Index: webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.h |
diff --git a/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.h b/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.h |
index 92f2126b98d5d6a5a9bd4d1ff2ef246b912ece71..0864c7bfb561b6af167c5ba13fd395571b81b8bf 100644 |
--- a/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.h |
+++ b/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.h |
@@ -110,6 +110,7 @@ class AudioEncoderOpus final : public AudioEncoder { |
float uplink_packet_loss_fraction) override; |
void OnReceivedTargetAudioBitrate(int target_audio_bitrate_bps) override; |
void OnReceivedRtt(int rtt_ms) override; |
+ void OnReceivedOverhead(size_t overhead_bytes_per_packet) override; |
void SetReceiverFrameLengthRange(int min_frame_length_ms, |
int max_frame_length_ms) override; |
rtc::ArrayView<const int> supported_frame_lengths_ms() const { |
@@ -159,6 +160,7 @@ class AudioEncoderOpus final : public AudioEncoder { |
std::unique_ptr<PacketLossFractionSmoother> packet_loss_fraction_smoother_; |
AudioNetworkAdaptorCreator audio_network_adaptor_creator_; |
std::unique_ptr<AudioNetworkAdaptor> audio_network_adaptor_; |
+ rtc::Optional<size_t> overhead_bytes_per_packet_; |
RTC_DISALLOW_COPY_AND_ASSIGN(AudioEncoderOpus); |
}; |