Index: media/cast/audio_receiver/audio_receiver.h |
diff --git a/media/cast/audio_receiver/audio_receiver.h b/media/cast/audio_receiver/audio_receiver.h |
index 5cc8f88f4cd44029d9b60d0da040bd48cdbb4f14..dd0decb44d99af099df5e9b372dac1ab0bd2f262 100644 |
--- a/media/cast/audio_receiver/audio_receiver.h |
+++ b/media/cast/audio_receiver/audio_receiver.h |
@@ -39,10 +39,6 @@ class AudioDecoder; |
// each step of the pipeline (i.e., encode frame, then transmit/retransmit from |
// the sender, then receive and re-order packets on the receiver, then decode |
// frame) can vary in duration and is typically very hard to predict. |
-// Heuristics will determine when the targeted playout delay is insufficient in |
-// the current environment; and the receiver can then increase the playout |
-// delay, notifying the sender, to account for the extra variance. |
-// TODO(miu): Make the last sentence true. http://crbug.com/360111 |
// |
// Two types of frames can be requested: 1) A frame of decoded audio data; or 2) |
// a frame of still-encoded audio data, to be passed into an external audio |
@@ -81,10 +77,6 @@ class AudioReceiver : public RtpReceiver, |
// Deliver another packet, possibly a duplicate, and possibly out-of-order. |
void IncomingPacket(scoped_ptr<Packet> packet); |
- // Update target audio delay used to compute the playout time. Rtcp |
- // will also be updated (will be included in all outgoing reports). |
- void SetTargetDelay(base::TimeDelta target_delay); |
- |
protected: |
friend class AudioReceiverTest; // Invokes OnReceivedPayloadData(). |
@@ -102,6 +94,10 @@ class AudioReceiver : public RtpReceiver, |
// the future to wait for missing/incomplete frames. |
void EmitAvailableEncodedFrames(); |
+ // Helper used by EmitAvailableEncodedFrames() to schedule itself to be called |
+ // again after |wait_time| has elapsed. |
+ void RetryEmitAfterWaiting(base::TimeDelta wait_time); |
+ |
// Clears the |is_waiting_for_consecutive_frame_| flag and invokes |
// EmitAvailableEncodedFrames(). |
void EmitAvailableEncodedFramesAfterWaiting(); |
@@ -113,10 +109,9 @@ class AudioReceiver : public RtpReceiver, |
scoped_ptr<transport::EncodedAudioFrame> encoded_frame, |
const base::TimeTicks& playout_time); |
- // Return the playout time based on the current time and rtp timestamp. |
- base::TimeTicks GetPlayoutTime(base::TimeTicks now, uint32 rtp_timestamp); |
- |
- void InitializeTimers(); |
+ // Computes the playout time for a frame with the given |rtp_timestamp|. If |
+ // lip-sync info is not available, a best-guess is returned (a hack). |
+ base::TimeTicks GetPlayoutTime(uint32 rtp_timestamp) const; |
// Schedule the next RTCP report. |
void ScheduleNextRtcpReport(); |
@@ -152,13 +147,11 @@ class AudioReceiver : public RtpReceiver, |
const transport::AudioCodec codec_; |
const int frequency_; |
- base::TimeDelta target_delay_delta_; |
+ base::TimeDelta target_playout_delay_; |
+ bool reports_are_scheduled_; |
hubbe
2014/05/14 23:12:23
Maybe add comments for these variables?
miu
2014/05/16 22:45:47
Done.
|
Framer framer_; |
scoped_ptr<AudioDecoder> audio_decoder_; |
Rtcp rtcp_; |
- base::TimeDelta time_offset_; |
- base::TimeTicks time_first_incoming_packet_; |
- uint32 first_incoming_rtp_timestamp_; |
transport::TransportEncryptionHandler decryptor_; |
// Outstanding callbacks to run to deliver on client requests for frames. |
@@ -166,7 +159,7 @@ class AudioReceiver : public RtpReceiver, |
// True while there's an outstanding task to re-invoke |
// EmitAvailableEncodedFrames(). |
- bool is_waiting_for_consecutive_frame_; |
+ bool is_waiting_to_emit_frames_; |
// This mapping allows us to log AUDIO_ACK_SENT as a frame event. In addition |
// it allows the event to be transmitted via RTCP. |