Use ChannelProxy for most calls on voe::Channel in Audio[Receive|Send]Stream.

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 48 matching lines @@
     : config_(config), audio_state_(audio_state) {
   LOG(LS_INFO) << "AudioSendStream: " << config_.ToString();
   RTC_DCHECK_NE(config_.voe_channel_id, -1);
   RTC_DCHECK(audio_state_.get());
 
   VoiceEngineImpl* voe_impl = static_cast<VoiceEngineImpl*>(voice_engine());
   channel_proxy_ = voe_impl->GetChannelProxy(config_.voe_channel_id);
   channel_proxy_->SetRTCPStatus(true);
   channel_proxy_->SetLocalSSRC(config.rtp.ssrc);
   channel_proxy_->SetRTCP_CNAME(config.rtp.c_name);
-
-  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::kAbsSendTime) {
-      int error = rtp->SetSendAbsoluteSenderTimeStatus(channel_id, true,
-                                                       extension.id);
-      RTC_DCHECK_EQ(0, error);
+      channel_proxy_->SetSendAbsoluteSenderTimeStatus(true, extension.id);
     } else if (extension.name == RtpExtension::kAudioLevel) {
-      int error = rtp->SetSendAudioLevelIndicationStatus(channel_id, true,
-                                                         extension.id);
-      RTC_DCHECK_EQ(0, error);
+      channel_proxy_->SetSendAudioLevelIndicationStatus(true, extension.id);
     } else {
       RTC_NOTREACHED() << "Registering unsupported RTP extension.";
     }
   }
 }
 
 AudioSendStream::~AudioSendStream() {
   RTC_DCHECK(thread_checker_.CalledOnValidThread());
   LOG(LS_INFO) << "~AudioSendStream: " << config_.ToString();
 }
@@ skipping 17 matching lines @@
   // RTC_DCHECK(!thread_checker_.CalledOnValidThread());
   return false;
 }
 
 webrtc::AudioSendStream::Stats AudioSendStream::GetStats() const {
   RTC_DCHECK(thread_checker_.CalledOnValidThread());
   webrtc::AudioSendStream::Stats stats;
   stats.local_ssrc = config_.rtp.ssrc;
   ScopedVoEInterface<VoEAudioProcessing> processing(voice_engine());
   ScopedVoEInterface<VoECodec> codec(voice_engine());
-  ScopedVoEInterface<VoERTP_RTCP> rtp(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();
   stats.bytes_sent = call_stats.bytesSent;
   stats.packets_sent = call_stats.packetsSent;
   // RTT isn't known until a RTCP report is received. Until then, VoiceEngine
   // returns 0 to indicate an error value.
   if (call_stats.rttMs > 0) {
     stats.rtt_ms = call_stats.rttMs;
   }
   // TODO(solenberg): [was ajm]: Re-enable this metric once we have a reliable
   //                  implementation.
   stats.aec_quality_min = -1;
 
   webrtc::CodecInst codec_inst = {0};
   if (codec->GetSendCodec(config_.voe_channel_id, codec_inst) != -1) {
     RTC_DCHECK_NE(codec_inst.pltype, -1);
     stats.codec_name = codec_inst.plname;
 
     // Get data from the last remote RTCP report.
-    std::vector<webrtc::ReportBlock> blocks;
-    error = rtp->GetRemoteRTCPReportBlocks(config_.voe_channel_id, &blocks);
-    RTC_DCHECK_EQ(0, error);
-    for (const webrtc::ReportBlock& block : blocks) {
+    for (const auto& block : channel_proxy_->GetRemoteRTCPReportBlocks()) {
       // Lookup report for send ssrc only.
       if (block.source_SSRC == stats.local_ssrc) {
         stats.packets_lost = block.cumulative_num_packets_lost;
         stats.fraction_lost = Q8ToFloat(block.fraction_lost);
         stats.ext_seqnum = block.extended_highest_sequence_number;
         // Convert samples to milliseconds.
         if (codec_inst.plfreq / 1000 > 0) {
           stats.jitter_ms =
               block.interarrival_jitter / (codec_inst.plfreq / 1000);
         }
         break;
       }
     }
   }
 
   // Local speech level.
   {
     unsigned int level = 0;
-    error = volume->GetSpeechInputLevelFullRange(level);
+    int error = volume->GetSpeechInputLevelFullRange(level);
     RTC_DCHECK_EQ(0, error);
     stats.audio_level = static_cast<int32_t>(level);
   }
 
   bool echo_metrics_on = false;
-  error = processing->GetEcMetricsStatus(echo_metrics_on);
+  int error = processing->GetEcMetricsStatus(echo_metrics_on);
   RTC_DCHECK_EQ(0, error);
   if (echo_metrics_on) {
     // These can also be negative, but in practice -1 is only used to signal
     // insufficient data, since the resolution is limited to multiples of 4 ms.
     int median = -1;
     int std = -1;
     float dummy = 0.0f;
     error = processing->GetEcDelayMetrics(median, std, dummy);
     RTC_DCHECK_EQ(0, error);
     stats.echo_delay_median_ms = median;
@@ skipping 25 matching lines @@
 
 VoiceEngine* AudioSendStream::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