Migrate modules/audio_coding, audio_mixer/ and audio_processing/ to webrtc::Mutex.
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}
diff --git a/modules/audio_coding/acm2/audio_coding_module.cc b/modules/audio_coding/acm2/audio_coding_module.cc
index a2d08ac..648ae6e 100644
--- a/modules/audio_coding/acm2/audio_coding_module.cc
+++ b/modules/audio_coding/acm2/audio_coding_module.cc
@@ -23,9 +23,9 @@
#include "modules/include/module_common_types_public.h"
#include "rtc_base/buffer.h"
#include "rtc_base/checks.h"
-#include "rtc_base/critical_section.h"
#include "rtc_base/logging.h"
#include "rtc_base/numerics/safe_conversions.h"
+#include "rtc_base/synchronization/mutex.h"
#include "rtc_base/thread_annotations.h"
#include "system_wrappers/include/metrics.h"
@@ -105,7 +105,7 @@
std::vector<int16_t> buffer;
};
- InputData input_data_ RTC_GUARDED_BY(acm_crit_sect_);
+ InputData input_data_ RTC_GUARDED_BY(acm_mutex_);
// This member class writes values to the named UMA histogram, but only if
// the value has changed since the last time (and always for the first call).
@@ -124,18 +124,18 @@
};
int Add10MsDataInternal(const AudioFrame& audio_frame, InputData* input_data)
- RTC_EXCLUSIVE_LOCKS_REQUIRED(acm_crit_sect_);
+ RTC_EXCLUSIVE_LOCKS_REQUIRED(acm_mutex_);
// TODO(bugs.webrtc.org/10739): change |absolute_capture_timestamp_ms| to
// int64_t when it always receives a valid value.
int Encode(const InputData& input_data,
absl::optional<int64_t> absolute_capture_timestamp_ms)
- RTC_EXCLUSIVE_LOCKS_REQUIRED(acm_crit_sect_);
+ RTC_EXCLUSIVE_LOCKS_REQUIRED(acm_mutex_);
- int InitializeReceiverSafe() RTC_EXCLUSIVE_LOCKS_REQUIRED(acm_crit_sect_);
+ int InitializeReceiverSafe() RTC_EXCLUSIVE_LOCKS_REQUIRED(acm_mutex_);
bool HaveValidEncoder(const char* caller_name) const
- RTC_EXCLUSIVE_LOCKS_REQUIRED(acm_crit_sect_);
+ RTC_EXCLUSIVE_LOCKS_REQUIRED(acm_mutex_);
// Preprocessing of input audio, including resampling and down-mixing if
// required, before pushing audio into encoder's buffer.
@@ -150,38 +150,38 @@
// 0: otherwise.
int PreprocessToAddData(const AudioFrame& in_frame,
const AudioFrame** ptr_out)
- RTC_EXCLUSIVE_LOCKS_REQUIRED(acm_crit_sect_);
+ RTC_EXCLUSIVE_LOCKS_REQUIRED(acm_mutex_);
// Change required states after starting to receive the codec corresponding
// to |index|.
int UpdateUponReceivingCodec(int index);
- rtc::CriticalSection acm_crit_sect_;
- rtc::Buffer encode_buffer_ RTC_GUARDED_BY(acm_crit_sect_);
- uint32_t expected_codec_ts_ RTC_GUARDED_BY(acm_crit_sect_);
- uint32_t expected_in_ts_ RTC_GUARDED_BY(acm_crit_sect_);
- acm2::ACMResampler resampler_ RTC_GUARDED_BY(acm_crit_sect_);
+ mutable Mutex acm_mutex_;
+ rtc::Buffer encode_buffer_ RTC_GUARDED_BY(acm_mutex_);
+ uint32_t expected_codec_ts_ RTC_GUARDED_BY(acm_mutex_);
+ uint32_t expected_in_ts_ RTC_GUARDED_BY(acm_mutex_);
+ acm2::ACMResampler resampler_ RTC_GUARDED_BY(acm_mutex_);
acm2::AcmReceiver receiver_; // AcmReceiver has it's own internal lock.
- ChangeLogger bitrate_logger_ RTC_GUARDED_BY(acm_crit_sect_);
+ ChangeLogger bitrate_logger_ RTC_GUARDED_BY(acm_mutex_);
// Current encoder stack, provided by a call to RegisterEncoder.
- std::unique_ptr<AudioEncoder> encoder_stack_ RTC_GUARDED_BY(acm_crit_sect_);
+ std::unique_ptr<AudioEncoder> encoder_stack_ RTC_GUARDED_BY(acm_mutex_);
// This is to keep track of CN instances where we can send DTMFs.
- uint8_t previous_pltype_ RTC_GUARDED_BY(acm_crit_sect_);
+ uint8_t previous_pltype_ RTC_GUARDED_BY(acm_mutex_);
- bool receiver_initialized_ RTC_GUARDED_BY(acm_crit_sect_);
+ bool receiver_initialized_ RTC_GUARDED_BY(acm_mutex_);
- AudioFrame preprocess_frame_ RTC_GUARDED_BY(acm_crit_sect_);
- bool first_10ms_data_ RTC_GUARDED_BY(acm_crit_sect_);
+ AudioFrame preprocess_frame_ RTC_GUARDED_BY(acm_mutex_);
+ bool first_10ms_data_ RTC_GUARDED_BY(acm_mutex_);
- bool first_frame_ RTC_GUARDED_BY(acm_crit_sect_);
- uint32_t last_timestamp_ RTC_GUARDED_BY(acm_crit_sect_);
- uint32_t last_rtp_timestamp_ RTC_GUARDED_BY(acm_crit_sect_);
+ bool first_frame_ RTC_GUARDED_BY(acm_mutex_);
+ uint32_t last_timestamp_ RTC_GUARDED_BY(acm_mutex_);
+ uint32_t last_rtp_timestamp_ RTC_GUARDED_BY(acm_mutex_);
- rtc::CriticalSection callback_crit_sect_;
+ Mutex callback_mutex_;
AudioPacketizationCallback* packetization_callback_
- RTC_GUARDED_BY(callback_crit_sect_);
+ RTC_GUARDED_BY(callback_mutex_);
int codec_histogram_bins_log_[static_cast<size_t>(
AudioEncoder::CodecType::kMaxLoggedAudioCodecTypes)];
@@ -298,7 +298,7 @@
}
{
- rtc::CritScope lock(&callback_crit_sect_);
+ MutexLock lock(&callback_mutex_);
if (packetization_callback_) {
packetization_callback_->SendData(
frame_type, encoded_info.payload_type, encoded_info.encoded_timestamp,
@@ -316,7 +316,7 @@
void AudioCodingModuleImpl::ModifyEncoder(
rtc::FunctionView<void(std::unique_ptr<AudioEncoder>*)> modifier) {
- rtc::CritScope lock(&acm_crit_sect_);
+ MutexLock lock(&acm_mutex_);
modifier(&encoder_stack_);
}
@@ -324,14 +324,14 @@
// the encoded buffers.
int AudioCodingModuleImpl::RegisterTransportCallback(
AudioPacketizationCallback* transport) {
- rtc::CritScope lock(&callback_crit_sect_);
+ MutexLock lock(&callback_mutex_);
packetization_callback_ = transport;
return 0;
}
// Add 10MS of raw (PCM) audio data to the encoder.
int AudioCodingModuleImpl::Add10MsData(const AudioFrame& audio_frame) {
- rtc::CritScope lock(&acm_crit_sect_);
+ MutexLock lock(&acm_mutex_);
int r = Add10MsDataInternal(audio_frame, &input_data_);
// TODO(bugs.webrtc.org/10739): add dcheck that
// |audio_frame.absolute_capture_timestamp_ms()| always has a value.
@@ -519,7 +519,7 @@
//
int AudioCodingModuleImpl::SetPacketLossRate(int loss_rate) {
- rtc::CritScope lock(&acm_crit_sect_);
+ MutexLock lock(&acm_mutex_);
if (HaveValidEncoder("SetPacketLossRate")) {
encoder_stack_->OnReceivedUplinkPacketLossFraction(loss_rate / 100.0);
}
@@ -531,7 +531,7 @@
//
int AudioCodingModuleImpl::InitializeReceiver() {
- rtc::CritScope lock(&acm_crit_sect_);
+ MutexLock lock(&acm_mutex_);
return InitializeReceiverSafe();
}
@@ -550,7 +550,7 @@
void AudioCodingModuleImpl::SetReceiveCodecs(
const std::map<int, SdpAudioFormat>& codecs) {
- rtc::CritScope lock(&acm_crit_sect_);
+ MutexLock lock(&acm_mutex_);
receiver_.SetCodecs(codecs);
}
@@ -597,7 +597,7 @@
}
ANAStats AudioCodingModuleImpl::GetANAStats() const {
- rtc::CritScope lock(&acm_crit_sect_);
+ MutexLock lock(&acm_mutex_);
if (encoder_stack_)
return encoder_stack_->GetANAStats();
// If no encoder is set, return default stats.