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

talk/media/webrtc/webrtcvoiceengine.h

 /*
  * libjingle
  * Copyright 2004 Google Inc.
  *
  * Redistribution and use in source and binary forms, with or without
  * modification, are permitted provided that the following conditions are met:
  *
  *  1. Redistributions of source code must retain the above copyright notice,
  *     this list of conditions and the following disclaimer.
  *  2. Redistributions in binary form must reproduce the above copyright notice,
@@ skipping 19 matching lines @@
 
 #include <map>
 #include <set>
 #include <string>
 #include <vector>
 
 #include "talk/media/base/rtputils.h"
 #include "talk/media/webrtc/webrtccommon.h"
 #include "talk/media/webrtc/webrtcvoe.h"
 #include "talk/session/media/channel.h"
+#include "webrtc/audio_state.h"
 #include "webrtc/base/buffer.h"
 #include "webrtc/base/byteorder.h"
 #include "webrtc/base/logging.h"
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/base/stream.h"
 #include "webrtc/base/thread_checker.h"
 #include "webrtc/call.h"
 #include "webrtc/common.h"
 #include "webrtc/config.h"
 
 namespace cricket {
 
 class AudioDeviceModule;
 class AudioRenderer;
 class VoETraceWrapper;
 class VoEWrapper;
 class WebRtcVoiceMediaChannel;
 
 // WebRtcVoiceEngine is a class to be used with CompositeMediaEngine.
 // It uses the WebRtc VoiceEngine library for audio handling.
-class WebRtcVoiceEngine
-    : public webrtc::VoiceEngineObserver,
-      public webrtc::TraceCallback  {
+class WebRtcVoiceEngine final : public webrtc::TraceCallback  {
   friend class WebRtcVoiceMediaChannel;
 
  public:
   WebRtcVoiceEngine();
   // Dependency injection for testing.
-  WebRtcVoiceEngine(VoEWrapper* voe_wrapper, VoETraceWrapper* tracing);
+  WebRtcVoiceEngine(VoEWrapper* voe_wrapper,
+                    VoETraceWrapper* tracing,
+                    webrtc::AudioState* audio_state);
   ~WebRtcVoiceEngine();
   bool Init(rtc::Thread* worker_thread);
   void Terminate();
 
-  webrtc::VoiceEngine* GetVoE() { return voe()->engine(); }
+  webrtc::AudioState* GetAudioState() { return audio_state_.get(); }
   VoiceMediaChannel* CreateChannel(webrtc::Call* call,
                                    const AudioOptions& options);
 
   AudioOptions GetOptions() const { return options_; }
   bool SetOptions(const AudioOptions& options);
   bool SetDevices(const Device* in_device, const Device* out_device);
   bool GetOutputVolume(int* level);
   bool SetOutputVolume(int level);
   int GetInputLevel();
 
@@ skipping 42 matching lines @@
   void SetTraceFilter(int filter);
   void SetTraceOptions(const std::string& options);
   // Every option that is "set" will be applied. Every option not "set" will be
   // ignored. This allows us to selectively turn on and off different options
   // easily at any time.
   bool ApplyOptions(const AudioOptions& options);
 
   // webrtc::TraceCallback:
   void Print(webrtc::TraceLevel level, const char* trace, int length) override;
 
-  // webrtc::VoiceEngineObserver:
-  void CallbackOnError(int channel_id, int errCode) override;
-
   // Given the device type, name, and id, find device id. Return true and
   // set the output parameter rtc_id if successful.
   bool FindWebRtcAudioDeviceId(
       bool is_input, const std::string& dev_name, int dev_id, int* rtc_id);
 
   void StartAecDump(const std::string& filename);
   int CreateVoEChannel();
 
   static const int kDefaultLogSeverity = rtc::LS_WARNING;
 
   // The primary instance of WebRtc VoiceEngine.
   rtc::scoped_ptr<VoEWrapper> voe_wrapper_;
   rtc::scoped_ptr<VoETraceWrapper> tracing_;
+  rtc::scoped_ptr<webrtc::AudioState> audio_state_;
   // The external audio device manager
-  webrtc::AudioDeviceModule* adm_;
+  webrtc::AudioDeviceModule* adm_ = nullptr;
   int log_filter_;
   std::string log_options_;
-  bool is_dumping_aec_;
+  bool is_dumping_aec_ = false;
   std::vector<AudioCodec> codecs_;
   std::vector<RtpHeaderExtension> rtp_header_extensions_;
   std::vector<WebRtcVoiceMediaChannel*> channels_;
-  // channels_ can be read from WebRtc callback thread. We need a lock on that
-  // callback as well as the RegisterChannel/UnregisterChannel.
-  rtc::CriticalSection channels_cs_;
   webrtc::AgcConfig default_agc_config_;
 
   webrtc::Config voe_config_;
 
-  bool initialized_;
+  bool initialized_ = false;
   AudioOptions options_;
 
   // Cache received extended_filter_aec, delay_agnostic_aec and experimental_ns
   // values, and apply them in case they are missing in the audio options. We
   // need to do this because SetExtraOptions() will revert to defaults for
   // options which are not provided.
   Settable<bool> extended_filter_aec_;
   Settable<bool> delay_agnostic_aec_;
   Settable<bool> experimental_ns_;
 
   RTC_DISALLOW_COPY_AND_ASSIGN(WebRtcVoiceEngine);
 };
 
 // WebRtcVoiceMediaChannel is an implementation of VoiceMediaChannel that uses
 // WebRtc Voice Engine.
-class WebRtcVoiceMediaChannel : public VoiceMediaChannel,
-                                public webrtc::Transport {
+class WebRtcVoiceMediaChannel final : public VoiceMediaChannel,
+                                      public webrtc::Transport {
  public:
   WebRtcVoiceMediaChannel(WebRtcVoiceEngine* engine,
                           const AudioOptions& options,
                           webrtc::Call* call);
   ~WebRtcVoiceMediaChannel() override;
 
   const AudioOptions& options() const { return options_; }
 
   bool SetSendParameters(const AudioSendParameters& params) override;
   bool SetRecvParameters(const AudioRecvParameters& params) override;
@@ skipping 41 matching lines @@
     rtc_options.packet_id = options.packet_id;
     return VoiceMediaChannel::SendPacket(&packet, rtc_options);
   }
 
   bool SendRtcp(const uint8_t* data, size_t len) override {
     rtc::Buffer packet(reinterpret_cast<const uint8_t*>(data), len,
                        kMaxRtpPacketLen);
     return VoiceMediaChannel::SendRtcp(&packet, rtc::PacketOptions());
   }
 
-  void OnError(int error);
-
   int GetReceiveChannelId(uint32_t ssrc) const;
   int GetSendChannelId(uint32_t ssrc) const;
 
  private:
   bool SetSendCodecs(const std::vector<AudioCodec>& codecs);
   bool SetSendRtpHeaderExtensions(
       const std::vector<RtpHeaderExtension>& extensions);
   bool SetOptions(const AudioOptions& options);
   bool SetMaxSendBandwidth(int bps);
   bool SetRecvCodecs(const std::vector<AudioCodec>& codecs);
   bool SetRecvRtpHeaderExtensions(
       const std::vector<RtpHeaderExtension>& extensions);
   bool SetLocalRenderer(uint32_t ssrc, AudioRenderer* renderer);
   bool MuteStream(uint32_t ssrc, bool mute);
 
   WebRtcVoiceEngine* engine() { return engine_; }
   int GetLastEngineError() { return engine()->GetLastEngineError(); }
   int GetOutputLevel(int channel);
   bool GetRedSendCodec(const AudioCodec& red_codec,
                        const std::vector<AudioCodec>& all_codecs,
                        webrtc::CodecInst* send_codec);
   bool SetPlayout(int channel, bool playout);
-  static Error WebRtcErrorToChannelError(int err_code);
 
   typedef int (webrtc::VoERTP_RTCP::* ExtensionSetterFunction)(int, bool,
       unsigned char);
 
   void SetNack(int channel, bool nack_enabled);
   bool SetSendCodec(int channel, const webrtc::CodecInst& send_codec);
   bool ChangePlayout(bool playout);
   bool ChangeSend(SendFlags send);
   bool ChangeSend(int channel, SendFlags send);
   bool ConfigureRecvChannel(int channel);
@@ skipping 14 matching lines @@
 
   bool SetChannelRecvRtpHeaderExtensions(
     int channel_id,
     const std::vector<RtpHeaderExtension>& extensions);
   bool SetChannelSendRtpHeaderExtensions(
     int channel_id,
     const std::vector<RtpHeaderExtension>& extensions);
 
   rtc::ThreadChecker thread_checker_;
 
-  WebRtcVoiceEngine* const engine_;
+  WebRtcVoiceEngine* const engine_ = nullptr;
   std::vector<AudioCodec> recv_codecs_;
   std::vector<AudioCodec> send_codecs_;
   rtc::scoped_ptr<webrtc::CodecInst> send_codec_;
-  bool send_bitrate_setting_;
-  int send_bitrate_bps_;
+  bool send_bitrate_setting_ = false;
+  int send_bitrate_bps_ = 0;
   AudioOptions options_;
-  bool dtmf_allowed_;
-  bool desired_playout_;
-  bool nack_enabled_;
-  bool playout_;
-  bool typing_noise_detected_;
-  SendFlags desired_send_;
-  SendFlags send_;
-  webrtc::Call* const call_;
+  bool dtmf_allowed_ = false;
+  bool desired_playout_ = false;
+  bool nack_enabled_ = false;
+  bool playout_ = false;
+  SendFlags desired_send_ = SEND_NOTHING;
+  SendFlags send_ = SEND_NOTHING;
+  webrtc::Call* const call_ = nullptr;
 
   // SSRC of unsignalled receive stream, or -1 if there isn't one.
   int64_t default_recv_ssrc_ = -1;
   // Volume for unsignalled stream, which may be set before the stream exists.
   double default_recv_volume_ = 1.0;
   // SSRC to use for RTCP receiver reports; default to 1 in case of no signaled
   // send streams. See: https://code.google.com/p/webrtc/issues/detail?id=4740
   uint32_t receiver_reports_ssrc_ = 1;
 
   class WebRtcAudioSendStream;
   std::map<uint32_t, WebRtcAudioSendStream*> send_streams_;
   std::vector<RtpHeaderExtension> send_extensions_;
 
   class WebRtcAudioReceiveStream;
   std::map<uint32_t, WebRtcAudioReceiveStream*> receive_channels_;
   std::map<uint32_t, webrtc::AudioReceiveStream*> receive_streams_;
   std::map<uint32_t, StreamParams> receive_stream_params_;
   // receive_channels_ can be read from WebRtc callback thread.  Access from
   // the WebRtc thread must be synchronized with edits on the worker thread.
   // Reads on the worker thread are ok.
   std::vector<RtpHeaderExtension> receive_extensions_;
   std::vector<webrtc::RtpExtension> recv_rtp_extensions_;
 
   RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(WebRtcVoiceMediaChannel);
 };
-
 }  // namespace cricket
 
 #endif  // TALK_MEDIA_WEBRTCVOICEENGINE_H_