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Migrate video/ except video/end_to_end_tests and video/adaptation to webrtc::Mutex.

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Also migrates test/ partly.

Bug: webrtc:11567
Change-Id: If5b2eae65c5f297f364b6e3c67f94946a09b4a96
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/178862
Commit-Queue: Markus Handell <handellm@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31672}
This commit is contained in:
Markus Handell
2020-07-08 13:13:32 +02:00
committed by Commit Bot
parent a9e1b49704
commit a376518817
34 changed files with 506 additions and 497 deletions
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6
video/rtp_streams_synchronizer.cc
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@ -51,7 +51,7 @@ RtpStreamsSynchronizer::RtpStreamsSynchronizer(Syncable* syncable_video)
RtpStreamsSynchronizer::~RtpStreamsSynchronizer() = default;
void RtpStreamsSynchronizer::ConfigureSync(Syncable* syncable_audio) {
rtc::CritScope lock(&crit_);
MutexLock lock(&mutex_);
if (syncable_audio == syncable_audio_) {
// This prevents expensive no-ops.
return;
@ -76,7 +76,7 @@ void RtpStreamsSynchronizer::Process() {
RTC_DCHECK_RUN_ON(&process_thread_checker_);
last_sync_time_ = rtc::TimeNanos();
rtc::CritScope lock(&crit_);
MutexLock lock(&mutex_);
if (!syncable_audio_) {
return;
}
@ -157,7 +157,7 @@ bool RtpStreamsSynchronizer::GetStreamSyncOffsetInMs(
int64_t* video_playout_ntp_ms,
int64_t* stream_offset_ms,
double* estimated_freq_khz) const {
rtc::CritScope lock(&crit_);
MutexLock lock(&mutex_);
if (!syncable_audio_) {
return false;
}