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..19dc3327578ec3c1853da96599052612f82b3e77 100644 |
--- a/webrtc/modules/audio_coding/codecs/audio_encoder.h |
+++ b/webrtc/modules/audio_coding/codecs/audio_encoder.h |
@@ -21,6 +21,8 @@ |
namespace webrtc { |
+class Clock; |
+ |
// This is the interface class for encoders in AudioCoding module. Each codec |
// type must have an implementation of this class. |
class AudioEncoder { |
@@ -162,6 +164,31 @@ class AudioEncoder { |
virtual rtc::ArrayView<std::unique_ptr<AudioEncoder>> |
ReclaimContainedEncoders(); |
+ // Enables audio network adaptor. Returns true if successful. |
+ virtual bool EnableAudioNetworkAdaptor(const std::string& config_string, |
+ const Clock* clock); |
+ |
+ // Disables audio network adaptor. |
+ virtual void DisableAudioNetworkAdaptor(); |
+ |
+ // 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); |
+ |
+ // To allow encoder to adapt its frame length, it must be provided the frame |
+ // length range that receives can accept. |
+ virtual void SetReceiverFrameLengthRange(int min_frame_length_ms, |
+ int max_frame_length_ms); |
+ |
protected: |
// Subclasses implement this to perform the actual encoding. Called by |
// Encode(). |