_2));
EXPECT_NE(nullptr, GetRecvStream(kSsrc1).sink());
// If we remove and add a default stream, it should get the same sink.
@@ -3373,14 +3373,14 @@
EXPECT_NE(nullptr, GetRecvStream(kSsrcX).sink());
// Reset the default sink - the second unsignaled stream should lose it.
- channel_->SetRawAudioSink(kSsrc0, nullptr);
+ channel_->SetDefaultRawAudioSink(nullptr);
if (kMaxUnsignaledRecvStreams > 1) {
EXPECT_EQ(nullptr, GetRecvStream(kSsrc1).sink());
}
EXPECT_EQ(nullptr, GetRecvStream(kSsrcX).sink());
// Try setting the default sink while two streams exists.
- channel_->SetRawAudioSink(kSsrc0, std::move(fake_sink_3));
+ channel_->SetDefaultRawAudioSink(std::move(fake_sink_3));
if (kMaxUnsignaledRecvStreams > 1) {
EXPECT_EQ(nullptr, GetRecvStream(kSsrc1).sink());
}
diff --git a/pc/audio_rtp_receiver.cc b/pc/audio_rtp_receiver.cc
index 6c0445a..e83e558 100644
--- a/pc/audio_rtp_receiver.cc
+++ b/pc/audio_rtp_receiver.cc
@@ -76,8 +76,8 @@
RTC_DCHECK(media_channel_);
RTC_DCHECK(!stopped_);
return worker_thread_->Invoke<bool>(RTC_FROM_HERE, [&] {
- // TODO(bugs.webrtc.org/8694): Stop using 0 to mean unsignalled SSRC value.
- return media_channel_->SetOutputVolume(ssrc_.value_or(0), volume);
+ return ssrc_ ? media_channel_->SetOutputVolume(*ssrc_, volume)
+ : media_channel_->SetDefaultOutputVolume(volume);
});
}
@@ -112,8 +112,8 @@
return RtpParameters();
}
return worker_thread_->Invoke<RtpParameters>(RTC_FROM_HERE, [&] {
- // TODO(bugs.webrtc.org/8694): Stop using 0 to mean unsignalled SSRC value.
- return media_channel_->GetRtpReceiveParameters(ssrc_.value_or(0));
+ return ssrc_ ? media_channel_->GetRtpReceiveParameters(*ssrc_)
+ : media_channel_->GetDefaultRtpReceiveParameters();
});
}
@@ -153,12 +153,12 @@
}
if (!stopped_) {
- source_->Stop(media_channel_, ssrc_.value_or(0));
+ source_->Stop(media_channel_, ssrc_);
delay_->OnStop();
}
ssrc_ = ssrc;
stopped_ = false;
- source_->Start(media_channel_, ssrc.value_or(0));
+ source_->Start(media_channel_, ssrc);
delay_->OnStart(media_channel_, ssrc.value_or(0));
Reconfigure();
}
diff --git a/pc/remote_audio_source.cc b/pc/remote_audio_source.cc
index d9752f3..05d0899 100644
--- a/pc/remote_audio_source.cc
+++ b/pc/remote_audio_source.cc
@@ -63,7 +63,7 @@
}
void RemoteAudioSource::Start(cricket::VoiceMediaChannel* media_channel,
- uint32_t ssrc) {
+ absl::optional<uint32_t> ssrc) {
RTC_DCHECK_RUN_ON(main_thread_);
RTC_DCHECK(media_channel);
@@ -71,18 +71,22 @@
// notified when a channel goes out of scope (signaled when "AudioDataProxy"
// is destroyed).
worker_thread_->Invoke<void>(RTC_FROM_HERE, [&] {
- media_channel->SetRawAudioSink(ssrc,
- std::make_unique<AudioDataProxy>(this));
+ ssrc ? media_channel->SetRawAudioSink(
+ *ssrc, std::make_unique<AudioDataProxy>(this))
+ : media_channel->SetDefaultRawAudioSink(
+ std::make_unique<AudioDataProxy>(this));
});
}
void RemoteAudioSource::Stop(cricket::VoiceMediaChannel* media_channel,
- uint32_t ssrc) {
+ absl::optional<uint32_t> ssrc) {
RTC_DCHECK_RUN_ON(main_thread_);
RTC_DCHECK(media_channel);
- worker_thread_->Invoke<void>(
- RTC_FROM_HERE, [&] { media_channel->SetRawAudioSink(ssrc, nullptr); });
+ worker_thread_->Invoke<void>(RTC_FROM_HERE, [&] {
+ ssrc ? media_channel->SetRawAudioSink(*ssrc, nullptr)
+ : media_channel->SetDefaultRawAudioSink(nullptr);
+ });
}
MediaSourceInterface::SourceState RemoteAudioSource::state() const {
diff --git a/pc/remote_audio_source.h b/pc/remote_audio_source.h
index 399e7e3..15dc75b 100644
--- a/pc/remote_audio_source.h
+++ b/pc/remote_audio_source.h
@@ -14,6 +14,7 @@
#include <list>
#include <string>
+#include "absl/types/optional.h"
#include "api/call/audio_sink.h"
#include "api/notifier.h"
#include "pc/channel.h"
@@ -37,8 +38,10 @@
// Register and unregister remote audio source with the underlying media
// engine.
- void Start(cricket::VoiceMediaChannel* media_channel, uint32_t ssrc);
- void Stop(cricket::VoiceMediaChannel* media_channel, uint32_t ssrc);
+ void Start(cricket::VoiceMediaChannel* media_channel,
+ absl::optional<uint32_t> ssrc);
+ void Stop(cricket::VoiceMediaChannel* media_channel,
+ absl::optional<uint32_t> ssrc);
// MediaSourceInterface implementation.
MediaSourceInterface::SourceState state() const override;
diff --git a/pc/video_rtp_receiver.cc b/pc/video_rtp_receiver.cc
index d50407d..24cc4ea 100644
--- a/pc/video_rtp_receiver.cc
+++ b/pc/video_rtp_receiver.cc
@@ -83,8 +83,8 @@
return RtpParameters();
}
return worker_thread_->Invoke<RtpParameters>(RTC_FROM_HERE, [&] {
- // TODO(bugs.webrtc.org/8694): Stop using 0 to mean unsignalled SSRC
- return media_channel_->GetRtpReceiveParameters(ssrc_.value_or(0));
+ return ssrc_ ? media_channel_->GetRtpReceiveParameters(*ssrc_)
+ : media_channel_->GetDefaultRtpReceiveParameters();
});
}
@@ -155,8 +155,11 @@
}
void VideoRtpReceiver::SetSink(rtc::VideoSinkInterface<VideoFrame>* sink) {
- // TODO(bugs.webrtc.org/8694): Stop using 0 to mean unsignalled SSRC
- media_channel_->SetSink(ssrc_.value_or(0), sink);
+ if (ssrc_) {
+ media_channel_->SetSink(*ssrc_, sink);
+ return;
+ }
+ media_channel_->SetDefaultSink(sink);
}
void VideoRtpReceiver::SetupMediaChannel(uint32_t ssrc) {
