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For implementations where the signaling and worker threads are not the same thread, this significantly cuts down on Thread::Invoke()s that would block the signaling thread while waiting for the worker thread. For Audio and Video Rtp receivers, the following methods now do not block the signaling thread: * GetParameters * SetJitterBufferMinimumDelay * GetSources * SetFrameDecryptor / GetFrameDecryptor * SetDepacketizerToDecoderFrameTransformer Importantly this change also makes the track() accessor accessible directly from the application thread (bypassing the proxy) since for receiver objects, the track object is const. Other changes: * Remove RefCountedObject inheritance, use make_ref_counted instead. * Every member variable in the rtp receiver classes is now RTC_GUARDED * Stop() now fully clears up worker thread state, and Stop() is consistently called before destruction. This means that there's one thread hop instead of at least 4 before (sometimes more), per receiver. * OnChanged triggered volume for audio tracks is done asynchronously. * Deleted most of the JitterBufferDelay implementation. Turns out that it was largely unnecessary overhead and complexity. It seems that these two classes are copy/pasted to a large extent so further refactoring would be good in the future, as to not have to fix each issue twice. Bug: chromium:1184611 Change-Id: I1ba5c3abbd1b0571f7d12850d64004fd2d83e5e2 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/218605 Commit-Queue: Tommi <tommi@webrtc.org> Reviewed-by: Markus Handell <handellm@webrtc.org> Cr-Commit-Position: refs/heads/master@{#34022}
40 lines
1.2 KiB
C++
40 lines
1.2 KiB
C++
/*
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* Copyright 2019 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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#ifndef PC_JITTER_BUFFER_DELAY_H_
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#define PC_JITTER_BUFFER_DELAY_H_
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#include <stdint.h>
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#include "absl/types/optional.h"
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#include "api/sequence_checker.h"
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#include "rtc_base/system/no_unique_address.h"
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namespace webrtc {
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// JitterBufferDelay converts delay from seconds to milliseconds for the
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// underlying media channel. It also handles cases when user sets delay before
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// the start of media_channel by caching its request.
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class JitterBufferDelay {
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public:
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JitterBufferDelay();
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void Set(absl::optional<double> delay_seconds);
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int GetMs() const;
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private:
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RTC_NO_UNIQUE_ADDRESS SequenceChecker worker_thread_checker_;
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absl::optional<double> cached_delay_seconds_
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RTC_GUARDED_BY(&worker_thread_checker_);
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};
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} // namespace webrtc
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#endif // PC_JITTER_BUFFER_DELAY_H_
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