| 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 c433dcdd0f1c9613cb7a9be44bf28d3f06237185..1216484b4d189d3dd3b93247019851db3a3f17f5 100644
|
| --- a/webrtc/modules/audio_coding/codecs/audio_encoder.cc
|
| +++ b/webrtc/modules/audio_coding/codecs/audio_encoder.cc
|
| @@ -67,4 +67,23 @@ void AudioEncoder::SetTargetBitrate(int target_bps) {}
|
| rtc::ArrayView<std::unique_ptr<AudioEncoder>>
|
| AudioEncoder::ReclaimContainedEncoders() { return nullptr; }
|
|
|
| +bool AudioEncoder::EnableAudioNetworkAdaptor(const std::string& config_string,
|
| + const Clock* clock) {
|
| + return false;
|
| +}
|
| +
|
| +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::OnReceivedRtt(int rtt_ms) {}
|
| +
|
| +void AudioEncoder::SetReceiverFrameLengthRange(int min_frame_length_ms,
|
| + int max_frame_length_ms) {}
|
| +
|
| } // namespace webrtc
|
|
|