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 63 matching lines @@
     // disable audio bitrate adaptation.
     // 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(int payload_type, const SdpAudioFormat& format);
+      ~SendCodecSpec();
       std::string ToString() const;
 
       bool operator==(const SendCodecSpec& rhs) const;
       bool operator!=(const SendCodecSpec& rhs) const {
         return !(*this == rhs);
       }
 
+      int payload_type;

[the sun, 2017/04/24 14:48:49]

nit: always init. Someone will change the codec later on.

[ossu, 2017/04/24 16:20:30]

The SendCodecSpec constructor explicitly asks for payload_type and format, so
they will be initialized.

+      SdpAudioFormat format;
       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;
-    } send_codec_spec;
+      rtc::Optional<int> cng_payload_type;
+      // If unset, use the encoder's default target bitrate.
+      rtc::Optional<int> target_bitrate_bps;
+    };
+
+    rtc::Optional<SendCodecSpec> send_codec_spec;
+    rtc::scoped_refptr<AudioEncoderFactory> encoder_factory;
   };
 
+  // Reconfigure the stream according to the Configuration.
+  virtual void Reconfigure(const Config& config) = 0;
+
   // 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_