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

Browse Source 
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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14
modules/audio_coding/acm2/acm_receiver.cc
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@ -86,7 +86,7 @@ int AcmReceiver::GetBaseMinimumDelayMs() const {
}
absl::optional<int> AcmReceiver::last_packet_sample_rate_hz() const {
rtc::CritScope lock(&crit_sect_);
MutexLock lock(&mutex_);
if (!last_decoder_) {
return absl::nullopt;
}
@ -118,7 +118,7 @@ int AcmReceiver::InsertPacket(const RTPHeader& rtp_header,
}
{
rtc::CritScope lock(&crit_sect_);
MutexLock lock(&mutex_);
if (absl::EqualsIgnoreCase(format->sdp_format.name, "cn")) {
if (last_decoder_ && last_decoder_->num_channels > 1) {
// This is a CNG and the audio codec is not mono, so skip pushing in
@ -131,7 +131,7 @@ int AcmReceiver::InsertPacket(const RTPHeader& rtp_header,
/*num_channels=*/format->num_channels,
/*sdp_format=*/std::move(format->sdp_format)};
}
} // |crit_sect_| is released.
} // |mutex_| is released.
if (neteq_->InsertPacket(rtp_header, incoming_payload) < 0) {
RTC_LOG(LERROR) << "AcmReceiver::InsertPacket "
@ -147,7 +147,7 @@ int AcmReceiver::GetAudio(int desired_freq_hz,
bool* muted) {
RTC_DCHECK(muted);
// Accessing members, take the lock.
rtc::CritScope lock(&crit_sect_);
MutexLock lock(&mutex_);
if (neteq_->GetAudio(audio_frame, muted) != NetEq::kOK) {
RTC_LOG(LERROR) << "AcmReceiver::GetAudio - NetEq Failed.";
@ -217,7 +217,7 @@ void AcmReceiver::FlushBuffers() {
}
void AcmReceiver::RemoveAllCodecs() {
rtc::CritScope lock(&crit_sect_);
MutexLock lock(&mutex_);
neteq_->RemoveAllPayloadTypes();
last_decoder_ = absl::nullopt;
}
@ -236,7 +236,7 @@ int AcmReceiver::TargetDelayMs() const {
absl::optional<std::pair<int, SdpAudioFormat>> AcmReceiver::LastDecoder()
const {
rtc::CritScope lock(&crit_sect_);
MutexLock lock(&mutex_);
if (!last_decoder_) {
return absl::nullopt;
}
@ -327,7 +327,7 @@ uint32_t AcmReceiver::NowInTimestamp(int decoder_sampling_rate) const {
void AcmReceiver::GetDecodingCallStatistics(
AudioDecodingCallStats* stats) const {
rtc::CritScope lock(&crit_sect_);
MutexLock lock(&mutex_);
*stats = call_stats_.GetDecodingStatistics();
}