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VideoRtpReceiver & AudioRtpReceiver threading fixes.

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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}
This commit is contained in:
Tommi
2021-05-17 14:50:10 +02:00
committed by WebRTC LUCI CQ
parent b27a9f9481
commit 4ccdf932e1
21 changed files with 585 additions and 583 deletions
Download Patch File Download Diff File Expand all files Collapse all files

64
pc/jitter_buffer_delay_unittest.cc
View File

@ -13,79 +13,47 @@
#include <stdint.h>
#include "absl/types/optional.h"
#include "api/scoped_refptr.h"
#include "pc/test/mock_delayable.h"
#include "rtc_base/ref_counted_object.h"
#include "rtc_base/thread.h"
#include "test/gmock.h"
#include "test/gtest.h"
using ::testing::Return;
namespace {
constexpr int kSsrc = 1234;
} // namespace
namespace webrtc {
class JitterBufferDelayTest : public ::testing::Test {
public:
JitterBufferDelayTest()
: delay_(
rtc::make_ref_counted<JitterBufferDelay>(rtc::Thread::Current())) {}
JitterBufferDelayTest() {}
protected:
rtc::scoped_refptr<JitterBufferDelayInterface> delay_;
MockDelayable delayable_;
JitterBufferDelay delay_;
};
TEST_F(JitterBufferDelayTest, Set) {
delay_->OnStart(&delayable_, kSsrc);
EXPECT_CALL(delayable_, SetBaseMinimumPlayoutDelayMs(kSsrc, 3000))
.WillOnce(Return(true));
// Delay in seconds.
delay_->Set(3.0);
delay_.Set(3.0);
EXPECT_EQ(delay_.GetMs(), 3000);
}
TEST_F(JitterBufferDelayTest, Caching) {
// Check that value is cached before start.
delay_->Set(4.0);
// Check that cached value applied on the start.
EXPECT_CALL(delayable_, SetBaseMinimumPlayoutDelayMs(kSsrc, 4000))
.WillOnce(Return(true));
delay_->OnStart(&delayable_, kSsrc);
TEST_F(JitterBufferDelayTest, DefaultValue) {
EXPECT_EQ(delay_.GetMs(), 0); // Default value is 0ms.
}
TEST_F(JitterBufferDelayTest, Clamping) {
delay_->OnStart(&delayable_, kSsrc);
// In current Jitter Buffer implementation (Audio or Video) maximum supported
// value is 10000 milliseconds.
EXPECT_CALL(delayable_, SetBaseMinimumPlayoutDelayMs(kSsrc, 10000))
.WillOnce(Return(true));
delay_->Set(10.5);
delay_.Set(10.5);
EXPECT_EQ(delay_.GetMs(), 10000);
// Test int overflow.
EXPECT_CALL(delayable_, SetBaseMinimumPlayoutDelayMs(kSsrc, 10000))
.WillOnce(Return(true));
delay_->Set(21474836470.0);
delay_.Set(21474836470.0);
EXPECT_EQ(delay_.GetMs(), 10000);
EXPECT_CALL(delayable_, SetBaseMinimumPlayoutDelayMs(kSsrc, 0))
.WillOnce(Return(true));
delay_->Set(-21474836470.0);
delay_.Set(-21474836470.0);
EXPECT_EQ(delay_.GetMs(), 0);
// Boundary value in seconds to milliseconds conversion.
EXPECT_CALL(delayable_, SetBaseMinimumPlayoutDelayMs(kSsrc, 0))
.WillOnce(Return(true));
delay_->Set(0.0009);
delay_.Set(0.0009);
EXPECT_EQ(delay_.GetMs(), 0);
EXPECT_CALL(delayable_, SetBaseMinimumPlayoutDelayMs(kSsrc, 0))
.WillOnce(Return(true));
delay_->Set(-2.0);
delay_.Set(-2.0);
EXPECT_EQ(delay_.GetMs(), 0);
}
} // namespace webrtc