Move VoiceEngineObserver into AudioSendStream so that we detect typing noises and return properly i…

webrtc/call.h

 /*
  *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 #ifndef WEBRTC_CALL_H_
 #define WEBRTC_CALL_H_
 
 #include <string>
 #include <vector>
 
 #include "webrtc/common_types.h"
 #include "webrtc/audio_receive_stream.h"
 #include "webrtc/audio_send_stream.h"
+#include "webrtc/audio_state.h"
 #include "webrtc/base/socket.h"
 #include "webrtc/video_receive_stream.h"
 #include "webrtc/video_send_stream.h"
 
 namespace webrtc {
 
-class AudioDeviceModule;
 class AudioProcessing;
-class VoiceEngine;
-class VoiceEngineObserver;
 
 const char* Version();
 
 enum class MediaType {
   ANY,
   AUDIO,
   VIDEO,
   DATA
 };
 
@@ skipping 28 matching lines @@
 };
 
 // A Call instance can contain several send and/or receive streams. All streams
 // are assumed to have the same remote endpoint and will share bitrate estimates
 // etc.
 class Call {
  public:
   struct Config {
     static const int kDefaultStartBitrateBps;
 
-    // VoiceEngine used for audio/video synchronization for this Call.
-    VoiceEngine* voice_engine = nullptr;
-
     // Bitrate config used until valid bitrate estimates are calculated. Also
     // used to cap total bitrate used.
     struct BitrateConfig {
       int min_bitrate_bps = 0;
       int start_bitrate_bps = kDefaultStartBitrateBps;
       int max_bitrate_bps = -1;
     } bitrate_config;
 
-    struct AudioConfig {
-      AudioDeviceModule* audio_device_module = nullptr;
-      AudioProcessing* audio_processing = nullptr;
-      VoiceEngineObserver* voice_engine_observer = nullptr;
-    } audio_config;
+    // AudioState which is possibly shared between multiple calls.
+    // TODO(solenberg): Change this to a shared_ptr once we can use C++11.
+    rtc::linked_ptr<AudioState> audio_state;
+
+    // Audio Processing Module to be used in this call.
+    // TODO(solenberg): Change this to a shared_ptr once we can use C++11.
+    AudioProcessing* audio_processing = nullptr;
   };
 
   struct Stats {
     int send_bandwidth_bps = 0;
     int recv_bandwidth_bps = 0;
     int64_t pacer_delay_ms = 0;
     int64_t rtt_ms = -1;
   };
 
   static Call* Create(const Call::Config& config);
@@ skipping 36 matching lines @@
   virtual void SignalNetworkState(NetworkState state) = 0;
 
   virtual void OnSentPacket(const rtc::SentPacket& sent_packet) = 0;
 
   virtual ~Call() {}
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_CALL_H_