Adding OnReceivedOverhead to AudioEncoder.

webrtc/modules/audio_coding/codecs/audio_encoder.cc

 /*
  *  Copyright (c) 2014 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/modules/audio_coding/codecs/audio_encoder.h"
 
 #include "webrtc/base/checks.h"
 #include "webrtc/base/trace_event.h"
+#include "webrtc/system_wrappers/include/field_trial.h"
 
 namespace webrtc {
 
 AudioEncoder::EncodedInfo::EncodedInfo() = default;
 AudioEncoder::EncodedInfo::EncodedInfo(const EncodedInfo&) = default;
 AudioEncoder::EncodedInfo::EncodedInfo(EncodedInfo&&) = default;
 AudioEncoder::EncodedInfo::~EncodedInfo() = default;
 AudioEncoder::EncodedInfo& AudioEncoder::EncodedInfo::operator=(
     const EncodedInfo&) = default;
 AudioEncoder::EncodedInfo& AudioEncoder::EncodedInfo::operator=(EncodedInfo&&) =
     default;
 
+AudioEncoder::AudioEncoder() = default;
+AudioEncoder::~AudioEncoder() = default;
+
 int AudioEncoder::RtpTimestampRateHz() const {
   return SampleRateHz();
 }
 
 AudioEncoder::EncodedInfo AudioEncoder::Encode(
     uint32_t rtp_timestamp,
     rtc::ArrayView<const int16_t> audio,
     rtc::Buffer* encoded) {
   TRACE_EVENT0("webrtc", "AudioEncoder::Encode");
   RTC_CHECK_EQ(audio.size(),
@@ skipping 38 matching lines @@
 void AudioEncoder::DisableAudioNetworkAdaptor() {}
 
 void AudioEncoder::OnReceivedUplinkBandwidth(int uplink_bandwidth_bps) {}
 
 void AudioEncoder::OnReceivedUplinkPacketLossFraction(
     float uplink_packet_loss_fraction) {
   SetProjectedPacketLossRate(uplink_packet_loss_fraction);
 }
 
 void AudioEncoder::OnReceivedTargetAudioBitrate(int target_audio_bitrate_bps) {
-  SetTargetBitrate(target_audio_bitrate_bps);
+  if (webrtc::field_trial::FindFullName("WebRTC-SendSideBwe-WithOverhead") ==
+      "Enabled") {
+    if (!overhead_bytes_per_packet_) {
+      // Ignore the target bitrate is overhead is unknown.
+      return;
+    }
+    int overhead_rate =
+        8 * 1000 * *overhead_bytes_per_packet_ / Num10MsFramesInNextPacket();

[michaelt, 2016/11/24 13:55:18]

I think this calculation is wrong by a a factor of 10 :) .
A unittest for this would be nice.

+    SetTargetBitrate(std::max(0, target_audio_bitrate_bps - overhead_rate));
+  } else {
+    SetTargetBitrate(target_audio_bitrate_bps);
+  }
 }
 
 void AudioEncoder::OnReceivedRtt(int rtt_ms) {}
 
+void AudioEncoder::OnReceivedOverhead(size_t overhead_bytes_per_packet) {
+  overhead_bytes_per_packet_ = rtc::Optional<size_t>(overhead_bytes_per_packet);
+}
+
 void AudioEncoder::SetReceiverFrameLengthRange(int min_frame_length_ms,
                                                int max_frame_length_ms) {}
 
 }  // namespace webrtc