Adding audio network adaptor to AudioEncoderOpus.
BUG=webrtc:6303
Review-Url: https://codereview.webrtc.org/2362703002
Cr-Commit-Position: refs/heads/master@{#14555}
diff --git a/webrtc/modules/audio_coding/codecs/audio_encoder.h b/webrtc/modules/audio_coding/codecs/audio_encoder.h
index f09525f..19dc332 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 @@
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().