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Unified Diff: talk/media/webrtc/webrtcvoiceengine.cc

Issue 1551813002: Storing raw audio sink for default audio track. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Minor cleanup. Created 4 years, 11 months ago
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Index: talk/media/webrtc/webrtcvoiceengine.cc
diff --git a/talk/media/webrtc/webrtcvoiceengine.cc b/talk/media/webrtc/webrtcvoiceengine.cc
index 9742564985bca6c0437865dafbe76803608e384a..79c807564a504e71939687582447e8fb749ded89 100644
--- a/talk/media/webrtc/webrtcvoiceengine.cc
+++ b/talk/media/webrtc/webrtcvoiceengine.cc
@@ -132,6 +132,16 @@ const int kMaxTelephoneEventCode = 255;
const int kMinTelephoneEventDuration = 100;
const int kMaxTelephoneEventDuration = 60000; // Actual limit is 2^16
+class ProxySink : public webrtc::AudioSinkInterface {
+ public:
+ ProxySink(AudioSinkInterface* sink) : sink_(sink) {}
+
+ void OnData(const Data& audio) override { sink_->OnData(audio); }
+
+ private:
+ webrtc::AudioSinkInterface* sink_;
+};
+
bool ValidateStreamParams(const StreamParams& sp) {
if (sp.ssrcs.empty()) {
LOG(LS_ERROR) << "No SSRCs in stream parameters: " << sp.ToString();
@@ -2186,6 +2196,9 @@ void WebRtcVoiceMediaChannel::OnPacketReceived(
}
default_recv_ssrc_ = ssrc;
SetOutputVolume(default_recv_ssrc_, default_recv_volume_);
+ rtc::scoped_ptr<webrtc::AudioSinkInterface> proxy_sink(
+ default_sink_ ? new ProxySink(default_sink_.get()) : nullptr);
+ SetRawAudioSink(default_recv_ssrc_, std::move(proxy_sink));
tommi 2016/01/13 23:14:24 Do we actually want to make this call if default_s
Taylor Brandstetter 2016/01/13 23:55:12 I had "if (default_sink_)" originally, but I was s
the sun 2016/01/14 09:24:39 Yes, now that we can avoid allocating an object it
}
// Forward packet to Call. If the SSRC is unknown we'll return after this.
@@ -2414,7 +2427,17 @@ void WebRtcVoiceMediaChannel::SetRawAudioSink(
uint32_t ssrc,
rtc::scoped_ptr<webrtc::AudioSinkInterface> sink) {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
- LOG(LS_VERBOSE) << "WebRtcVoiceMediaChannel::SetRawAudioSink";
+ LOG(LS_VERBOSE) << "WebRtcVoiceMediaChannel::SetRawAudioSink: ssrc:" << ssrc
+ << " " << (sink ? "(ptr)" : "NULL");
+ if (ssrc == 0) {
+ if (default_recv_ssrc_ != -1) {
+ rtc::scoped_ptr<webrtc::AudioSinkInterface> proxy_sink(
+ sink ? new ProxySink(sink.get()) : nullptr);
+ SetRawAudioSink(default_recv_ssrc_, std::move(proxy_sink));
+ }
+ default_sink_ = std::move(sink);
tommi 2016/01/13 23:14:24 nit: do you mind moving this above the check for -
Taylor Brandstetter 2016/01/13 23:55:12 Is the sink accessed on any other threads? If so I
the sun 2016/01/14 09:24:39 I'd still like you to avoid the duplicate log line
tommi (sloooow) - chröme 2016/01/14 12:34:13 ah of course
Taylor Brandstetter 2016/01/14 15:48:47 My reasoning is that SetRenderer does the same thi
the sun 2016/01/14 15:51:44 Acknowledged.
+ return;
+ }
const auto it = recv_streams_.find(ssrc);
if (it == recv_streams_.end()) {
LOG(LS_WARNING) << "SetRawAudioSink: no recv stream" << ssrc;

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