Injectable audio encoders: voice_engine/channel changes.

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);

[the sun, 2017/02/22 14:08:38]

Why are we sending the factory all the way down here? Isn't it possible to
instantiate the encoder in WVoMC and pass it in AudioSendStream::Config?

[ossu, 2017/02/22 14:24:41]

Ah, yes: since we need the information from SdpAudioFormat to register the
payload with the RtpRtcpModule, we'd need to send along the SdpAudioFormat
anyway. To me, it seemed strange to send the encoder and the format together,
since they could theoretically be out-of-sync. This way, the format and audio
encoder will be as in-sync as the factory allows. We're at least not handing any
ownership over to Channel here.

I'm willing to entertain other variants, such as making it possible to query the
SdpAudioFormat from an AudioEncoder.

I believe this could be made clearer after an overhaul to the audio parts of the
RtpRtcpModule. I'm planning that as a follow-up to this (if only to remove its
use of CodecInst).

+
   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 68 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_