Injectable audio encoders: WebRtcVoiceEngine and company

webrtc/voice_engine/channel.h

 /*
  *  Copyright (c) 2012 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_H_
 #define WEBRTC_VOICE_ENGINE_CHANNEL_H_
 
 #include <memory>
 
 #include "webrtc/api/audio/audio_mixer.h"
 #include "webrtc/api/call/audio_sink.h"
 #include "webrtc/base/criticalsection.h"
 #include "webrtc/base/optional.h"
 #include "webrtc/common_audio/resampler/include/push_resampler.h"
 #include "webrtc/common_types.h"
 #include "webrtc/modules/audio_coding/acm2/codec_manager.h"
 #include "webrtc/modules/audio_coding/acm2/rent_a_codec.h"
+#include "webrtc/modules/audio_coding/codecs/audio_encoder_factory.h"
 #include "webrtc/modules/audio_coding/include/audio_coding_module.h"
 #include "webrtc/modules/audio_conference_mixer/include/audio_conference_mixer_defines.h"
 #include "webrtc/modules/audio_processing/rms_level.h"
 #include "webrtc/modules/rtp_rtcp/include/remote_ntp_time_estimator.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
 #include "webrtc/voice_engine/file_player.h"
 #include "webrtc/voice_engine/file_recorder.h"
 #include "webrtc/voice_engine/include/voe_audio_processing.h"
 #include "webrtc/voice_engine/include/voe_base.h"
@@ skipping 128 matching lines @@
   int32_t UpdateLocalTimeStamp();
 
   void SetSink(std::unique_ptr<AudioSinkInterface> sink);
 
   // TODO(ossu): Don't use! It's only here to confirm that the decoder factory
   // passed into AudioReceiveStream is the same as the one set when creating the
   // ADM. Once Channel creation is moved into Audio{Send,Receive}Stream this can
   // go.
   const rtc::scoped_refptr<AudioDecoderFactory>& GetAudioDecoderFactory() const;
 
+  AudioCodingModule& GetAudioCodingModule();
+
   // API methods
 
   // VoEBase
   int32_t StartPlayout();
   int32_t StopPlayout();
   int32_t StartSend();
   int32_t StopSend();
   int32_t RegisterVoiceEngineObserver(VoiceEngineObserver& observer);
   int32_t DeRegisterVoiceEngineObserver();
 
   // VoECodec
   int32_t GetSendCodec(CodecInst& codec);
   int32_t GetRecCodec(CodecInst& codec);
   int32_t SetSendCodec(const CodecInst& codec);
+  bool SetSendFormat(int payload_type,
+                     const SdpAudioFormat& format,
+                     AudioEncoderFactory* factory);  // Not part of VoECodec!

[kwiberg-webrtc, 2017/02/21 23:35:04]

Then maybe don't put it under the "// VoECodec" header?

   void SetBitRate(int bitrate_bps, int64_t probing_interval_ms);
   int32_t SetVADStatus(bool enableVAD, ACMVADMode mode, bool disableDTX);
   int32_t GetVADStatus(bool& enabledVAD, ACMVADMode& mode, bool& disabledDTX);
   int32_t SetRecPayloadType(const CodecInst& codec);
   int32_t SetRecPayloadType(int payload_type, const SdpAudioFormat& format);
   int32_t GetRecPayloadType(CodecInst& codec);
   int32_t SetSendCNPayloadType(int type, PayloadFrequencies frequency);
   int SetOpusMaxPlaybackRate(int frequency_hz);
   int SetOpusDtx(bool enable_dtx);
   int GetOpusDtx(bool* enabled);
@@ skipping 317 matching lines @@
   std::unique_ptr<RateLimiter> retransmission_rate_limiter_;
 
   // TODO(ossu): Remove once GetAudioDecoderFactory() is no longer needed.
   rtc::scoped_refptr<AudioDecoderFactory> decoder_factory_;
 };
 
 }  // namespace voe
 }  // namespace webrtc
 
 #endif  // WEBRTC_VOICE_ENGINE_CHANNEL_H_