Remove webrtc/stream.h and unutilized inheritance.

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 125 matching lines @@
 }
 
 void AudioReceiveStream::Start() {
   RTC_DCHECK(thread_checker_.CalledOnValidThread());
 }
 
 void AudioReceiveStream::Stop() {
   RTC_DCHECK(thread_checker_.CalledOnValidThread());
 }
 
-void AudioReceiveStream::SignalNetworkState(NetworkState state) {
-  RTC_DCHECK(thread_checker_.CalledOnValidThread());
-}
-
-bool AudioReceiveStream::DeliverRtcp(const uint8_t* packet, size_t length) {
-  // TODO(solenberg): Tests call this function on a network thread, libjingle
-  // calls on the worker thread. We should move towards always using a network
-  // thread. Then this check can be enabled.
-  // RTC_DCHECK(!thread_checker_.CalledOnValidThread());
-  return channel_proxy_->ReceivedRTCPPacket(packet, length);
-}
-
-bool AudioReceiveStream::DeliverRtp(const uint8_t* packet,
-                                    size_t length,
-                                    const PacketTime& packet_time) {
-  // TODO(solenberg): Tests call this function on a network thread, libjingle
-  // calls on the worker thread. We should move towards always using a network
-  // thread. Then this check can be enabled.
-  // RTC_DCHECK(!thread_checker_.CalledOnValidThread());
-  RTPHeader header;
-  if (!rtp_header_parser_->Parse(packet, length, &header)) {
-    return false;
-  }
-
-  // Only forward if the parsed header has one of the headers necessary for
-  // bandwidth estimation. RTP timestamps has different rates for audio and
-  // video and shouldn't be mixed.
-  if (remote_bitrate_estimator_ &&
-      header.extension.hasTransportSequenceNumber) {
-    int64_t arrival_time_ms = TickTime::MillisecondTimestamp();
-    if (packet_time.timestamp >= 0)
-      arrival_time_ms = (packet_time.timestamp + 500) / 1000;
-    size_t payload_size = length - header.headerLength;
-    remote_bitrate_estimator_->IncomingPacket(arrival_time_ms, payload_size,
-                                              header, false);
-  }
-
-  return channel_proxy_->ReceivedRTPPacket(packet, length, packet_time);
-}
-
 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());
 
   webrtc::CallStatistics call_stats = channel_proxy_->GetRTCPStatistics();
   webrtc::CodecInst codec_inst = {0};
   if (codec->GetRecCodec(config_.voe_channel_id, codec_inst) == -1) {
     return stats;
@@ skipping 38 matching lines @@
 void AudioReceiveStream::SetSink(std::unique_ptr<AudioSinkInterface> sink) {
   RTC_DCHECK(thread_checker_.CalledOnValidThread());
   channel_proxy_->SetSink(std::move(sink));
 }
 
 const webrtc::AudioReceiveStream::Config& AudioReceiveStream::config() const {
   RTC_DCHECK(thread_checker_.CalledOnValidThread());
   return config_;
 }
 
+void AudioReceiveStream::SignalNetworkState(NetworkState state) {
+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
+}
+
+bool AudioReceiveStream::DeliverRtcp(const uint8_t* packet, size_t length) {
+  // TODO(solenberg): Tests call this function on a network thread, libjingle
+  // calls on the worker thread. We should move towards always using a network
+  // thread. Then this check can be enabled.
+  // RTC_DCHECK(!thread_checker_.CalledOnValidThread());
+  return channel_proxy_->ReceivedRTCPPacket(packet, length);
+}
+
+bool AudioReceiveStream::DeliverRtp(const uint8_t* packet,
+                                    size_t length,
+                                    const PacketTime& packet_time) {
+  // TODO(solenberg): Tests call this function on a network thread, libjingle
+  // calls on the worker thread. We should move towards always using a network
+  // thread. Then this check can be enabled.
+  // RTC_DCHECK(!thread_checker_.CalledOnValidThread());
+  RTPHeader header;
+  if (!rtp_header_parser_->Parse(packet, length, &header)) {
+    return false;
+  }
+
+  // Only forward if the parsed header has one of the headers necessary for
+  // bandwidth estimation. RTP timestamps has different rates for audio and
+  // video and shouldn't be mixed.
+  if (remote_bitrate_estimator_ &&
+      header.extension.hasTransportSequenceNumber) {
+    int64_t arrival_time_ms = TickTime::MillisecondTimestamp();
+    if (packet_time.timestamp >= 0)
+      arrival_time_ms = (packet_time.timestamp + 500) / 1000;
+    size_t payload_size = length - header.headerLength;
+    remote_bitrate_estimator_->IncomingPacket(arrival_time_ms, payload_size,
+                                              header, false);
+  }
+
+  return channel_proxy_->ReceivedRTPPacket(packet, length, packet_time);
+}
+
 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