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 e164fb5065dcf49cd1ed8e831e0bfa8974ac0fea..2c8d9ce5ece0bf97fa35820087ad0773f827c641 100644 |
--- a/webrtc/modules/audio_coding/codecs/audio_encoder.h |
+++ b/webrtc/modules/audio_coding/codecs/audio_encoder.h |
@@ -144,6 +144,17 @@ |
// implementation does nothing. |
virtual void SetMaxPlaybackRate(int frequency_hz); |
+ // Tells the encoder what the projected packet loss rate is. The rate is in |
+ // the range [0.0, 1.0]. The encoder would typically use this information to |
+ // adjust channel coding efforts, such as FEC. The default implementation |
+ // does nothing. |
+ virtual void SetProjectedPacketLossRate(double fraction); |
+ |
+ // Tells the encoder what average bitrate we'd like it to produce. The |
+ // encoder is free to adjust or disregard the given bitrate (the default |
+ // implementation does the latter). |
+ virtual void SetTargetBitrate(int target_bps); |
+ |
// Causes this encoder to let go of any other encoders it contains, and |
// returns a pointer to an array where they are stored (which is required to |
// live as long as this encoder). Unless the returned array is empty, you may |
@@ -164,7 +175,6 @@ |
virtual void OnReceivedUplinkBandwidth(int uplink_bandwidth_bps); |
// Provides uplink packet loss fraction to this encoder to allow it to adapt. |
- // |uplink_packet_loss_fraction| is in the range [0.0, 1.0]. |
virtual void OnReceivedUplinkPacketLossFraction( |
float uplink_packet_loss_fraction); |