Adding OnReceivedOverhead to AudioEncoder.

webrtc/modules/audio_coding/codecs/audio_encoder.h

Index: webrtc/modules/audio_coding/codecs/audio_encoder.h
diff --git a/webrtc/modules/audio_coding/codecs/audio_encoder.h b/webrtc/modules/audio_coding/codecs/audio_encoder.h
index 2c8d9ce5ece0bf97fa35820087ad0773f827c641..c4d239f0c00eeb45d1ebae8722e6233f12a1a85b 100644
--- a/webrtc/modules/audio_coding/codecs/audio_encoder.h
+++ b/webrtc/modules/audio_coding/codecs/audio_encoder.h
@@ -17,6 +17,7 @@
 #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 {
@@ -76,7 +77,8 @@ class AudioEncoder {
     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.
@@ -184,6 +186,10 @@ class AudioEncoder {
   // 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.
+  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,
@@ -195,6 +201,7 @@ class AudioEncoder {
   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_;
 };
 }  // namespace webrtc
 #endif  // WEBRTC_MODULES_AUDIO_CODING_CODECS_AUDIO_ENCODER_H_