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Unified Diff: media/cast/sender/audio_sender.h

Issue 514263002: Move common code and variables from audio/video sender to frame sender. (Closed) Base URL: https://chromium.googlesource.com/chromium/src.git@master
Patch Set: Created 6 years, 4 months ago
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Index: media/cast/sender/audio_sender.h
diff --git a/media/cast/sender/audio_sender.h b/media/cast/sender/audio_sender.h
index e07b892a73eadd6b40c1fff3629ea540972f1b18..4ca539b2fc220b3f527e3e1d89c0eae5b0f74efb 100644
--- a/media/cast/sender/audio_sender.h
+++ b/media/cast/sender/audio_sender.h
@@ -55,15 +55,6 @@ class AudioSender : public FrameSender,
void OnReceivedCastFeedback(const RtcpCastMessage& cast_feedback);
private:
- // Schedule and execute periodic checks for re-sending packets. If no
- // acknowledgements have been received for "too long," AudioSender will
- // speculatively re-send certain packets of an unacked frame to kick-start
- // re-transmission. This is a last resort tactic to prevent the session from
- // getting stuck after a long outage.
- void ScheduleNextResendCheck();
- void ResendCheck();
- void ResendForKickstart();
-
// Returns true if there are too many frames in flight, as defined by the
// configured target playout delay plus simple logic. When this is true,
// InsertAudio() will silenty drop frames instead of sending them to the audio
@@ -77,39 +68,6 @@ class AudioSender : public FrameSender,
scoped_ptr<AudioEncoder> audio_encoder_;
const int configured_encoder_bitrate_;
- // Counts how many RTCP reports are being "aggressively" sent (i.e., one per
- // frame) at the start of the session. Once a threshold is reached, RTCP
- // reports are instead sent at the configured interval + random drift.
- int num_aggressive_rtcp_reports_sent_;
-
- // This is "null" until the first frame is sent. Thereafter, this tracks the
- // last time any frame was sent or re-sent.
- base::TimeTicks last_send_time_;
-
- // The ID of the last frame sent. Logic throughout AudioSender assumes this
- // can safely wrap-around. This member is invalid until
- // |!last_send_time_.is_null()|.
- uint32 last_sent_frame_id_;
-
- // The ID of the latest (not necessarily the last) frame that has been
- // acknowledged. Logic throughout AudioSender assumes this can safely
- // wrap-around. This member is invalid until |!last_send_time_.is_null()|.
- uint32 latest_acked_frame_id_;
-
- // Counts the number of duplicate ACK that are being received. When this
- // number reaches a threshold, the sender will take this as a sign that the
- // receiver hasn't yet received the first packet of the next frame. In this
- // case, AudioSender will trigger a re-send of the next frame.
- int duplicate_ack_counter_;
-
- // If this sender is ready for use, this is STATUS_AUDIO_INITIALIZED.
- CastInitializationStatus cast_initialization_status_;
-
- // This is a "good enough" mapping for finding the RTP timestamp associated
- // with a video frame. The key is the lowest 8 bits of frame id (which is
- // what is sent via RTCP). This map is used for logging purposes.
- RtpTimestamp frame_id_to_rtp_timestamp_[256];
-
// NOTE: Weak pointers must be invalidated before all other member variables.
base::WeakPtrFactory<AudioSender> weak_factory_;
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