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4ccdf932e1
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}
42 lines
1.2 KiB
C++
42 lines
1.2 KiB
C++
/*
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* Copyright 2015 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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#include "pc/rtp_receiver.h"
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#include <stddef.h>
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#include <utility>
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#include <vector>
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#include "api/media_stream_proxy.h"
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#include "pc/media_stream.h"
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#include "rtc_base/location.h"
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namespace webrtc {
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// This function is only expected to be called on the signalling thread.
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int RtpReceiverInternal::GenerateUniqueId() {
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static int g_unique_id = 0;
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return ++g_unique_id;
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}
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std::vector<rtc::scoped_refptr<MediaStreamInterface>>
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RtpReceiverInternal::CreateStreamsFromIds(std::vector<std::string> stream_ids) {
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std::vector<rtc::scoped_refptr<MediaStreamInterface>> streams(
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stream_ids.size());
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for (size_t i = 0; i < stream_ids.size(); ++i) {
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streams[i] = MediaStreamProxy::Create(
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rtc::Thread::Current(), MediaStream::Create(std::move(stream_ids[i])));
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}
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return streams;
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}
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} // namespace webrtc
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