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bef7b058f5
Setting a minimum delay can fail in some cases. It is important that the AV sync code is aware of failures and can act accordingly to recover and prevent sync delays that keep increasing indefinitely. Bug: webrtc:11805 Change-Id: I0deed951dc6c6d0905536a949af875e0a6d9f7fa Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/183360 Commit-Queue: Ivo Creusen <ivoc@webrtc.org> Reviewed-by: Åsa Persson <asapersson@webrtc.org> Reviewed-by: Björn Terelius <terelius@webrtc.org> Reviewed-by: Sam Zackrisson <saza@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32062}
46 lines
1.4 KiB
C++
46 lines
1.4 KiB
C++
/*
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* Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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// Syncable is used by RtpStreamsSynchronizer in VideoReceiveStream, and
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// implemented by AudioReceiveStream.
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#ifndef CALL_SYNCABLE_H_
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#define CALL_SYNCABLE_H_
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#include <stdint.h>
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#include "absl/types/optional.h"
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namespace webrtc {
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class Syncable {
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public:
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struct Info {
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int64_t latest_receive_time_ms = 0;
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uint32_t latest_received_capture_timestamp = 0;
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uint32_t capture_time_ntp_secs = 0;
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uint32_t capture_time_ntp_frac = 0;
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uint32_t capture_time_source_clock = 0;
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int current_delay_ms = 0;
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};
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virtual ~Syncable();
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virtual uint32_t id() const = 0;
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virtual absl::optional<Info> GetInfo() const = 0;
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virtual bool GetPlayoutRtpTimestamp(uint32_t* rtp_timestamp,
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int64_t* time_ms) const = 0;
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virtual bool SetMinimumPlayoutDelay(int delay_ms) = 0;
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virtual void SetEstimatedPlayoutNtpTimestampMs(int64_t ntp_timestamp_ms,
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int64_t time_ms) = 0;
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};
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} // namespace webrtc
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#endif // CALL_SYNCABLE_H_
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