Remove remaining calls to VoE APIs from Audio[Send|Receive]Stream.

webrtc/audio/audio_receive_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.
  */
 
 #include "webrtc/audio/audio_receive_stream.h"
 
 #include <string>
 #include <utility>
 
 #include "webrtc/api/call/audio_sink.h"
 #include "webrtc/audio/audio_send_stream.h"
 #include "webrtc/audio/audio_state.h"
 #include "webrtc/audio/conversion.h"
 #include "webrtc/base/checks.h"
 #include "webrtc/base/logging.h"
 #include "webrtc/base/timeutils.h"
 #include "webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_receiver.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
 #include "webrtc/voice_engine/channel_proxy.h"
 #include "webrtc/voice_engine/include/voe_base.h"
-#include "webrtc/voice_engine/include/voe_codec.h"
-#include "webrtc/voice_engine/include/voe_neteq_stats.h"
-#include "webrtc/voice_engine/include/voe_rtp_rtcp.h"
-#include "webrtc/voice_engine/include/voe_video_sync.h"
-#include "webrtc/voice_engine/include/voe_volume_control.h"
 #include "webrtc/voice_engine/voice_engine_impl.h"
 
 namespace webrtc {
 
 std::string AudioReceiveStream::Config::Rtp::ToString() const {
   std::stringstream ss;
   ss << "{remote_ssrc: " << remote_ssrc;
   ss << ", local_ssrc: " << local_ssrc;
   ss << ", transport_cc: " << (transport_cc ? "on" : "off");
   ss << ", nack: " << nack.ToString();
@@ skipping 127 matching lines @@
   playing_ = false;
 
   audio_state()->mixer()->RemoveSource(this);
   SetVoiceEnginePlayout(false);
 }
 
 webrtc::AudioReceiveStream::Stats AudioReceiveStream::GetStats() const {
   RTC_DCHECK_RUN_ON(&worker_thread_checker_);
   webrtc::AudioReceiveStream::Stats stats;
   stats.remote_ssrc = config_.rtp.remote_ssrc;
-  ScopedVoEInterface<VoECodec> codec(voice_engine());
 
   webrtc::CallStatistics call_stats = channel_proxy_->GetRTCPStatistics();
+  // TODO(solenberg): Don't return here if we can't get the codec - return the
+  //                  stats we *can* get.
   webrtc::CodecInst codec_inst = {0};
-  if (codec->GetRecCodec(config_.voe_channel_id, codec_inst) == -1) {
+  if (!channel_proxy_->GetRecCodec(&codec_inst)) {
     return stats;
   }
 
   stats.bytes_rcvd = call_stats.bytesReceived;
   stats.packets_rcvd = call_stats.packetsReceived;
   stats.packets_lost = call_stats.cumulativeLost;
   stats.fraction_lost = Q8ToFloat(call_stats.fractionLost);
   stats.capture_start_ntp_time_ms = call_stats.capture_start_ntp_time_ms_;
   if (codec_inst.pltype != -1) {
     stats.codec_name = codec_inst.plname;
@@ skipping 171 matching lines @@
 int AudioReceiveStream::SetVoiceEnginePlayout(bool playout) {
   ScopedVoEInterface<VoEBase> base(voice_engine());
   if (playout) {
     return base->StartPlayout(config_.voe_channel_id);
   } else {
     return base->StopPlayout(config_.voe_channel_id);
   }
 }
 }  // namespace internal
 }  // namespace webrtc