zqz/platform-external-webrtc
1
0
Fork 0
Code Issues Pull Requests Actions Packages Projects Releases Wiki Activity

Migrate modules/video_coding to webrtc::Mutex.

Browse Source 
Bug: webrtc:11567
Change-Id: I8023fbe7595f7ba8ae7c7db3583fc2e560ec3df2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/178803
Commit-Queue: Markus Handell <handellm@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31644}
This commit is contained in:
Markus Handell
2020-07-07 12:17:12 +02:00
committed by Commit Bot
parent fb6f975401
commit 6deec38ede
23 changed files with 190 additions and 191 deletions
Download Patch File Download Diff File Expand all files Collapse all files

32
modules/video_coding/timing.h
View File

@ -16,7 +16,7 @@
#include "absl/types/optional.h"
#include "api/video/video_timing.h"
#include "modules/video_coding/codec_timer.h"
#include "rtc_base/critical_section.h"
#include "rtc_base/synchronization/mutex.h"
#include "rtc_base/thread_annotations.h"
namespace webrtc {
@ -104,30 +104,30 @@ class VCMTiming {
enum { kDelayMaxChangeMsPerS = 100 };
protected:
int RequiredDecodeTimeMs() const RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
int RequiredDecodeTimeMs() const RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
int64_t RenderTimeMsInternal(uint32_t frame_timestamp, int64_t now_ms) const
RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
int TargetDelayInternal() const RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
int TargetDelayInternal() const RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
private:
rtc::CriticalSection crit_sect_;
mutable Mutex mutex_;
Clock* const clock_;
bool master_ RTC_GUARDED_BY(crit_sect_);
TimestampExtrapolator* ts_extrapolator_ RTC_GUARDED_BY(crit_sect_);
std::unique_ptr<VCMCodecTimer> codec_timer_ RTC_GUARDED_BY(crit_sect_);
int render_delay_ms_ RTC_GUARDED_BY(crit_sect_);
bool master_ RTC_GUARDED_BY(mutex_);
TimestampExtrapolator* ts_extrapolator_ RTC_GUARDED_BY(mutex_);
std::unique_ptr<VCMCodecTimer> codec_timer_ RTC_GUARDED_BY(mutex_);
int render_delay_ms_ RTC_GUARDED_BY(mutex_);
// Best-effort playout delay range for frames from capture to render.
// The receiver tries to keep the delay between |min_playout_delay_ms_|
// and |max_playout_delay_ms_| taking the network jitter into account.
// A special case is where min_playout_delay_ms_ = max_playout_delay_ms_ = 0,
// in which case the receiver tries to play the frames as they arrive.
int min_playout_delay_ms_ RTC_GUARDED_BY(crit_sect_);
int max_playout_delay_ms_ RTC_GUARDED_BY(crit_sect_);
int jitter_delay_ms_ RTC_GUARDED_BY(crit_sect_);
int current_delay_ms_ RTC_GUARDED_BY(crit_sect_);
uint32_t prev_frame_timestamp_ RTC_GUARDED_BY(crit_sect_);
absl::optional<TimingFrameInfo> timing_frame_info_ RTC_GUARDED_BY(crit_sect_);
size_t num_decoded_frames_ RTC_GUARDED_BY(crit_sect_);
int min_playout_delay_ms_ RTC_GUARDED_BY(mutex_);
int max_playout_delay_ms_ RTC_GUARDED_BY(mutex_);
int jitter_delay_ms_ RTC_GUARDED_BY(mutex_);
int current_delay_ms_ RTC_GUARDED_BY(mutex_);
uint32_t prev_frame_timestamp_ RTC_GUARDED_BY(mutex_);
absl::optional<TimingFrameInfo> timing_frame_info_ RTC_GUARDED_BY(mutex_);
size_t num_decoded_frames_ RTC_GUARDED_BY(mutex_);
};
} // namespace webrtc