| 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);
|
| };
|
|
|