Adding OnReceivedOverhead to AudioEncoder.

webrtc/modules/audio_coding/codecs/audio_encoder.h

 /*
  *  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.
  */
 
 #ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_AUDIO_ENCODER_H_
 #define WEBRTC_MODULES_AUDIO_CODING_CODECS_AUDIO_ENCODER_H_
 
 #include <algorithm>
 #include <vector>
 
 #include "webrtc/base/array_view.h"
 #include "webrtc/base/buffer.h"
 #include "webrtc/base/deprecation.h"
+#include "webrtc/base/optional.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
 
 class Clock;
 
 // This is the interface class for encoders in AudioCoding module. Each codec
 // type must have an implementation of this class.
 class AudioEncoder {
  public:
@@ skipping 39 matching lines @@
     EncodedInfo();
     EncodedInfo(const EncodedInfo&);
     EncodedInfo(EncodedInfo&&);
     ~EncodedInfo();
     EncodedInfo& operator=(const EncodedInfo&);
     EncodedInfo& operator=(EncodedInfo&&);
 
     std::vector<EncodedInfoLeaf> redundant;
   };
 
-  virtual ~AudioEncoder() = default;
+  AudioEncoder();
+  virtual ~AudioEncoder();
 
   // Returns the input sample rate in Hz and the number of input channels.
   // These are constants set at instantiation time.
   virtual int SampleRateHz() const = 0;
   virtual size_t NumChannels() const = 0;
 
   // Returns the rate at which the RTP timestamps are updated. The default
   // implementation returns SampleRateHz().
   virtual int RtpTimestampRateHz() const;
 
@@ skipping 87 matching lines @@
   // Provides uplink packet loss fraction to this encoder to allow it to adapt.
   virtual void OnReceivedUplinkPacketLossFraction(
       float uplink_packet_loss_fraction);
 
   // Provides target audio bitrate to this encoder to allow it to adapt.
   virtual void OnReceivedTargetAudioBitrate(int target_audio_bitrate_bps);
 
   // Provides RTT to this encoder to allow it to adapt.
   virtual void OnReceivedRtt(int rtt_ms);
 
+  // Provides overhead size to this encoder for it to determine the bitrate
+  // for the payload.

[kwiberg-webrtc, 2016/11/29 09:46:24]

Please consider spending a few more words describing what this number means. Is
it the number of bytes added to each packet the encoder emits?

[minyue-webrtc, 2016/12/02 10:07:34]

Done.

+  virtual void OnReceivedOverhead(size_t overhead_bytes_per_packet);
+
   // To allow encoder to adapt its frame length, it must be provided the frame
   // length range that receivers can accept.
   virtual void SetReceiverFrameLengthRange(int min_frame_length_ms,
                                            int max_frame_length_ms);
 
  protected:
   // Subclasses implement this to perform the actual encoding. Called by
   // Encode().
   virtual EncodedInfo EncodeImpl(uint32_t rtp_timestamp,
                                  rtc::ArrayView<const int16_t> audio,
                                  rtc::Buffer* encoded) = 0;
+  rtc::Optional<size_t> overhead_bytes_per_packet_;

[kwiberg-webrtc, 2016/11/29 09:46:24]

Interface classes should not have data members!

In this particular case, the reason you need a data member seems to be that the
new default implementation of OnReceivedTargetAudioBitrate uses it when
computing the value to pass to SetTargetBitrate. But the default implementation
of SetTargetBitrate does nothing, so it would seem appropriate to let the
default OnReceivedTargetAudioBitrate and OnReceivedOverhead do nothing as well,
and let the classes that actually want to do something with the per-packet
overhead implement this.

[minyue-webrtc, 2016/12/02 10:07:34]

Done.

 };
 }  // namespace webrtc
 #endif  // WEBRTC_MODULES_AUDIO_CODING_CODECS_AUDIO_ENCODER_H_