- Introduce internal classes WebRtcAudio[Send|Receive]Stream in WebRtcVoiceMediaChannel.
- Remove WebRtcVoiceMediaChannel::WebRtcVoiceChannelRenderer
- Create webrtc::AudioSendStreams.
BUG=webrtc:4690
Review URL: https://codereview.webrtc.org/1415563003
Cr-Commit-Position: refs/heads/master@{#10361}
diff --git a/talk/media/webrtc/webrtcvoiceengine.cc b/talk/media/webrtc/webrtcvoiceengine.cc
index 5375358..eed1195 100644
--- a/talk/media/webrtc/webrtcvoiceengine.cc
+++ b/talk/media/webrtc/webrtcvoiceengine.cc
@@ -1313,15 +1313,25 @@
return voe_wrapper_->base()->CreateChannel(voe_config_);
}
-class WebRtcVoiceMediaChannel::WebRtcVoiceChannelRenderer
+class WebRtcVoiceMediaChannel::WebRtcAudioSendStream
: public AudioRenderer::Sink {
public:
- WebRtcVoiceChannelRenderer(int ch,
- webrtc::AudioTransport* voe_audio_transport)
+ WebRtcAudioSendStream(int ch, webrtc::AudioTransport* voe_audio_transport,
+ uint32_t ssrc, webrtc::Call* call)
: channel_(ch),
voe_audio_transport_(voe_audio_transport),
- renderer_(NULL) {}
- ~WebRtcVoiceChannelRenderer() override { Stop(); }
+ call_(call) {
+ RTC_DCHECK(call);
+ webrtc::AudioSendStream::Config config(nullptr);
+ config.voe_channel_id = channel_;
+ config.rtp.ssrc = ssrc;
+ stream_ = call_->CreateAudioSendStream(config);
+ RTC_DCHECK(stream_);
+ }
+ ~WebRtcAudioSendStream() override {
+ Stop();
+ call_->DestroyAudioSendStream(stream_);
+ }
// Starts the rendering by setting a sink to the renderer to get data
// callback.
@@ -1329,8 +1339,8 @@
// TODO(xians): Make sure Start() is called only once.
void Start(AudioRenderer* renderer) {
rtc::CritScope lock(&lock_);
- RTC_DCHECK(renderer != NULL);
- if (renderer_ != NULL) {
+ RTC_DCHECK(renderer);
+ if (renderer_) {
RTC_DCHECK(renderer_ == renderer);
return;
}
@@ -1338,14 +1348,14 @@
renderer_ = renderer;
}
- // Stops rendering by setting the sink of the renderer to NULL. No data
+ // Stops rendering by setting the sink of the renderer to nullptr. No data
// callback will be received after this method.
// This method is called on the libjingle worker thread.
void Stop() {
rtc::CritScope lock(&lock_);
- if (renderer_ != NULL) {
- renderer_->SetSink(NULL);
- renderer_ = NULL;
+ if (renderer_) {
+ renderer_->SetSink(nullptr);
+ renderer_ = nullptr;
}
}
@@ -1356,6 +1366,7 @@
int sample_rate,
int number_of_channels,
size_t number_of_frames) override {
+ RTC_DCHECK(voe_audio_transport_);
voe_audio_transport_->OnData(channel_,
audio_data,
bits_per_sample,
@@ -1368,25 +1379,42 @@
// never been called, this callback won't be triggered.
void OnClose() override {
rtc::CritScope lock(&lock_);
- // Set |renderer_| to NULL to make sure no more callback will get into
+ // Set |renderer_| to nullptr to make sure no more callback will get into
// the renderer.
- renderer_ = NULL;
+ renderer_ = nullptr;
}
// Accessor to the VoE channel ID.
int channel() const { return channel_; }
private:
- const int channel_;
- webrtc::AudioTransport* const voe_audio_transport_;
+ const int channel_ = -1;
+ webrtc::AudioTransport* const voe_audio_transport_ = nullptr;
+ webrtc::Call* call_ = nullptr;
+ webrtc::AudioSendStream* stream_ = nullptr;
// Raw pointer to AudioRenderer owned by LocalAudioTrackHandler.
// PeerConnection will make sure invalidating the pointer before the object
// goes away.
- AudioRenderer* renderer_;
+ AudioRenderer* renderer_ = nullptr;
// Protects |renderer_| in Start(), Stop() and OnClose().
rtc::CriticalSection lock_;
+
+ RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(WebRtcAudioSendStream);
+};
+
+class WebRtcVoiceMediaChannel::WebRtcAudioReceiveStream {
+ public:
+ explicit WebRtcAudioReceiveStream(int voe_channel_id)
+ : channel_(voe_channel_id) {}
+
+ int channel() { return channel_; }
+
+ private:
+ int channel_;
+
+ RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(WebRtcAudioReceiveStream);
};
// WebRtcVoiceMediaChannel
@@ -1417,8 +1445,8 @@
LOG(LS_VERBOSE) << "WebRtcVoiceMediaChannel::~WebRtcVoiceMediaChannel";
// Remove any remaining send streams.
- while (!send_channels_.empty()) {
- RemoveSendStream(send_channels_.begin()->first);
+ while (!send_streams_.empty()) {
+ RemoveSendStream(send_streams_.begin()->first);
}
// Remove any remaining receive streams.
@@ -1481,6 +1509,7 @@
}
}
+ // TODO(solenberg): Don't recreate unless options changed.
RecreateAudioReceiveStreams();
LOG(LS_INFO) << "Set voice channel options. Current options: "
@@ -1760,7 +1789,7 @@
// Cache the codecs in order to configure the channel created later.
send_codecs_ = codecs;
- for (const auto& ch : send_channels_) {
+ for (const auto& ch : send_streams_) {
if (!SetSendCodecs(ch.second->channel(), codecs)) {
return false;
}
@@ -1871,7 +1900,7 @@
return true;
}
- for (const auto& ch : send_channels_) {
+ for (const auto& ch : send_streams_) {
if (!SetChannelSendRtpHeaderExtensions(ch.second->channel(), extensions)) {
return false;
}
@@ -1935,8 +1964,9 @@
bool WebRtcVoiceMediaChannel::SetSend(SendFlags send) {
desired_send_ = send;
- if (!send_channels_.empty())
+ if (!send_streams_.empty()) {
return ChangeSend(desired_send_);
+ }
return true;
}
@@ -1959,7 +1989,7 @@
}
// Change the settings on each send channel.
- for (const auto& ch : send_channels_) {
+ for (const auto& ch : send_streams_) {
if (!ChangeSend(ch.second->channel(), send)) {
return false;
}
@@ -2075,13 +2105,13 @@
return false;
}
- // Save the channel to send_channels_, so that RemoveSendStream() can still
+ // Save the channel to send_streams_, so that RemoveSendStream() can still
// delete the channel in case failure happens below.
webrtc::AudioTransport* audio_transport =
engine()->voe()->base()->audio_transport();
- send_channels_.insert(
+ send_streams_.insert(
std::make_pair(ssrc,
- new WebRtcVoiceChannelRenderer(channel, audio_transport)));
+ new WebRtcAudioSendStream(channel, audio_transport, ssrc, call_)));
// Set the current codecs to be used for the new channel. We need to do this
// after adding the channel to send_channels_, because of how max bitrate is
@@ -2094,7 +2124,7 @@
// At this point the channel's local SSRC has been updated. If the channel is
// the first send channel make sure that all the receive channels are updated
// with the same SSRC in order to send receiver reports.
- if (send_channels_.size() == 1) {
+ if (send_streams_.size() == 1) {
receiver_reports_ssrc_ = ssrc;
for (const auto& ch : receive_channels_) {
int recv_channel = ch.second->channel();
@@ -2113,8 +2143,8 @@
bool WebRtcVoiceMediaChannel::RemoveSendStream(uint32_t ssrc) {
RTC_DCHECK(thread_checker_.CalledOnValidThread());
- ChannelMap::iterator it = send_channels_.find(ssrc);
- if (it == send_channels_.end()) {
+ auto it = send_streams_.find(ssrc);
+ if (it == send_streams_.end()) {
LOG(LS_WARNING) << "Try to remove stream with ssrc " << ssrc
<< " which doesn't exist.";
return false;
@@ -2126,7 +2156,7 @@
// Delete the WebRtcVoiceChannelRenderer object connected to the channel,
// this will disconnect the audio renderer with the send channel.
delete it->second;
- send_channels_.erase(it);
+ send_streams_.erase(it);
// Clean up and delete the send channel.
LOG(LS_INFO) << "Removing audio send stream " << ssrc
@@ -2134,7 +2164,7 @@
if (!DeleteChannel(channel)) {
return false;
}
- if (send_channels_.empty()) {
+ if (send_streams_.empty()) {
ChangeSend(SEND_NOTHING);
}
return true;
@@ -2176,11 +2206,8 @@
return false;
}
- webrtc::AudioTransport* audio_transport =
- engine()->voe()->base()->audio_transport();
- WebRtcVoiceChannelRenderer* channel_renderer =
- new WebRtcVoiceChannelRenderer(channel, audio_transport);
- receive_channels_.insert(std::make_pair(ssrc, channel_renderer));
+ WebRtcAudioReceiveStream* stream = new WebRtcAudioReceiveStream(channel);
+ receive_channels_.insert(std::make_pair(ssrc, stream));
receive_stream_params_[ssrc] = sp;
AddAudioReceiveStream(ssrc);
@@ -2249,7 +2276,7 @@
RTC_DCHECK(thread_checker_.CalledOnValidThread());
LOG(LS_INFO) << "RemoveRecvStream: " << ssrc;
- ChannelMap::iterator it = receive_channels_.find(ssrc);
+ auto it = receive_channels_.find(ssrc);
if (it == receive_channels_.end()) {
LOG(LS_WARNING) << "Try to remove stream with ssrc " << ssrc
<< " which doesn't exist.";
@@ -2259,9 +2286,6 @@
RemoveAudioReceiveStream(ssrc);
receive_stream_params_.erase(ssrc);
- // Delete the WebRtcVoiceChannelRenderer object connected to the channel, this
- // will disconnect the audio renderer with the receive channel.
- // Cache the channel before the deletion.
const int channel = it->second->channel();
delete it->second;
receive_channels_.erase(it);
@@ -2278,8 +2302,8 @@
bool WebRtcVoiceMediaChannel::SetLocalRenderer(uint32_t ssrc,
AudioRenderer* renderer) {
- ChannelMap::iterator it = send_channels_.find(ssrc);
- if (it == send_channels_.end()) {
+ auto it = send_streams_.find(ssrc);
+ if (it == send_streams_.end()) {
if (renderer) {
// Return an error if trying to set a valid renderer with an invalid ssrc.
LOG(LS_ERROR) << "SetLocalRenderer failed with ssrc "<< ssrc;
@@ -2388,8 +2412,8 @@
if (flags & cricket::DF_SEND) {
int channel = -1;
if (ssrc == 0) {
- if (send_channels_.size() > 0) {
- channel = send_channels_.begin()->second->channel();
+ if (send_streams_.size() > 0) {
+ channel = send_streams_.begin()->second->channel();
}
} else {
channel = GetSendChannelId(ssrc);
@@ -2499,7 +2523,7 @@
// SR may continue RR and any RR entry may correspond to any one of the send
// channels. So all RTCP packets must be forwarded all send channels. VoE
// will filter out RR internally.
- for (const auto& ch : send_channels_) {
+ for (const auto& ch : send_streams_) {
engine()->voe()->network()->ReceivedRTCPPacket(
ch.second->channel(), packet->data(), packet->size());
}
@@ -2521,7 +2545,7 @@
// the mic channel is muted/unmuted. We can't do it today because there
// is no good way to know which stream is mapping to the mic channel.
bool all_muted = muted;
- for (const auto& ch : send_channels_) {
+ for (const auto& ch : send_streams_) {
if (!all_muted) {
break;
}
@@ -2570,7 +2594,7 @@
if (is_multi_rate) {
// If codec is multi-rate then just set the bitrate.
codec.rate = bps;
- for (const auto& ch : send_channels_) {
+ for (const auto& ch : send_streams_) {
if (!SetSendCodec(ch.second->channel(), codec)) {
LOG(LS_INFO) << "Failed to set codec " << codec.plname
<< " to bitrate " << bps << " bps.";
@@ -2630,7 +2654,7 @@
webrtc::CodecInst codec;
unsigned int level;
- for (const auto& ch : send_channels_) {
+ for (const auto& ch : send_streams_) {
const int channel = ch.second->channel();
// Fill in the sender info, based on what we know, and what the
@@ -2791,7 +2815,7 @@
int WebRtcVoiceMediaChannel::GetReceiveChannelId(uint32_t ssrc) const {
RTC_DCHECK(thread_checker_.CalledOnValidThread());
- ChannelMap::const_iterator it = receive_channels_.find(ssrc);
+ const auto it = receive_channels_.find(ssrc);
if (it != receive_channels_.end()) {
return it->second->channel();
}
@@ -2800,8 +2824,8 @@
int WebRtcVoiceMediaChannel::GetSendChannelId(uint32_t ssrc) const {
RTC_DCHECK(thread_checker_.CalledOnValidThread());
- ChannelMap::const_iterator it = send_channels_.find(ssrc);
- if (it != send_channels_.end()) {
+ const auto it = send_streams_.find(ssrc);
+ if (it != send_streams_.end()) {
return it->second->channel();
}
return -1;
@@ -2927,8 +2951,8 @@
void WebRtcVoiceMediaChannel::AddAudioReceiveStream(uint32_t ssrc) {
RTC_DCHECK(thread_checker_.CalledOnValidThread());
- WebRtcVoiceChannelRenderer* channel = receive_channels_[ssrc];
- RTC_DCHECK(channel != nullptr);
+ WebRtcAudioReceiveStream* stream = receive_channels_[ssrc];
+ RTC_DCHECK(stream != nullptr);
RTC_DCHECK(receive_streams_.find(ssrc) == receive_streams_.end());
webrtc::AudioReceiveStream::Config config;
config.rtp.remote_ssrc = ssrc;
@@ -2936,7 +2960,7 @@
config.rtp.extensions = recv_rtp_extensions_;
config.combined_audio_video_bwe =
options_.combined_audio_video_bwe.GetWithDefaultIfUnset(false);
- config.voe_channel_id = channel->channel();
+ config.voe_channel_id = stream->channel();
config.sync_group = receive_stream_params_[ssrc].sync_label;
webrtc::AudioReceiveStream* s = call_->CreateAudioReceiveStream(config);
receive_streams_.insert(std::make_pair(ssrc, s));