Index: webrtc/modules/audio_coding/codecs/audio_encoder.cc |
diff --git a/webrtc/modules/audio_coding/codecs/audio_encoder.cc b/webrtc/modules/audio_coding/codecs/audio_encoder.cc |
index 7b7325dc551311f0fcab9c412c7c73aefb033879..61d0c1b1f88086ec2adfa04c5c10e61ba40445d8 100644 |
--- a/webrtc/modules/audio_coding/codecs/audio_encoder.cc |
+++ b/webrtc/modules/audio_coding/codecs/audio_encoder.cc |
@@ -72,12 +72,17 @@ bool AudioEncoder::EnableAudioNetworkAdaptor(const std::string& config_string, |
void AudioEncoder::DisableAudioNetworkAdaptor() {} |
-void AudioEncoder::OnReceivedUplinkBandwidth(int uplink_bandwidth_bps) {} |
- |
void AudioEncoder::OnReceivedUplinkPacketLossFraction( |
float uplink_packet_loss_fraction) {} |
-void AudioEncoder::OnReceivedTargetAudioBitrate(int target_audio_bitrate_bps) {} |
+void AudioEncoder::OnReceivedTargetAudioBitrate(int target_audio_bitrate_bps) { |
+ OnReceivedTargetAudioBitrate(target_audio_bitrate_bps, |
+ rtc::Optional<int64_t>()); |
+} |
+ |
+void AudioEncoder::OnReceivedTargetAudioBitrate( |
+ int target_audio_bitrate_bps, |
+ rtc::Optional<int64_t> probing_interval_ms) {} |
void AudioEncoder::OnReceivedRtt(int rtt_ms) {} |