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Migrate modules/audio_coding, audio_mixer/ and audio_processing/ to webrtc::Mutex.

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Bug: webrtc:11567
Change-Id: I03b78bd2e411e9bcca199f85e4457511826cd17e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176745
Commit-Queue: Markus Handell <handellm@webrtc.org>
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31649}
This commit is contained in:
Markus Handell
2020-07-07 15:53:34 +02:00
committed by Commit Bot
parent 1e257cacbf
commit 0df0faefd5
16 changed files with 314 additions and 312 deletions
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16
modules/audio_coding/test/Channel.cc
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@ -58,7 +58,7 @@ int32_t Channel::SendData(AudioFrameType frameType,
}
}
_channelCritSect.Enter();
_channelCritSect.Lock();
if (_saveBitStream) {
// fwrite(payloadData, sizeof(uint8_t), payloadSize, _bitStreamFile);
}
@ -69,7 +69,7 @@ int32_t Channel::SendData(AudioFrameType frameType,
_useLastFrameSize = false;
_lastInTimestamp = timeStamp;
_totalBytes += payloadDataSize;
_channelCritSect.Leave();
_channelCritSect.Unlock();
if (_useFECTestWithPacketLoss) {
_packetLoss += 1;
@ -238,7 +238,7 @@ void Channel::RegisterReceiverACM(AudioCodingModule* acm) {
void Channel::ResetStats() {
int n;
int k;
_channelCritSect.Enter();
_channelCritSect.Lock();
_lastPayloadType = -1;
for (n = 0; n < MAX_NUM_PAYLOADS; n++) {
_payloadStats[n].payloadType = -1;
@ -253,23 +253,23 @@ void Channel::ResetStats() {
}
_beginTime = rtc::TimeMillis();
_totalBytes = 0;
_channelCritSect.Leave();
_channelCritSect.Unlock();
}
uint32_t Channel::LastInTimestamp() {
uint32_t timestamp;
_channelCritSect.Enter();
_channelCritSect.Lock();
timestamp = _lastInTimestamp;
_channelCritSect.Leave();
_channelCritSect.Unlock();
return timestamp;
}
double Channel::BitRate() {
double rate;
uint64_t currTime = rtc::TimeMillis();
_channelCritSect.Enter();
_channelCritSect.Lock();
rate = ((double)_totalBytes * 8.0) / (double)(currTime - _beginTime);
_channelCritSect.Leave();
_channelCritSect.Unlock();
return rate;
}

4
modules/audio_coding/test/Channel.h
View File

@ -15,7 +15,7 @@
#include "modules/audio_coding/include/audio_coding_module.h"
#include "modules/include/module_common_types.h"
#include "rtc_base/critical_section.h"
#include "rtc_base/synchronization/mutex.h"
namespace webrtc {
@ -88,7 +88,7 @@ class Channel : public AudioPacketizationCallback {
// 60msec * 32 sample(max)/msec * 2 description (maybe) * 2 bytes/sample
uint8_t _payloadData[60 * 32 * 2 * 2];
rtc::CriticalSection _channelCritSect;
Mutex _channelCritSect;
FILE* _bitStreamFile;
bool _saveBitStream;
int16_t _lastPayloadType;