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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}
59 lines
1.4 KiB
C++
59 lines
1.4 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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#include "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 "test/gtest.h"
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namespace webrtc {
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class JitterBufferDelayTest : public ::testing::Test {
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public:
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JitterBufferDelayTest() {}
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protected:
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JitterBufferDelay delay_;
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};
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TEST_F(JitterBufferDelayTest, Set) {
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// Delay in seconds.
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delay_.Set(3.0);
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EXPECT_EQ(delay_.GetMs(), 3000);
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}
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TEST_F(JitterBufferDelayTest, DefaultValue) {
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EXPECT_EQ(delay_.GetMs(), 0); // Default value is 0ms.
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}
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TEST_F(JitterBufferDelayTest, Clamping) {
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// In current Jitter Buffer implementation (Audio or Video) maximum supported
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// value is 10000 milliseconds.
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delay_.Set(10.5);
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EXPECT_EQ(delay_.GetMs(), 10000);
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// Test int overflow.
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delay_.Set(21474836470.0);
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EXPECT_EQ(delay_.GetMs(), 10000);
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delay_.Set(-21474836470.0);
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EXPECT_EQ(delay_.GetMs(), 0);
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// Boundary value in seconds to milliseconds conversion.
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delay_.Set(0.0009);
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EXPECT_EQ(delay_.GetMs(), 0);
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delay_.Set(-2.0);
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EXPECT_EQ(delay_.GetMs(), 0);
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}
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} // namespace webrtc
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