- Rename VoiceChannel::MuteStream() -> SetAudioSend() (incl. media channel)
- Rename VideoChannel::MuteStream() -> SetVideoSend() (incl. media channel)
- Collapse NnChannel::SetChannelOptions() into the above.
- Collapse VoiceChannel::SetLocalRenderer into SetAudioSend().
BUG=webrtc:4690
Review URL: https://codereview.webrtc.org/1311533009
Cr-Commit-Position: refs/heads/master@{#9915}
diff --git a/talk/app/webrtc/webrtcsession.cc b/talk/app/webrtc/webrtcsession.cc
index 082e214..179d793 100644
--- a/talk/app/webrtc/webrtcsession.cc
+++ b/talk/app/webrtc/webrtcsession.cc
@@ -1182,20 +1182,9 @@
LOG(LS_ERROR) << "SetAudioSend: No audio channel exists.";
return;
}
- if (!voice_channel_->SetLocalRenderer(ssrc, renderer)) {
- // SetRenderer() can fail if the ssrc does not match any send channel.
+ if (!voice_channel_->SetAudioSend(ssrc, !enable, &options, renderer)) {
LOG(LS_ERROR) << "SetAudioSend: ssrc is incorrect: " << ssrc;
- return;
}
- if (!voice_channel_->MuteStream(ssrc, !enable)) {
- // Allow that MuteStream fail if |enable| is false but assert otherwise.
- // This in the normal case when the underlying media channel has already
- // been deleted.
- ASSERT(enable == false);
- return;
- }
- if (enable)
- voice_channel_->SetChannelOptions(options);
}
void WebRtcSession::SetAudioPlayoutVolume(uint32 ssrc, double volume) {
@@ -1254,15 +1243,12 @@
LOG(LS_WARNING) << "SetVideoSend: No video channel exists.";
return;
}
- if (!video_channel_->MuteStream(ssrc, !enable)) {
+ if (!video_channel_->SetVideoSend(ssrc, !enable, options)) {
// Allow that MuteStream fail if |enable| is false but assert otherwise.
// This in the normal case when the underlying media channel has already
// been deleted.
ASSERT(enable == false);
- return;
}
- if (enable && options)
- video_channel_->SetChannelOptions(*options);
}
bool WebRtcSession::CanInsertDtmf(const std::string& track_id) {
diff --git a/talk/media/base/fakemediaengine.h b/talk/media/base/fakemediaengine.h
index 47ec402..56986c5 100644
--- a/talk/media/base/fakemediaengine.h
+++ b/talk/media/base/fakemediaengine.h
@@ -137,15 +137,6 @@
virtual bool RemoveRecvStream(uint32 ssrc) {
return RemoveStreamBySsrc(&receive_streams_, ssrc);
}
- virtual bool MuteStream(uint32 ssrc, bool on) {
- if (!HasSendStream(ssrc) && ssrc != 0)
- return false;
- if (on)
- muted_streams_.insert(ssrc);
- else
- muted_streams_.erase(ssrc);
- return true;
- }
bool IsStreamMuted(uint32 ssrc) const {
bool ret = muted_streams_.find(ssrc) != muted_streams_.end();
// If |ssrc = 0| check if the first send stream is muted.
@@ -188,6 +179,15 @@
}
protected:
+ bool MuteStream(uint32 ssrc, bool mute) {
+ if (!HasSendStream(ssrc) && ssrc != 0)
+ return false;
+ if (mute)
+ muted_streams_.insert(ssrc);
+ else
+ muted_streams_.erase(ssrc);
+ return true;
+ }
bool set_sending(bool send) {
sending_ = send;
return true;
@@ -283,6 +283,20 @@
}
return set_sending(flag != SEND_NOTHING);
}
+ virtual bool SetAudioSend(uint32 ssrc, bool mute,
+ const AudioOptions* options,
+ AudioRenderer* renderer) {
+ if (!SetLocalRenderer(ssrc, renderer)) {
+ return false;
+ }
+ if (!RtpHelper<VoiceMediaChannel>::MuteStream(ssrc, mute)) {
+ return false;
+ }
+ if (!mute && options) {
+ return SetOptions(*options);
+ }
+ return true;
+ }
virtual bool SetMaxSendBandwidth(int bps) { return true; }
virtual bool AddRecvStream(const StreamParams& sp) {
if (!RtpHelper<VoiceMediaChannel>::AddRecvStream(sp))
@@ -316,26 +330,6 @@
}
return true;
}
- virtual bool SetLocalRenderer(uint32 ssrc, AudioRenderer* renderer) {
- std::map<uint32, VoiceChannelAudioSink*>::iterator it =
- local_renderers_.find(ssrc);
- if (renderer) {
- if (it != local_renderers_.end()) {
- ASSERT(it->second->renderer() == renderer);
- } else {
- local_renderers_.insert(std::make_pair(
- ssrc, new VoiceChannelAudioSink(renderer)));
- }
- } else {
- if (it != local_renderers_.end()) {
- delete it->second;
- local_renderers_.erase(it);
- } else {
- return false;
- }
- }
- return true;
- }
virtual bool GetActiveStreams(AudioInfo::StreamList* streams) { return true; }
virtual int GetOutputLevel() { return 0; }
@@ -442,6 +436,23 @@
AudioRenderer* renderer_;
};
+ bool SetLocalRenderer(uint32 ssrc, AudioRenderer* renderer) {
+ auto it = local_renderers_.find(ssrc);
+ if (renderer) {
+ if (it != local_renderers_.end()) {
+ ASSERT(it->second->renderer() == renderer);
+ } else {
+ local_renderers_.insert(std::make_pair(
+ ssrc, new VoiceChannelAudioSink(renderer)));
+ }
+ } else {
+ if (it != local_renderers_.end()) {
+ delete it->second;
+ local_renderers_.erase(it);
+ }
+ }
+ return true;
+ }
FakeVoiceEngine* engine_;
std::vector<AudioCodec> recv_codecs_;
@@ -556,6 +567,17 @@
}
virtual bool SetSend(bool send) { return set_sending(send); }
+ virtual bool SetVideoSend(uint32 ssrc, bool mute,
+ const VideoOptions* options) {
+ if (!RtpHelper<VideoMediaChannel>::MuteStream(ssrc, mute)) {
+ return false;
+ }
+ if (!mute && options) {
+ return SetOptions(*options);
+ } else {
+ return true;
+ }
+ }
virtual bool SetCapturer(uint32 ssrc, VideoCapturer* capturer) {
capturers_[ssrc] = capturer;
return true;
diff --git a/talk/media/base/mediachannel.h b/talk/media/base/mediachannel.h
index 7c76d2b..cfeb2d8 100644
--- a/talk/media/base/mediachannel.h
+++ b/talk/media/base/mediachannel.h
@@ -566,9 +566,6 @@
// multiple SSRCs.
virtual bool RemoveRecvStream(uint32 ssrc) = 0;
- // Mutes the channel.
- virtual bool MuteStream(uint32 ssrc, bool on) = 0;
-
// Sets the RTP extension headers and IDs to use when sending RTP.
virtual bool SetRecvRtpHeaderExtensions(
const std::vector<RtpHeaderExtension>& extensions) = 0;
@@ -1097,10 +1094,11 @@
virtual bool SetPlayout(bool playout) = 0;
// Starts or stops sending (and potentially capture) of local audio.
virtual bool SetSend(SendFlags flag) = 0;
+ // Configure stream for sending.
+ virtual bool SetAudioSend(uint32 ssrc, bool mute, const AudioOptions* options,
+ AudioRenderer* renderer) = 0;
// Sets the renderer object to be used for the specified remote audio stream.
virtual bool SetRemoteRenderer(uint32 ssrc, AudioRenderer* renderer) = 0;
- // Sets the renderer object to be used for the specified local audio stream.
- virtual bool SetLocalRenderer(uint32 ssrc, AudioRenderer* renderer) = 0;
// Gets current energy levels for all incoming streams.
virtual bool GetActiveStreams(AudioInfo::StreamList* actives) = 0;
// Get the current energy level of the stream sent to the speaker.
@@ -1196,6 +1194,9 @@
virtual bool SetRender(bool render) = 0;
// Starts or stops transmission (and potentially capture) of local video.
virtual bool SetSend(bool send) = 0;
+ // Configure stream for sending.
+ virtual bool SetVideoSend(uint32 ssrc, bool mute,
+ const VideoOptions* options) = 0;
// Sets the renderer object to be used for the specified stream.
// If SSRC is 0, the renderer is used for the 'default' stream.
virtual bool SetRenderer(uint32 ssrc, VideoRenderer* renderer) = 0;
@@ -1337,7 +1338,6 @@
virtual bool SetSendCodecs(const std::vector<DataCodec>& codecs) = 0;
virtual bool SetRecvCodecs(const std::vector<DataCodec>& codecs) = 0;
- virtual bool MuteStream(uint32 ssrc, bool on) { return false; }
// TODO(pthatcher): Implement this.
virtual bool GetStats(DataMediaInfo* info) { return true; }
diff --git a/talk/media/base/videoengine_unittest.h b/talk/media/base/videoengine_unittest.h
index 5f00a0e..c74479c 100644
--- a/talk/media/base/videoengine_unittest.h
+++ b/talk/media/base/videoengine_unittest.h
@@ -1647,48 +1647,41 @@
EXPECT_TRUE(channel_->SetRender(true));
EXPECT_TRUE(channel_->SetRenderer(kDefaultReceiveSsrc, &renderer_));
EXPECT_EQ(0, renderer_.num_rendered_frames());
-
// Mute the channel and expect black output frame.
int frame_count = 0;
- EXPECT_TRUE(channel_->MuteStream(kSsrc, true));
+ EXPECT_TRUE(channel_->SetVideoSend(kSsrc, true, nullptr));
EXPECT_TRUE(video_capturer.CaptureFrame());
++frame_count;
EXPECT_EQ_WAIT(frame_count, renderer_.num_rendered_frames(), kTimeout);
EXPECT_TRUE(renderer_.black_frame());
-
// Unmute the channel and expect non-black output frame.
- EXPECT_TRUE(channel_->MuteStream(kSsrc, false));
+ EXPECT_TRUE(channel_->SetVideoSend(kSsrc, false, nullptr));
EXPECT_TRUE(rtc::Thread::Current()->ProcessMessages(30));
EXPECT_TRUE(video_capturer.CaptureFrame());
++frame_count;
EXPECT_EQ_WAIT(frame_count, renderer_.num_rendered_frames(), kTimeout);
EXPECT_FALSE(renderer_.black_frame());
-
// Test that we can also Mute using the correct send stream SSRC.
- EXPECT_TRUE(channel_->MuteStream(kSsrc, true));
+ EXPECT_TRUE(channel_->SetVideoSend(kSsrc, true, nullptr));
EXPECT_TRUE(rtc::Thread::Current()->ProcessMessages(30));
EXPECT_TRUE(video_capturer.CaptureFrame());
++frame_count;
EXPECT_EQ_WAIT(frame_count, renderer_.num_rendered_frames(), kTimeout);
EXPECT_TRUE(renderer_.black_frame());
-
- EXPECT_TRUE(channel_->MuteStream(kSsrc, false));
+ EXPECT_TRUE(channel_->SetVideoSend(kSsrc, false, nullptr));
EXPECT_TRUE(rtc::Thread::Current()->ProcessMessages(30));
EXPECT_TRUE(video_capturer.CaptureFrame());
++frame_count;
EXPECT_EQ_WAIT(frame_count, renderer_.num_rendered_frames(), kTimeout);
EXPECT_FALSE(renderer_.black_frame());
-
// Test that muting an existing stream succeeds even if it's muted.
- EXPECT_TRUE(channel_->MuteStream(kSsrc, true));
- EXPECT_TRUE(channel_->MuteStream(kSsrc, true));
-
+ EXPECT_TRUE(channel_->SetVideoSend(kSsrc, true, nullptr));
+ EXPECT_TRUE(channel_->SetVideoSend(kSsrc, true, nullptr));
// Test that unmuting an existing stream succeeds even if it's not muted.
- EXPECT_TRUE(channel_->MuteStream(kSsrc, false));
- EXPECT_TRUE(channel_->MuteStream(kSsrc, false));
-
+ EXPECT_TRUE(channel_->SetVideoSend(kSsrc, false, nullptr));
+ EXPECT_TRUE(channel_->SetVideoSend(kSsrc, false, nullptr));
// Test that muting an invalid stream fails.
- EXPECT_FALSE(channel_->MuteStream(kSsrc+1, true));
+ EXPECT_FALSE(channel_->SetVideoSend(kSsrc+1, true, nullptr));
EXPECT_TRUE(channel_->SetCapturer(kSsrc, NULL));
}
diff --git a/talk/media/webrtc/webrtcvideoengine2.cc b/talk/media/webrtc/webrtcvideoengine2.cc
index 60a7022..e94c0b1 100644
--- a/talk/media/webrtc/webrtcvideoengine2.cc
+++ b/talk/media/webrtc/webrtcvideoengine2.cc
@@ -1095,6 +1095,20 @@
return true;
}
+bool WebRtcVideoChannel2::SetVideoSend(uint32 ssrc, bool mute,
+ const VideoOptions* options) {
+ // TODO(solenberg): The state change should be fully rolled back if any one of
+ // these calls fail.
+ if (!MuteStream(ssrc, mute)) {
+ return false;
+ }
+ if (!mute && options) {
+ return SetOptions(*options);
+ } else {
+ return true;
+ }
+}
+
bool WebRtcVideoChannel2::ValidateSendSsrcAvailability(
const StreamParams& sp) const {
for (uint32_t ssrc: sp.ssrcs) {
diff --git a/talk/media/webrtc/webrtcvideoengine2.h b/talk/media/webrtc/webrtcvideoengine2.h
index eabea76..f27099b 100644
--- a/talk/media/webrtc/webrtcvideoengine2.h
+++ b/talk/media/webrtc/webrtcvideoengine2.h
@@ -193,7 +193,8 @@
bool SetSendStreamFormat(uint32 ssrc, const VideoFormat& format) override;
bool SetRender(bool render) override;
bool SetSend(bool send) override;
-
+ bool SetVideoSend(uint32 ssrc, bool mute,
+ const VideoOptions* options) override;
bool AddSendStream(const StreamParams& sp) override;
bool RemoveSendStream(uint32 ssrc) override;
bool AddRecvStream(const StreamParams& sp) override;
@@ -210,7 +211,6 @@
void OnRtcpReceived(rtc::Buffer* packet,
const rtc::PacketTime& packet_time) override;
void OnReadyToSend(bool ready) override;
- bool MuteStream(uint32 ssrc, bool mute) override;
// Set send/receive RTP header extensions. This must be done before creating
// streams as it only has effect on future streams.
@@ -233,6 +233,7 @@
bool GetRenderer(uint32 ssrc, VideoRenderer** renderer);
private:
+ bool MuteStream(uint32 ssrc, bool mute);
class WebRtcVideoReceiveStream;
void ConfigureReceiverRtp(webrtc::VideoReceiveStream::Config* config,
const StreamParams& sp) const;
diff --git a/talk/media/webrtc/webrtcvoiceengine.cc b/talk/media/webrtc/webrtcvoiceengine.cc
index bcf1738..43cbc2e 100644
--- a/talk/media/webrtc/webrtcvoiceengine.cc
+++ b/talk/media/webrtc/webrtcvoiceengine.cc
@@ -2374,6 +2374,23 @@
return true;
}
+bool WebRtcVoiceMediaChannel::SetAudioSend(uint32 ssrc, bool mute,
+ const AudioOptions* options,
+ AudioRenderer* renderer) {
+ // TODO(solenberg): The state change should be fully rolled back if any one of
+ // these calls fail.
+ if (!SetLocalRenderer(ssrc, renderer)) {
+ return false;
+ }
+ if (!MuteStream(ssrc, mute)) {
+ return false;
+ }
+ if (!mute && options) {
+ return SetOptions(*options);
+ }
+ return true;
+}
+
// TODO(ronghuawu): Change this method to return bool.
void WebRtcVoiceMediaChannel::ConfigureSendChannel(int channel) {
if (engine()->voe()->network()->RegisterExternalTransport(
diff --git a/talk/media/webrtc/webrtcvoiceengine.h b/talk/media/webrtc/webrtcvoiceengine.h
index 4b35ba7..951c878 100644
--- a/talk/media/webrtc/webrtcvoiceengine.h
+++ b/talk/media/webrtc/webrtcvoiceengine.h
@@ -302,12 +302,13 @@
bool SetSend(SendFlags send) override;
bool PauseSend();
bool ResumeSend();
+ bool SetAudioSend(uint32 ssrc, bool mute, const AudioOptions* options,
+ AudioRenderer* renderer) override;
bool AddSendStream(const StreamParams& sp) override;
bool RemoveSendStream(uint32 ssrc) override;
bool AddRecvStream(const StreamParams& sp) override;
bool RemoveRecvStream(uint32 ssrc) override;
bool SetRemoteRenderer(uint32 ssrc, AudioRenderer* renderer) override;
- bool SetLocalRenderer(uint32 ssrc, AudioRenderer* renderer) override;
bool GetActiveStreams(AudioInfo::StreamList* actives) override;
int GetOutputLevel() override;
int GetTimeSinceLastTyping() override;
@@ -328,7 +329,6 @@
void OnRtcpReceived(rtc::Buffer* packet,
const rtc::PacketTime& packet_time) override;
void OnReadyToSend(bool ready) override {}
- bool MuteStream(uint32 ssrc, bool on) override;
bool SetMaxSendBandwidth(int bps) override;
bool GetStats(VoiceMediaInfo* info) override;
// Gets last reported error from WebRtc voice engine. This should be only
@@ -359,6 +359,8 @@
void SetCall(webrtc::Call* call);
private:
+ bool SetLocalRenderer(uint32 ssrc, AudioRenderer* renderer);
+ bool MuteStream(uint32 ssrc, bool mute);
WebRtcVoiceEngine* engine() { return engine_; }
int GetLastEngineError() { return engine()->GetLastEngineError(); }
int GetOutputLevel(int channel);
diff --git a/talk/session/media/channel.cc b/talk/session/media/channel.cc
index 4288291..294681e 100644
--- a/talk/session/media/channel.cc
+++ b/talk/session/media/channel.cc
@@ -322,14 +322,6 @@
return true;
}
-bool BaseChannel::MuteStream(uint32 ssrc, bool mute) {
- return InvokeOnWorker(Bind(&BaseChannel::MuteStream_w, this, ssrc, mute));
-}
-
-bool BaseChannel::IsStreamMuted(uint32 ssrc) {
- return InvokeOnWorker(Bind(&BaseChannel::IsStreamMuted_w, this, ssrc));
-}
-
bool BaseChannel::AddRecvStream(const StreamParams& sp) {
return InvokeOnWorker(Bind(&BaseChannel::AddRecvStream_w, this, sp));
}
@@ -723,23 +715,6 @@
ChangeState();
}
-bool BaseChannel::MuteStream_w(uint32 ssrc, bool mute) {
- ASSERT(worker_thread_ == rtc::Thread::Current());
- bool ret = media_channel()->MuteStream(ssrc, mute);
- if (ret) {
- if (mute)
- muted_streams_.insert(ssrc);
- else
- muted_streams_.erase(ssrc);
- }
- return ret;
-}
-
-bool BaseChannel::IsStreamMuted_w(uint32 ssrc) {
- ASSERT(worker_thread_ == rtc::Thread::Current());
- return muted_streams_.find(ssrc) != muted_streams_.end();
-}
-
void BaseChannel::ChannelWritable_w() {
ASSERT(worker_thread_ == rtc::Thread::Current());
if (writable_)
@@ -1289,9 +1264,11 @@
media_channel(), ssrc, renderer));
}
-bool VoiceChannel::SetLocalRenderer(uint32 ssrc, AudioRenderer* renderer) {
- return InvokeOnWorker(Bind(&VoiceMediaChannel::SetLocalRenderer,
- media_channel(), ssrc, renderer));
+bool VoiceChannel::SetAudioSend(uint32 ssrc, bool mute,
+ const AudioOptions* options,
+ AudioRenderer* renderer) {
+ return InvokeOnWorker(Bind(&VoiceMediaChannel::SetAudioSend,
+ media_channel(), ssrc, mute, options, renderer));
}
bool VoiceChannel::SetRingbackTone(const void* buf, int len) {
@@ -1560,11 +1537,6 @@
return media_channel()->InsertDtmf(ssrc, event, duration, flags);
}
-bool VoiceChannel::SetChannelOptions(const AudioOptions& options) {
- return InvokeOnWorker(Bind(&VoiceMediaChannel::SetOptions,
- media_channel(), options));
-}
-
void VoiceChannel::OnMessage(rtc::Message *pmsg) {
switch (pmsg->message_id) {
case MSG_EARLYMEDIATIMEOUT:
@@ -1735,6 +1707,12 @@
return true;
}
+bool VideoChannel::SetVideoSend(uint32 ssrc, bool mute,
+ const VideoOptions* options) {
+ return InvokeOnWorker(Bind(&VideoMediaChannel::SetVideoSend,
+ media_channel(), ssrc, mute, options));
+}
+
void VideoChannel::ChangeState() {
// Render incoming data if we're the active call, and we have the local
// content. We receive data on the default channel and multiplexed streams.
@@ -1952,11 +1930,6 @@
SignalScreencastWindowEvent(ssrc, we);
}
-bool VideoChannel::SetChannelOptions(const VideoOptions &options) {
- return InvokeOnWorker(Bind(&VideoMediaChannel::SetOptions,
- media_channel(), options));
-}
-
void VideoChannel::OnMessage(rtc::Message *pmsg) {
switch (pmsg->message_id) {
case MSG_SCREENCASTWINDOWEVENT: {
@@ -2066,7 +2039,6 @@
}
}
-
void VideoChannel::GetSrtpCiphers(std::vector<std::string>* ciphers) const {
GetSupportedVideoCryptoSuites(ciphers);
}
diff --git a/talk/session/media/channel.h b/talk/session/media/channel.h
index c09d51c..bb430bf 100644
--- a/talk/session/media/channel.h
+++ b/talk/session/media/channel.h
@@ -61,7 +61,7 @@
};
// BaseChannel contains logic common to voice and video, including
-// enable/mute, marshaling calls to a worker thread, and
+// enable, marshaling calls to a worker thread, and
// connection and media monitors.
//
// WARNING! SUBCLASSES MUST CALL Deinit() IN THEIR DESTRUCTORS!
@@ -103,7 +103,6 @@
bool secure_required() const { return secure_required_; }
bool writable() const { return writable_; }
- bool IsStreamMuted(uint32 ssrc);
// Activate RTCP mux, regardless of the state so far. Once
// activated, it can not be deactivated, and if the remote
@@ -125,9 +124,6 @@
std::string* error_desc);
bool Enable(bool enable);
- // Mute sending media on the stream with SSRC |ssrc|
- // If there is only one sending stream SSRC 0 can be used.
- bool MuteStream(uint32 ssrc, bool mute);
// Multiplexing
bool AddRecvStream(const StreamParams& sp);
@@ -227,8 +223,6 @@
void EnableMedia_w();
void DisableMedia_w();
- bool MuteStream_w(uint32 ssrc, bool mute);
- bool IsStreamMuted_w(uint32 ssrc);
void ChannelWritable_w();
void ChannelNotWritable_w();
bool AddRecvStream_w(const StreamParams& sp);
@@ -319,7 +313,6 @@
bool was_ever_writable_;
MediaContentDirection local_content_direction_;
MediaContentDirection remote_content_direction_;
- std::set<uint32> muted_streams_;
bool has_received_packet_;
bool dtls_keyed_;
bool secure_required_;
@@ -336,7 +329,11 @@
~VoiceChannel();
bool Init();
bool SetRemoteRenderer(uint32 ssrc, AudioRenderer* renderer);
- bool SetLocalRenderer(uint32 ssrc, AudioRenderer* renderer);
+
+ // Configure sending media on the stream with SSRC |ssrc|
+ // If there is only one sending stream SSRC 0 can be used.
+ bool SetAudioSend(uint32 ssrc, bool mute, const AudioOptions* options,
+ AudioRenderer* renderer);
// downcasts a MediaChannel
virtual VoiceMediaChannel* media_channel() const {
@@ -387,9 +384,6 @@
sigslot::signal3<VoiceChannel*, uint32, VoiceMediaChannel::Error>
SignalMediaError;
- // Configuration and setting.
- bool SetChannelOptions(const AudioOptions& options);
-
private:
// overrides from BaseChannel
virtual void OnChannelRead(TransportChannel* channel,
@@ -480,8 +474,9 @@
sigslot::signal3<VideoChannel*, uint32, VideoMediaChannel::Error>
SignalMediaError;
- // Configuration and setting.
- bool SetChannelOptions(const VideoOptions& options);
+ // Configure sending media on the stream with SSRC |ssrc|
+ // If there is only one sending stream SSRC 0 can be used.
+ bool SetVideoSend(uint32 ssrc, bool mute, const VideoOptions* options);
private:
typedef std::map<uint32, VideoCapturer*> ScreencastMap;
diff --git a/talk/session/media/channel_unittest.cc b/talk/session/media/channel_unittest.cc
index fa182c3..a7d5c15 100644
--- a/talk/session/media/channel_unittest.cc
+++ b/talk/session/media/channel_unittest.cc
@@ -894,28 +894,6 @@
EXPECT_TRUE(media_channel1_->sending());
}
- void TestMuteStream() {
- CreateChannels(0, 0);
- // Test that we can Mute the default channel even though the sending SSRC is
- // unknown.
- EXPECT_FALSE(media_channel1_->IsStreamMuted(0));
- EXPECT_TRUE(channel1_->MuteStream(0, true));
- EXPECT_TRUE(media_channel1_->IsStreamMuted(0));
- EXPECT_TRUE(channel1_->MuteStream(0, false));
- EXPECT_FALSE(media_channel1_->IsStreamMuted(0));
-
- // Test that we can not mute an unknown SSRC.
- EXPECT_FALSE(channel1_->MuteStream(kSsrc1, true));
-
- SendInitiate();
- // After the local session description has been set, we can mute a stream
- // with its SSRC.
- EXPECT_TRUE(channel1_->MuteStream(kSsrc1, true));
- EXPECT_TRUE(media_channel1_->IsStreamMuted(kSsrc1));
- EXPECT_TRUE(channel1_->MuteStream(kSsrc1, false));
- EXPECT_FALSE(media_channel1_->IsStreamMuted(kSsrc1));
- }
-
// Test that changing the MediaContentDirection in the local and remote
// session description start playout and sending at the right time.
void TestMediaContentDirection() {
@@ -1832,30 +1810,9 @@
class VoiceChannelTest
: public ChannelTest<VoiceTraits> {
public:
- typedef ChannelTest<VoiceTraits>
- Base;
+ typedef ChannelTest<VoiceTraits> Base;
VoiceChannelTest() : Base(kPcmuFrame, sizeof(kPcmuFrame),
- kRtcpReport, sizeof(kRtcpReport)) {
- }
-
- void TestSetChannelOptions() {
- CreateChannels(0, 0);
-
- cricket::AudioOptions options1;
- options1.echo_cancellation.Set(false);
- cricket::AudioOptions options2;
- options2.echo_cancellation.Set(true);
-
- channel1_->SetChannelOptions(options1);
- channel2_->SetChannelOptions(options1);
- EXPECT_EQ(options1, media_channel1_->options());
- EXPECT_EQ(options1, media_channel2_->options());
-
- channel1_->SetChannelOptions(options2);
- channel2_->SetChannelOptions(options2);
- EXPECT_EQ(options2, media_channel1_->options());
- EXPECT_EQ(options2, media_channel2_->options());
- }
+ kRtcpReport, sizeof(kRtcpReport)) {}
};
// override to add NULL parameter
@@ -1917,29 +1874,9 @@
class VideoChannelTest
: public ChannelTest<VideoTraits> {
public:
- typedef ChannelTest<VideoTraits>
- Base;
+ typedef ChannelTest<VideoTraits> Base;
VideoChannelTest() : Base(kH264Packet, sizeof(kH264Packet),
- kRtcpReport, sizeof(kRtcpReport)) {
- }
-
- void TestSetChannelOptions() {
- CreateChannels(0, 0);
-
- cricket::VideoOptions o1;
- o1.video_noise_reduction.Set(true);
-
- channel1_->SetChannelOptions(o1);
- channel2_->SetChannelOptions(o1);
- EXPECT_EQ(o1, media_channel1_->options());
- EXPECT_EQ(o1, media_channel2_->options());
-
- o1.video_start_bitrate.Set(123);
- channel1_->SetChannelOptions(o1);
- channel2_->SetChannelOptions(o1);
- EXPECT_EQ(o1, media_channel1_->options());
- EXPECT_EQ(o1, media_channel2_->options());
- }
+ kRtcpReport, sizeof(kRtcpReport)) {}
};
@@ -1992,7 +1929,25 @@
}
TEST_F(VoiceChannelTest, TestMuteStream) {
- Base::TestMuteStream();
+ CreateChannels(0, 0);
+ // Test that we can Mute the default channel even though the sending SSRC
+ // is unknown.
+ EXPECT_FALSE(media_channel1_->IsStreamMuted(0));
+ EXPECT_TRUE(channel1_->SetAudioSend(0, true, nullptr, nullptr));
+ EXPECT_TRUE(media_channel1_->IsStreamMuted(0));
+ EXPECT_TRUE(channel1_->SetAudioSend(0, false, nullptr, nullptr));
+ EXPECT_FALSE(media_channel1_->IsStreamMuted(0));
+
+ // Test that we can not mute an unknown SSRC.
+ EXPECT_FALSE(channel1_->SetAudioSend(kSsrc1, true, nullptr, nullptr));
+
+ SendInitiate();
+ // After the local session description has been set, we can mute a stream
+ // with its SSRC.
+ EXPECT_TRUE(channel1_->SetAudioSend(kSsrc1, true, nullptr, nullptr));
+ EXPECT_TRUE(media_channel1_->IsStreamMuted(kSsrc1));
+ EXPECT_TRUE(channel1_->SetAudioSend(kSsrc1, false, nullptr, nullptr));
+ EXPECT_FALSE(media_channel1_->IsStreamMuted(kSsrc1));
}
TEST_F(VoiceChannelTest, TestMediaContentDirection) {
@@ -2106,17 +2061,6 @@
Base::TestMediaMonitor();
}
-// Test that MuteStream properly forwards to the media channel and does
-// not signal.
-TEST_F(VoiceChannelTest, TestVoiceSpecificMuteStream) {
- CreateChannels(0, 0);
- EXPECT_FALSE(media_channel1_->IsStreamMuted(0));
- EXPECT_TRUE(channel1_->MuteStream(0, true));
- EXPECT_TRUE(media_channel1_->IsStreamMuted(0));
- EXPECT_TRUE(channel1_->MuteStream(0, false));
- EXPECT_FALSE(media_channel1_->IsStreamMuted(0));
-}
-
// Test that PressDTMF properly forwards to the media channel.
TEST_F(VoiceChannelTest, TestDtmf) {
CreateChannels(0, 0);
@@ -2314,10 +2258,6 @@
kAudioPts, ARRAY_SIZE(kAudioPts), true, true);
}
-TEST_F(VoiceChannelTest, TestSetChannelOptions) {
- TestSetChannelOptions();
-}
-
// VideoChannelTest
TEST_F(VideoChannelTest, TestInit) {
Base::TestInit();
@@ -2394,7 +2334,23 @@
}
TEST_F(VideoChannelTest, TestMuteStream) {
- Base::TestMuteStream();
+ CreateChannels(0, 0);
+ // Test that we can Mute the default channel even though the sending SSRC
+ // is unknown.
+ EXPECT_FALSE(media_channel1_->IsStreamMuted(0));
+ EXPECT_TRUE(channel1_->SetVideoSend(0, true, nullptr));
+ EXPECT_TRUE(media_channel1_->IsStreamMuted(0));
+ EXPECT_TRUE(channel1_->SetVideoSend(0, false, nullptr));
+ EXPECT_FALSE(media_channel1_->IsStreamMuted(0));
+ // Test that we can not mute an unknown SSRC.
+ EXPECT_FALSE(channel1_->SetVideoSend(kSsrc1, true, nullptr));
+ SendInitiate();
+ // After the local session description has been set, we can mute a stream
+ // with its SSRC.
+ EXPECT_TRUE(channel1_->SetVideoSend(kSsrc1, true, nullptr));
+ EXPECT_TRUE(media_channel1_->IsStreamMuted(kSsrc1));
+ EXPECT_TRUE(channel1_->SetVideoSend(kSsrc1, false, nullptr));
+ EXPECT_FALSE(media_channel1_->IsStreamMuted(kSsrc1));
}
TEST_F(VideoChannelTest, TestMediaContentDirection) {
@@ -2624,10 +2580,6 @@
EXPECT_EQ(0, send_format.height);
}
-TEST_F(VideoChannelTest, TestSetChannelOptions) {
- TestSetChannelOptions();
-}
-
// DataChannelTest
