Use ChannelProxy for most calls on voe::Channel in Audio[Receive|Send]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.
  */
 
@@ skipping 61 matching lines @@
       rtp_header_parser_(RtpHeaderParser::Create()) {
   LOG(LS_INFO) << "AudioReceiveStream: " << config_.ToString();
   RTC_DCHECK_NE(config_.voe_channel_id, -1);
   RTC_DCHECK(remote_bitrate_estimator_);
   RTC_DCHECK(audio_state_.get());
   RTC_DCHECK(rtp_header_parser_);
 
   VoiceEngineImpl* voe_impl = static_cast<VoiceEngineImpl*>(voice_engine());
   channel_proxy_ = voe_impl->GetChannelProxy(config_.voe_channel_id);
   channel_proxy_->SetLocalSSRC(config.rtp.local_ssrc);
-
-  const int channel_id = config.voe_channel_id;
-  ScopedVoEInterface<VoERTP_RTCP> rtp(voice_engine());
   for (const auto& extension : config.rtp.extensions) {
-    // One-byte-extension local identifiers are in the range 1-14 inclusive.
-    RTC_DCHECK_GE(extension.id, 1);
-    RTC_DCHECK_LE(extension.id, 14);
     if (extension.name == RtpExtension::kAudioLevel) {
-      int error = rtp->SetReceiveAudioLevelIndicationStatus(channel_id, true,
-                                                            extension.id);
-      RTC_DCHECK_EQ(0, error);
+      channel_proxy_->SetReceiveAudioLevelIndicationStatus(true, extension.id);
       bool registered = rtp_header_parser_->RegisterRtpHeaderExtension(
           kRtpExtensionAudioLevel, extension.id);
       RTC_DCHECK(registered);
     } else if (extension.name == RtpExtension::kAbsSendTime) {
-      int error = rtp->SetReceiveAbsoluteSenderTimeStatus(channel_id, true,
-                                                          extension.id);
-      RTC_DCHECK_EQ(0, error);
+      channel_proxy_->SetReceiveAbsoluteSenderTimeStatus(true, extension.id);
       bool registered = rtp_header_parser_->RegisterRtpHeaderExtension(
           kRtpExtensionAbsoluteSendTime, extension.id);
       RTC_DCHECK(registered);
     } else if (extension.name == RtpExtension::kTransportSequenceNumber) {
       // TODO(holmer): Need to do something here or in  DeliverRtp() to actually
       //               handle audio packets with this header extension.
       bool registered = rtp_header_parser_->RegisterRtpHeaderExtension(
           kRtpExtensionTransportSequenceNumber, extension.id);
       RTC_DCHECK(registered);
     } else {
@@ skipping 51 matching lines @@
                                               header, false);
   }
   return true;
 }
 
 webrtc::AudioReceiveStream::Stats AudioReceiveStream::GetStats() const {
   RTC_DCHECK(thread_checker_.CalledOnValidThread());
   webrtc::AudioReceiveStream::Stats stats;
   stats.remote_ssrc = config_.rtp.remote_ssrc;
   ScopedVoEInterface<VoECodec> codec(voice_engine());
-  ScopedVoEInterface<VoENetEqStats> neteq(voice_engine());
-  ScopedVoEInterface<VoERTP_RTCP> rtp(voice_engine());
-  ScopedVoEInterface<VoEVideoSync> sync(voice_engine());
-  ScopedVoEInterface<VoEVolumeControl> volume(voice_engine());
 
-  webrtc::CallStatistics call_stats = {0};
-  int error = rtp->GetRTCPStatistics(config_.voe_channel_id, call_stats);
-  RTC_DCHECK_EQ(0, error);
+  webrtc::CallStatistics call_stats = channel_proxy_->GetRTCPStatistics();
   webrtc::CodecInst codec_inst = {0};
   if (codec->GetRecCodec(config_.voe_channel_id, codec_inst) == -1) {
     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;
   }
   stats.ext_seqnum = call_stats.extendedMax;
   if (codec_inst.plfreq / 1000 > 0) {
     stats.jitter_ms = call_stats.jitterSamples / (codec_inst.plfreq / 1000);
   }
-  {
-    int jitter_buffer_delay_ms = 0;
-    int playout_buffer_delay_ms = 0;
-    sync->GetDelayEstimate(config_.voe_channel_id, &jitter_buffer_delay_ms,
-                           &playout_buffer_delay_ms);
-    stats.delay_estimate_ms = jitter_buffer_delay_ms + playout_buffer_delay_ms;
-  }
-  {
-    unsigned int level = 0;
-    error = volume->GetSpeechOutputLevelFullRange(config_.voe_channel_id,
-                                                  level);
-    RTC_DCHECK_EQ(0, error);
-    stats.audio_level = static_cast<int32_t>(level);
-  }
+  stats.delay_estimate_ms = channel_proxy_->GetDelayEstimate();
+  stats.audio_level = channel_proxy_->GetSpeechOutputLevelFullRange();
 
   // Get jitter buffer and total delay (alg + jitter + playout) stats.
-  webrtc::NetworkStatistics ns = {0};
-  error = neteq->GetNetworkStatistics(config_.voe_channel_id, ns);
-  RTC_DCHECK_EQ(0, error);
+  auto ns = channel_proxy_->GetNetworkStatistics();
   stats.jitter_buffer_ms = ns.currentBufferSize;
   stats.jitter_buffer_preferred_ms = ns.preferredBufferSize;
   stats.expand_rate = Q14ToFloat(ns.currentExpandRate);
   stats.speech_expand_rate = Q14ToFloat(ns.currentSpeechExpandRate);
   stats.secondary_decoded_rate = Q14ToFloat(ns.currentSecondaryDecodedRate);
   stats.accelerate_rate = Q14ToFloat(ns.currentAccelerateRate);
   stats.preemptive_expand_rate = Q14ToFloat(ns.currentPreemptiveRate);
 
-  webrtc::AudioDecodingCallStats ds;
-  error = neteq->GetDecodingCallStatistics(config_.voe_channel_id, &ds);
-  RTC_DCHECK_EQ(0, error);
+  auto ds = channel_proxy_->GetDecodingCallStatistics();
   stats.decoding_calls_to_silence_generator = ds.calls_to_silence_generator;
   stats.decoding_calls_to_neteq = ds.calls_to_neteq;
   stats.decoding_normal = ds.decoded_normal;
   stats.decoding_plc = ds.decoded_plc;
   stats.decoding_cng = ds.decoded_cng;
   stats.decoding_plc_cng = ds.decoded_plc_cng;
 
   return stats;
 }
 
 const webrtc::AudioReceiveStream::Config& AudioReceiveStream::config() const {
   RTC_DCHECK(thread_checker_.CalledOnValidThread());
   return config_;
 }
 
 VoiceEngine* AudioReceiveStream::voice_engine() const {
   internal::AudioState* audio_state =
       static_cast<internal::AudioState*>(audio_state_.get());
   VoiceEngine* voice_engine = audio_state->voice_engine();
   RTC_DCHECK(voice_engine);
   return voice_engine;
 }
 }  // namespace internal
 }  // namespace webrtc