| 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..751f9cf6fff4f6f59a071bc09c8fa053d3c36119 100644
|
| --- a/webrtc/modules/audio_coding/codecs/audio_encoder.cc
|
| +++ b/webrtc/modules/audio_coding/codecs/audio_encoder.cc
|
| @@ -72,12 +72,16 @@ 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) {
|
| + OnReceivedUplinkBandwidth(target_audio_bitrate_bps, rtc::Optional<int64_t>());
|
| +}
|
| +
|
| +void AudioEncoder::OnReceivedUplinkBandwidth(
|
| + int target_audio_bitrate_bps,
|
| + rtc::Optional<int64_t> probing_interval_ms) {}
|
|
|
| void AudioEncoder::OnReceivedRtt(int rtt_ms) {}
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|
|
|
|