Injectable audio encoders: WebRtcVoiceEngine and company

webrtc/video/video_quality_test.cc

 /*
  *  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.
  */
 #include "webrtc/video/video_quality_test.h"
@@ skipping 1413 matching lines @@
   // Add extension to enable audio send side BWE, and allow audio bit rate
   // adaptation.
   audio_send_config_.rtp.extensions.clear();
   if (params_.call.send_side_bwe) {
     audio_send_config_.rtp.extensions.push_back(
         webrtc::RtpExtension(webrtc::RtpExtension::kTransportSequenceNumberUri,
                              test::kTransportSequenceNumberExtensionId));
     audio_send_config_.min_bitrate_bps = kOpusMinBitrateBps;
     audio_send_config_.max_bitrate_bps = kOpusBitrateFbBps;
   }
-  audio_send_config_.send_codec_spec.codec_inst =
-      CodecInst{120, "OPUS", 48000, 960, 2, 64000};
+  audio_send_config_.send_codec_spec.payload_type = 120;
+  // TODO(ossu): "Add stereo here?"

[kwiberg-webrtc, 2017/02/21 23:35:03]

Presumably yes---the "2" in the CodecInst in the old code meant stereo, and
that's also consistent with the 64k bitrate. Hard to know if it was truly
intentional, though.

+  audio_send_config_.send_codec_spec.format.format = {"OPUS", 48000, 2};
 
   audio_send_stream_ = call->CreateAudioSendStream(audio_send_config_);
 
   AudioReceiveStream::Config audio_config;
   audio_config.rtp.local_ssrc = kReceiverLocalAudioSsrc;
   audio_config.rtcp_send_transport = transport;
   audio_config.voe_channel_id = receive_channel_id;
   audio_config.rtp.remote_ssrc = audio_send_config_.rtp.ssrc;
   audio_config.rtp.transport_cc = params_.call.send_side_bwe;
   audio_config.rtp.extensions = audio_send_config_.rtp.extensions;
@@ skipping 158 matching lines @@
     std::ostringstream str;
     str << receive_logs_++;
     std::string path =
         params_.video.encoded_frame_base_path + "." + str.str() + ".recv.ivf";
     stream->EnableEncodedFrameRecording(rtc::CreatePlatformFile(path),
                                         10000000);
   }
 }
 
 }  // namespace webrtc