Injectable audio encoders: WebRtcVoiceEngine and company

webrtc/call/audio_send_stream.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_CALL_AUDIO_SEND_STREAM_H_
 #define WEBRTC_CALL_AUDIO_SEND_STREAM_H_
 
 #include <memory>
 #include <string>
 #include <vector>
 
+#include "webrtc/api/audio_codecs/audio_format.h"
 #include "webrtc/api/call/transport.h"
 #include "webrtc/base/optional.h"
 #include "webrtc/config.h"
-#include "webrtc/modules/audio_coding/codecs/audio_encoder.h"
+#include "webrtc/modules/audio_coding/codecs/audio_encoder_factory.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
 
 // WORK IN PROGRESS
 // This class is under development and is not yet intended for for use outside
 // of WebRtc/Libjingle. Please use the VoiceEngine API instead.
 // See: https://bugs.chromium.org/p/webrtc/issues/detail?id=4690
 
 class AudioSendStream {
@@ skipping 64 matching lines @@
     // Note: This is still an experimental feature and not ready for real usage.
     int min_bitrate_bps = -1;
     int max_bitrate_bps = -1;
 
     // Defines whether to turn on audio network adaptor, and defines its config
     // string.
     rtc::Optional<std::string> audio_network_adaptor_config;
 
     struct SendCodecSpec {
       SendCodecSpec();
+      ~SendCodecSpec();
       std::string ToString() const;
 
       bool operator==(const SendCodecSpec& rhs) const;
       bool operator!=(const SendCodecSpec& rhs) const {
         return !(*this == rhs);
       }
 
       bool nack_enabled = false;
       bool transport_cc_enabled = false;
-      bool enable_codec_fec = false;
-      bool enable_opus_dtx = false;
-      int opus_max_playback_rate = 0;
       int cng_payload_type = -1;
       int cng_plfreq = -1;
-      int max_ptime_ms = -1;
-      int min_ptime_ms = -1;
-      webrtc::CodecInst codec_inst;
+      int payload_type;
+      rtc::Optional<int> target_bitrate_bps;
+      webrtc::AudioCodecSpec format;

[ossu, 2017/02/21 11:04:14]

This isn't very nice. It leads to us having to refer to format.format and
format.info. I believe I'd prefer it if instead the AudioFormatInfo was
generated by the stream calling encoder_factory->QueryAudioFormat(), but that
call fits so well in WebRtcVoiceEngine, since it allows us to check if the
format is supported as well, and IIRC, we want the information to be able to set
target_bitrate_bps. If that one goes (see previous discussion about bitrate)
then cleaning this one up should also be easier.

     } send_codec_spec;
+
+    rtc::scoped_refptr<AudioEncoderFactory> encoder_factory;
   };
 
   // Starts stream activity.
   // When a stream is active, it can receive, process and deliver packets.
   virtual void Start() = 0;
   // Stops stream activity.
   // When a stream is stopped, it can't receive, process or deliver packets.
   virtual void Stop() = 0;
 
   // TODO(solenberg): Make payload_type a config property instead.
   virtual bool SendTelephoneEvent(int payload_type, int payload_frequency,
                                   int event, int duration_ms) = 0;
 
   virtual void SetMuted(bool muted) = 0;
 
   virtual Stats GetStats() const = 0;
 
  protected:
   virtual ~AudioSendStream() {}
 };
 }  // namespace webrtc
 
 #endif  // WEBRTC_CALL_AUDIO_SEND_STREAM_H_