Injectable audio encoders: WebRtcVoiceEngine and company

webrtc/audio/audio_send_stream.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.
  */
 
@@ skipping 12 matching lines @@
 #include "webrtc/modules/congestion_controller/include/congestion_controller.h"
 #include "webrtc/modules/pacing/paced_sender.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "webrtc/voice_engine/channel_proxy.h"
 #include "webrtc/voice_engine/include/voe_base.h"
 #include "webrtc/voice_engine/include/voe_volume_control.h"
 #include "webrtc/voice_engine/voice_engine_impl.h"
 
 namespace webrtc {
 
-namespace {
-
-constexpr char kOpusCodecName[] = "opus";
-
-bool IsCodec(const webrtc::CodecInst& codec, const char* ref_name) {
-  return (STR_CASE_CMP(codec.plname, ref_name) == 0);
-}
-}  // namespace
-
 namespace internal {
 AudioSendStream::AudioSendStream(
     const webrtc::AudioSendStream::Config& config,
     const rtc::scoped_refptr<webrtc::AudioState>& audio_state,
     rtc::TaskQueue* worker_queue,
     PacketRouter* packet_router,
     CongestionController* congestion_controller,
     BitrateAllocator* bitrate_allocator,
     RtcEventLog* event_log,
     RtcpRttStats* rtcp_rtt_stats)
@@ skipping 226 matching lines @@
   channel_proxy_->SetVADStatus(false);
   channel_proxy_->SetCodecFECStatus(false);
 
   // We disable audio network adaptor here. This will on one hand make sure that
   // audio network adaptor is disabled by default, and on the other allow audio
   // network adaptor to be reconfigured, since SetReceiverFrameLengthRange can
   // be only called when audio network adaptor is disabled.
   channel_proxy_->DisableAudioNetworkAdaptor();
 
   const auto& send_codec_spec = config_.send_codec_spec;
-
-  // We set the codec first, since the below extra configuration is only applied
-  // to the "current" codec.
-
-  // If codec is already configured, we do not it again.
-  // TODO(minyue): check if this check is really needed, or can we move it into
-  // |codec->SetSendCodec|.
-  webrtc::CodecInst current_codec = {0};
-  if (!channel_proxy_->GetSendCodec(&current_codec) ||
-      (send_codec_spec.codec_inst != current_codec)) {
-    if (!channel_proxy_->SetSendCodec(send_codec_spec.codec_inst)) {
-      LOG(LS_WARNING) << "SetSendCodec() failed.";
-      return false;
-    }
+  channel_proxy_->SetSendFormat(send_codec_spec.payload_type,
+                                send_codec_spec.format.format,
+                                config_.encoder_factory.get());
+  // TODO(ossu): Formalize bandwidth parameters and send along to encoder

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

This is one of the open questions: cricket::AudioCodec has bitrate, which is
currently propagated through CodecInst and into the encoder. From what I can
find, it's never set to anything but 0, except when being filled in with values
from ACMCodecDB.

If there's no way to signal a fixed bitrate - and I'm not sure there should be -
then the application developer will have much more control over bitrates by
customizing an AudioEncoderFactory over anything that could be changed at higher
levels of the API.

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

It'd be good if we could do that. But that's only accessible from C++---I don't
know if that's a problem.

+  // constructor.
+  if (send_codec_spec.target_bitrate_bps) {
+    channel_proxy_->SetBitrate(*send_codec_spec.target_bitrate_bps, 0);
   }
-
-  // Codec internal FEC. Treat any failure as fatal internal error.
-  if (send_codec_spec.enable_codec_fec) {
-    if (!channel_proxy_->SetCodecFECStatus(true)) {
-      LOG(LS_WARNING) << "SetCodecFECStatus() failed.";
-      return false;
-    }
-  }
-
-  // DTX and maxplaybackrate are only set if current codec is Opus.
-  if (IsCodec(send_codec_spec.codec_inst, kOpusCodecName)) {
-    if (!channel_proxy_->SetOpusDtx(send_codec_spec.enable_opus_dtx)) {
-      LOG(LS_WARNING) << "SetOpusDtx() failed.";
-      return false;
-    }
-
-    // If opus_max_playback_rate <= 0, the default maximum playback rate
-    // (48 kHz) will be used.
-    if (send_codec_spec.opus_max_playback_rate > 0) {
-      if (!channel_proxy_->SetOpusMaxPlaybackRate(
-              send_codec_spec.opus_max_playback_rate)) {
-        LOG(LS_WARNING) << "SetOpusMaxPlaybackRate() failed.";
-        return false;
-      }
-    }
-
-    if (config_.audio_network_adaptor_config) {
-      // Audio network adaptor is only allowed for Opus currently.
-      // |SetReceiverFrameLengthRange| needs to be called before
-      // |EnableAudioNetworkAdaptor|.
-      channel_proxy_->SetReceiverFrameLengthRange(send_codec_spec.min_ptime_ms,
-                                                  send_codec_spec.max_ptime_ms);
-      channel_proxy_->EnableAudioNetworkAdaptor(
-          *config_.audio_network_adaptor_config);
-      LOG(LS_INFO) << "Audio network adaptor enabled on SSRC "
-                   << config_.rtp.ssrc;
-    }
+  if (config_.audio_network_adaptor_config) {
+    // Audio network adaptor is only allowed for Opus currently.
+    channel_proxy_->EnableAudioNetworkAdaptor(
+        *config_.audio_network_adaptor_config);
+    LOG(LS_INFO) << "Audio network adaptor enabled on SSRC "
+                 << config_.rtp.ssrc;
   }
 
   // Set the CN payloadtype and the VAD status.
   if (send_codec_spec.cng_payload_type != -1) {
     // The CN payload type for 8000 Hz clockrate is fixed at 13.
     if (send_codec_spec.cng_plfreq != 8000) {
       webrtc::PayloadFrequencies cn_freq;
       switch (send_codec_spec.cng_plfreq) {
         case 16000:
           cn_freq = webrtc::kFreq16000Hz;
@@ skipping 14 matching lines @@
         // Not returning false because the SetSendCNPayloadType will fail if
         // the channel is already sending.
         // This can happen if the remote description is applied twice, for
         // example in the case of ROAP on top of JSEP, where both side will
         // send the offer.
       }
     }
 
     // Only turn on VAD if we have a CN payload type that matches the
     // clockrate for the codec we are going to use.
-    if (send_codec_spec.cng_plfreq == send_codec_spec.codec_inst.plfreq &&
-        send_codec_spec.codec_inst.channels == 1) {
+    if (send_codec_spec.cng_plfreq ==
+            send_codec_spec.format.format.clockrate_hz &&
+        send_codec_spec.format.info.num_channels == 1) {
       // TODO(minyue): If CN frequency == 48000 Hz is allowed, consider the
       // interaction between VAD and Opus FEC.
       if (!channel_proxy_->SetVADStatus(true)) {
         LOG(LS_WARNING) << "SetVADStatus() failed.";
         return false;
       }
     }
   }
   return true;
 }
 
 }  // namespace internal
 }  // namespace webrtc