Injectable audio encoders: WebRtcVoiceEngine and company

webrtc/voice_engine/channel_proxy.h

 /*
  *  Copyright (c) 2015 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_VOICE_ENGINE_CHANNEL_PROXY_H_
 #define WEBRTC_VOICE_ENGINE_CHANNEL_PROXY_H_
 
 #include "webrtc/api/audio/audio_mixer.h"
 #include "webrtc/base/constructormagic.h"
 #include "webrtc/base/race_checker.h"
 #include "webrtc/base/thread_checker.h"
+#include "webrtc/modules/audio_coding/codecs/audio_encoder_factory.h"
 #include "webrtc/voice_engine/channel_manager.h"
 #include "webrtc/voice_engine/include/voe_rtp_rtcp.h"
 
 #include <memory>
 #include <string>
 #include <vector>
 
 namespace webrtc {
 
 class AudioSinkInterface;
@@ skipping 17 matching lines @@
 //  1. Allow mocking just the interfaces used, instead of the entire
 //     voe::Channel class.
 //  2. Provide a refined interface for the stream classes, including assumptions
 //     on return values and input adaptation.
 class ChannelProxy {
  public:
   ChannelProxy();
   explicit ChannelProxy(const ChannelOwner& channel_owner);
   virtual ~ChannelProxy();
 
+  virtual bool SetSendFormat(int payload_type,
+                             const SdpAudioFormat& format,
+                             AudioEncoderFactory* factory);
+
   virtual void SetRTCPStatus(bool enable);
   virtual void SetLocalSSRC(uint32_t ssrc);
   virtual void SetRTCP_CNAME(const std::string& c_name);
   virtual void SetNACKStatus(bool enable, int max_packets);
   virtual void SetSendAudioLevelIndicationStatus(bool enable, int id);
   virtual void SetReceiveAudioLevelIndicationStatus(bool enable, int id);
   virtual void EnableSendTransportSequenceNumber(int id);
   virtual void EnableReceiveTransportSequenceNumber(int id);
   virtual void RegisterSenderCongestionControlObjects(
       RtpPacketSender* rtp_packet_sender,
@@ skipping 18 matching lines @@
   virtual void SetSink(std::unique_ptr<AudioSinkInterface> sink);
   virtual void SetInputMute(bool muted);
   virtual void RegisterExternalTransport(Transport* transport);
   virtual void DeRegisterExternalTransport();
   virtual bool ReceivedRTPPacket(const uint8_t* packet,
                                  size_t length,
                                  const PacketTime& packet_time);
   virtual bool ReceivedRTCPPacket(const uint8_t* packet, size_t length);
   virtual const rtc::scoped_refptr<AudioDecoderFactory>&
       GetAudioDecoderFactory() const;
+  virtual AudioCodingModule& GetAudioCodingModule();
   virtual void SetChannelOutputVolumeScaling(float scaling);
   virtual void SetRtcEventLog(RtcEventLog* event_log);
   virtual void EnableAudioNetworkAdaptor(const std::string& config_string);
   virtual void DisableAudioNetworkAdaptor();
   virtual void SetReceiverFrameLengthRange(int min_frame_length_ms,
                                            int max_frame_length_ms);
   virtual AudioMixer::Source::AudioFrameInfo GetAudioFrameWithInfo(
       int sample_rate_hz,
       AudioFrame* audio_frame);
   virtual int NeededFrequency() const;
@@ skipping 30 matching lines @@
   rtc::RaceChecker audio_thread_race_checker_;
   rtc::RaceChecker video_capture_thread_race_checker_;
   ChannelOwner channel_owner_;
 
   RTC_DISALLOW_COPY_AND_ASSIGN(ChannelProxy);
 };
 }  // namespace voe
 }  // namespace webrtc
 
 #endif  // WEBRTC_VOICE_ENGINE_CHANNEL_PROXY_H_