1 files changed, 15 insertions, 8 deletions
diff --git a/webrtc/audio/audio_send_stream.h b/webrtc/audio/audio_send_stream.h
index ae81dfc8fc..8b96350590 100644
--- a/webrtc/audio/audio_send_stream.h
+++ b/webrtc/audio/audio_send_stream.h
@@ -12,20 +12,24 @@
 #define WEBRTC_AUDIO_AUDIO_SEND_STREAM_H_
 
 #include "webrtc/audio_send_stream.h"
-#include "webrtc/audio/scoped_voe_interface.h"
+#include "webrtc/audio_state.h"
 #include "webrtc/base/thread_checker.h"
-#include "webrtc/voice_engine/include/voe_base.h"
+#include "webrtc/base/scoped_ptr.h"
 
 namespace webrtc {
-
+class CongestionController;
 class VoiceEngine;
 
-namespace internal {
+namespace voe {
+class ChannelProxy;
+}  // namespace voe
 
+namespace internal {
 class AudioSendStream final : public webrtc::AudioSendStream {
  public:
   AudioSendStream(const webrtc::AudioSendStream::Config& config,
-                  VoiceEngine* voice_engine);
+                  const rtc::scoped_refptr<webrtc::AudioState>& audio_state,
+                  CongestionController* congestion_controller);
   ~AudioSendStream() override;
 
   // webrtc::SendStream implementation.
@@ -35,16 +39,19 @@ class AudioSendStream final : public webrtc::AudioSendStream {
   bool DeliverRtcp(const uint8_t* packet, size_t length) override;
 
   // webrtc::AudioSendStream implementation.
+  bool SendTelephoneEvent(int payload_type, uint8_t event,
+                          uint32_t duration_ms) override;
   webrtc::AudioSendStream::Stats GetStats() const override;
 
   const webrtc::AudioSendStream::Config& config() const;
 
  private:
+  VoiceEngine* voice_engine() const;
+
   rtc::ThreadChecker thread_checker_;
   const webrtc::AudioSendStream::Config config_;
-  VoiceEngine* voice_engine_;
-  // We hold one interface pointer to the VoE to make sure it is kept alive.
-  ScopedVoEInterface<VoEBase> voe_base_;
+  rtc::scoped_refptr<webrtc::AudioState> audio_state_;
+  rtc::scoped_ptr<voe::ChannelProxy> channel_proxy_;
 
   RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(AudioSendStream);
 };