Index: webrtc/modules/audio_coding/codecs/audio_encoder.h |
diff --git a/webrtc/modules/audio_coding/codecs/audio_encoder.h b/webrtc/modules/audio_coding/codecs/audio_encoder.h |
index f09525f145d2cc661bf2da00931ac9e2da10f22f..8869c974cffe00bcecebd1e24e572d0486da1fdf 100644 |
--- a/webrtc/modules/audio_coding/codecs/audio_encoder.h |
+++ b/webrtc/modules/audio_coding/codecs/audio_encoder.h |
@@ -162,6 +162,19 @@ class AudioEncoder { |
virtual rtc::ArrayView<std::unique_ptr<AudioEncoder>> |
ReclaimContainedEncoders(); |
+ // Provides uplink bandwidth to this encoder to allow it to adapt. |
+ virtual void OnReceivedUplinkBandwidth(int uplink_bandwidth_bps); |
+ |
+ // Provides uplink packet loss fraction to this encoder to allow it to adapt. |
+ virtual void OnReceivedUplinkPacketLossFraction( |
+ float uplink_packet_loss_fraction); |
+ |
+ // Provides target audio bitrate to this encoder to allow it to adapt. |
+ virtual void OnReceivedTargetAudioBitrate(int target_audio_bitrate_bps); |
+ |
+ // Provides RTT to this encoder to allow it to adapt. |
+ virtual void OnReceivedRtt(int rtt_ms); |
+ |
protected: |
// Subclasses implement this to perform the actual encoding. Called by |
// Encode(). |