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 f30d657d133358eb20d84afec5a4e98c0062138f..a7cbebeb122d63f134b29e18c8bc0e04e171168e 100644 |
--- a/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.h |
+++ b/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.h |
@@ -113,6 +113,7 @@ 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 { |