Storing raw audio sink for default audio track.

BUG=webrtc:5250

Review URL: https://codereview.webrtc.org/1551813002

Cr-Commit-Position: refs/heads/master@{#11230}

8 files changed, 19 insertions, 14 deletions

diff --git a/webrtc/audio/audio_receive_stream.cc b/webrtc/audio/audio_receive_stream.cc
index 64d008326d..8b31130554 100644
--- a/webrtc/audio/audio_receive_stream.cc
+++ b/webrtc/audio/audio_receive_stream.cc
@@ -235,9 +235,10 @@ webrtc::AudioReceiveStream::Stats AudioReceiveStream::GetStats() const {
   return stats;
 }
 
-void AudioReceiveStream::SetSink(rtc::scoped_ptr<AudioSinkInterface> sink) {
+void AudioReceiveStream::SetSink(
+    const rtc::scoped_refptr<AudioSinkInterface>& sink) {
   RTC_DCHECK(thread_checker_.CalledOnValidThread());
-  channel_proxy_->SetSink(std::move(sink));
+  channel_proxy_->SetSink(sink);
 }
 
 const webrtc::AudioReceiveStream::Config& AudioReceiveStream::config() const {
diff --git a/webrtc/audio/audio_receive_stream.h b/webrtc/audio/audio_receive_stream.h
index 4940c6a64c..7a070be2e0 100644
--- a/webrtc/audio/audio_receive_stream.h
+++ b/webrtc/audio/audio_receive_stream.h
@@ -45,7 +45,7 @@ class AudioReceiveStream final : public webrtc::AudioReceiveStream {
   // webrtc::AudioReceiveStream implementation.
   webrtc::AudioReceiveStream::Stats GetStats() const override;
 
-  void SetSink(rtc::scoped_ptr<AudioSinkInterface> sink) override;
+  void SetSink(const rtc::scoped_refptr<AudioSinkInterface>& sink) override;
 
   const webrtc::AudioReceiveStream::Config& config() const;
 
diff --git a/webrtc/audio/audio_sink.h b/webrtc/audio/audio_sink.h
index 999644f4ce..50ce7a109a 100644
--- a/webrtc/audio/audio_sink.h
+++ b/webrtc/audio/audio_sink.h
@@ -19,10 +19,12 @@
 #include <inttypes.h>
 #include <stddef.h>
 
+#include "webrtc/base/refcount.h"
+
 namespace webrtc {
 
 // Represents a simple push audio sink.
-class AudioSinkInterface {
+class AudioSinkInterface : public rtc::RefCountInterface {
  public:
   virtual ~AudioSinkInterface() {}
 
diff --git a/webrtc/audio_receive_stream.h b/webrtc/audio_receive_stream.h
index 8cab094f4b..062aedb43e 100644
--- a/webrtc/audio_receive_stream.h
+++ b/webrtc/audio_receive_stream.h
@@ -15,7 +15,7 @@
 #include <string>
 #include <vector>
 
-#include "webrtc/base/scoped_ptr.h"
+#include "webrtc/base/scoped_ref_ptr.h"
 #include "webrtc/config.h"
 #include "webrtc/stream.h"
 #include "webrtc/transport.h"
@@ -112,12 +112,12 @@ class AudioReceiveStream : public ReceiveStream {
   // Sets an audio sink that receives unmixed audio from the receive stream.
   // Ownership of the sink is passed to the stream and can be used by the
   // caller to do lifetime management (i.e. when the sink's dtor is called).
-  // Only one sink can be set and passing a null sink, clears an existing one.
+  // Only one sink can be set and passing a null sink clears an existing one.
   // NOTE: Audio must still somehow be pulled through AudioTransport for audio
   // to stream through this sink. In practice, this happens if mixed audio
   // is being pulled+rendered and/or if audio is being pulled for the purposes
   // of feeding to the AEC.
-  virtual void SetSink(rtc::scoped_ptr<AudioSinkInterface> sink) = 0;
+  virtual void SetSink(const rtc::scoped_refptr<AudioSinkInterface>& sink) = 0;
 };
 }  // namespace webrtc
 
diff --git a/webrtc/voice_engine/channel.cc b/webrtc/voice_engine/channel.cc
index b1b55e8acc..de7d13a10f 100644
--- a/webrtc/voice_engine/channel.cc
+++ b/webrtc/voice_engine/channel.cc
@@ -1185,9 +1185,9 @@ Channel::UpdateLocalTimeStamp()
     return 0;
 }
 
-void Channel::SetSink(rtc::scoped_ptr<AudioSinkInterface> sink) {
+void Channel::SetSink(const rtc::scoped_refptr<AudioSinkInterface>& sink) {
   CriticalSectionScoped cs(&_callbackCritSect);
-  audio_sink_ = std::move(sink);
+  audio_sink_ = sink;
 }
 
 int32_t
diff --git a/webrtc/voice_engine/channel.h b/webrtc/voice_engine/channel.h
index d15f9dbca7..771b3c43ec 100644
--- a/webrtc/voice_engine/channel.h
+++ b/webrtc/voice_engine/channel.h
@@ -14,6 +14,7 @@
 #include "webrtc/audio/audio_sink.h"
 #include "webrtc/base/criticalsection.h"
 #include "webrtc/base/scoped_ptr.h"
+#include "webrtc/base/scoped_ref_ptr.h"
 #include "webrtc/common_audio/resampler/include/push_resampler.h"
 #include "webrtc/common_types.h"
 #include "webrtc/modules/audio_coding/include/audio_coding_module.h"
@@ -193,7 +194,7 @@ public:
         CriticalSectionWrapper* callbackCritSect);
     int32_t UpdateLocalTimeStamp();
 
-    void SetSink(rtc::scoped_ptr<AudioSinkInterface> sink);
+    void SetSink(const rtc::scoped_refptr<AudioSinkInterface>& sink);
 
     // API methods
 
@@ -511,7 +512,7 @@ private:
     TelephoneEventHandler* telephone_event_handler_;
     rtc::scoped_ptr<RtpRtcp> _rtpRtcpModule;
     rtc::scoped_ptr<AudioCodingModule> audio_coding_;
-    rtc::scoped_ptr<AudioSinkInterface> audio_sink_;
+    rtc::scoped_refptr<AudioSinkInterface> audio_sink_;
     AudioLevel _outputAudioLevel;
     bool _externalTransport;
     AudioFrame _audioFrame;
diff --git a/webrtc/voice_engine/channel_proxy.cc b/webrtc/voice_engine/channel_proxy.cc
index f54c81ec47..711f2abd03 100644
--- a/webrtc/voice_engine/channel_proxy.cc
+++ b/webrtc/voice_engine/channel_proxy.cc
@@ -139,9 +139,9 @@ bool ChannelProxy::SendTelephoneEventOutband(uint8_t event,
       channel()->SendTelephoneEventOutband(event, duration_ms, 10, false) == 0;
 }
 
-void ChannelProxy::SetSink(rtc::scoped_ptr<AudioSinkInterface> sink) {
+void ChannelProxy::SetSink(const rtc::scoped_refptr<AudioSinkInterface>& sink) {
   RTC_DCHECK(thread_checker_.CalledOnValidThread());
-  channel()->SetSink(std::move(sink));
+  channel()->SetSink(sink);
 }
 
 Channel* ChannelProxy::channel() const {
diff --git a/webrtc/voice_engine/channel_proxy.h b/webrtc/voice_engine/channel_proxy.h
index b990d91734..ba206cb8bc 100644
--- a/webrtc/voice_engine/channel_proxy.h
+++ b/webrtc/voice_engine/channel_proxy.h
@@ -11,6 +11,7 @@
 #ifndef WEBRTC_VOICE_ENGINE_CHANNEL_PROXY_H_
 #define WEBRTC_VOICE_ENGINE_CHANNEL_PROXY_H_
 
+#include "webrtc/base/scoped_ref_ptr.h"
 #include "webrtc/base/thread_checker.h"
 #include "webrtc/voice_engine/channel_manager.h"
 #include "webrtc/voice_engine/include/voe_rtp_rtcp.h"
@@ -65,7 +66,7 @@ class ChannelProxy {
   virtual bool SetSendTelephoneEventPayloadType(int payload_type);
   virtual bool SendTelephoneEventOutband(uint8_t event, uint32_t duration_ms);
 
-  virtual void SetSink(rtc::scoped_ptr<AudioSinkInterface> sink);
+  virtual void SetSink(const rtc::scoped_refptr<AudioSinkInterface>& sink);
 
  private:
   Channel* channel() const;