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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
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31
pc/jitter_buffer_delay.h
View File

@ -14,36 +14,25 @@
#include <stdint.h>
#include "absl/types/optional.h"
#include "media/base/delayable.h"
#include "pc/jitter_buffer_delay_interface.h"
#include "rtc_base/thread.h"
#include "api/sequence_checker.h"
#include "rtc_base/system/no_unique_address.h"
namespace webrtc {
// JitterBufferDelay converts delay from seconds to milliseconds for the
// underlying media channel. It also handles cases when user sets delay before
// the start of media_channel by caching its request. Note, this class is not
// thread safe. Its thread safe version is defined in
// pc/jitter_buffer_delay_proxy.h
class JitterBufferDelay : public JitterBufferDelayInterface {
// the start of media_channel by caching its request.
class JitterBufferDelay {
public:
// Must be called on signaling thread.
explicit JitterBufferDelay(rtc::Thread* worker_thread);
JitterBufferDelay();
void OnStart(cricket::Delayable* media_channel, uint32_t ssrc) override;
void OnStop() override;
void Set(absl::optional<double> delay_seconds) override;
void Set(absl::optional<double> delay_seconds);
int GetMs() const;
private:
// Throughout webrtc source, sometimes it is also called as |main_thread_|.
rtc::Thread* const signaling_thread_;
rtc::Thread* const worker_thread_;
// Media channel and ssrc together uniqely identify audio stream.
cricket::Delayable* media_channel_ = nullptr;
absl::optional<uint32_t> ssrc_;
absl::optional<double> cached_delay_seconds_;
RTC_NO_UNIQUE_ADDRESS SequenceChecker worker_thread_checker_;
absl::optional<double> cached_delay_seconds_
RTC_GUARDED_BY(&worker_thread_checker_);
};
} // namespace webrtc