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/acm_receiver.cc b/modules/audio_coding/acm2/acm_receiver.cc
index 29eff19..33142c7 100644
--- a/modules/audio_coding/acm2/acm_receiver.cc
+++ b/modules/audio_coding/acm2/acm_receiver.cc
@@ -86,7 +86,7 @@
}
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 @@
}
{
- 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 @@
/*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 @@
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::RemoveAllCodecs() {
- rtc::CritScope lock(&crit_sect_);
+ MutexLock lock(&mutex_);
neteq_->RemoveAllPayloadTypes();
last_decoder_ = absl::nullopt;
}
@@ -236,7 +236,7 @@
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 @@
void AcmReceiver::GetDecodingCallStatistics(
AudioDecodingCallStats* stats) const {
- rtc::CritScope lock(&crit_sect_);
+ MutexLock lock(&mutex_);
*stats = call_stats_.GetDecodingStatistics();
}
diff --git a/modules/audio_coding/acm2/acm_receiver.h b/modules/audio_coding/acm2/acm_receiver.h
index 1512656..d451a94 100644
--- a/modules/audio_coding/acm2/acm_receiver.h
+++ b/modules/audio_coding/acm2/acm_receiver.h
@@ -26,7 +26,7 @@
#include "modules/audio_coding/acm2/acm_resampler.h"
#include "modules/audio_coding/acm2/call_statistics.h"
#include "modules/audio_coding/include/audio_coding_module.h"
-#include "rtc_base/critical_section.h"
+#include "rtc_base/synchronization/mutex.h"
#include "rtc_base/thread_annotations.h"
namespace webrtc {
@@ -212,14 +212,14 @@
uint32_t NowInTimestamp(int decoder_sampling_rate) const;
- rtc::CriticalSection crit_sect_;
- absl::optional<DecoderInfo> last_decoder_ RTC_GUARDED_BY(crit_sect_);
- ACMResampler resampler_ RTC_GUARDED_BY(crit_sect_);
- std::unique_ptr<int16_t[]> last_audio_buffer_ RTC_GUARDED_BY(crit_sect_);
- CallStatistics call_stats_ RTC_GUARDED_BY(crit_sect_);
+ mutable Mutex mutex_;
+ absl::optional<DecoderInfo> last_decoder_ RTC_GUARDED_BY(mutex_);
+ ACMResampler resampler_ RTC_GUARDED_BY(mutex_);
+ std::unique_ptr<int16_t[]> last_audio_buffer_ RTC_GUARDED_BY(mutex_);
+ CallStatistics call_stats_ RTC_GUARDED_BY(mutex_);
const std::unique_ptr<NetEq> neteq_; // NetEq is thread-safe; no lock needed.
Clock* const clock_;
- bool resampled_last_output_frame_ RTC_GUARDED_BY(crit_sect_);
+ bool resampled_last_output_frame_ RTC_GUARDED_BY(mutex_);
};
} // namespace acm2
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.
diff --git a/modules/audio_coding/acm2/audio_coding_module_unittest.cc b/modules/audio_coding/acm2/audio_coding_module_unittest.cc
index b53d456..efd7b04 100644
--- a/modules/audio_coding/acm2/audio_coding_module_unittest.cc
+++ b/modules/audio_coding/acm2/audio_coding_module_unittest.cc
@@ -39,12 +39,12 @@
#include "modules/audio_coding/neteq/tools/output_wav_file.h"
#include "modules/audio_coding/neteq/tools/packet.h"
#include "modules/audio_coding/neteq/tools/rtp_file_source.h"
-#include "rtc_base/critical_section.h"
#include "rtc_base/event.h"
#include "rtc_base/message_digest.h"
#include "rtc_base/numerics/safe_conversions.h"
#include "rtc_base/platform_thread.h"
#include "rtc_base/ref_counted_object.h"
+#include "rtc_base/synchronization/mutex.h"
#include "rtc_base/system/arch.h"
#include "rtc_base/thread_annotations.h"
#include "system_wrappers/include/clock.h"
@@ -113,7 +113,7 @@
const uint8_t* payload_data,
size_t payload_len_bytes,
int64_t absolute_capture_timestamp_ms) override {
- rtc::CritScope lock(&crit_sect_);
+ MutexLock lock(&mutex_);
++num_calls_;
last_frame_type_ = frame_type;
last_payload_type_ = payload_type;
@@ -123,42 +123,42 @@
}
int num_calls() const {
- rtc::CritScope lock(&crit_sect_);
+ MutexLock lock(&mutex_);
return num_calls_;
}
int last_payload_len_bytes() const {
- rtc::CritScope lock(&crit_sect_);
+ MutexLock lock(&mutex_);
return rtc::checked_cast<int>(last_payload_vec_.size());
}
AudioFrameType last_frame_type() const {
- rtc::CritScope lock(&crit_sect_);
+ MutexLock lock(&mutex_);
return last_frame_type_;
}
int last_payload_type() const {
- rtc::CritScope lock(&crit_sect_);
+ MutexLock lock(&mutex_);
return last_payload_type_;
}
uint32_t last_timestamp() const {
- rtc::CritScope lock(&crit_sect_);
+ MutexLock lock(&mutex_);
return last_timestamp_;
}
void SwapBuffers(std::vector<uint8_t>* payload) {
- rtc::CritScope lock(&crit_sect_);
+ MutexLock lock(&mutex_);
last_payload_vec_.swap(*payload);
}
private:
- int num_calls_ RTC_GUARDED_BY(crit_sect_);
- AudioFrameType last_frame_type_ RTC_GUARDED_BY(crit_sect_);
- int last_payload_type_ RTC_GUARDED_BY(crit_sect_);
- uint32_t last_timestamp_ RTC_GUARDED_BY(crit_sect_);
- std::vector<uint8_t> last_payload_vec_ RTC_GUARDED_BY(crit_sect_);
- rtc::CriticalSection crit_sect_;
+ int num_calls_ RTC_GUARDED_BY(mutex_);
+ AudioFrameType last_frame_type_ RTC_GUARDED_BY(mutex_);
+ int last_payload_type_ RTC_GUARDED_BY(mutex_);
+ uint32_t last_timestamp_ RTC_GUARDED_BY(mutex_);
+ std::vector<uint8_t> last_payload_vec_ RTC_GUARDED_BY(mutex_);
+ mutable Mutex mutex_;
};
class AudioCodingModuleTestOldApi : public ::testing::Test {
@@ -472,7 +472,7 @@
virtual bool TestDone() {
if (packet_cb_.num_calls() > kNumPackets) {
- rtc::CritScope lock(&crit_sect_);
+ MutexLock lock(&mutex_);
if (pull_audio_count_ > kNumPullCalls) {
// Both conditions for completion are met. End the test.
return true;
@@ -515,7 +515,7 @@
void CbInsertPacketImpl() {
SleepMs(1);
{
- rtc::CritScope lock(&crit_sect_);
+ MutexLock lock(&mutex_);
if (clock_->TimeInMilliseconds() < next_insert_packet_time_ms_) {
return;
}
@@ -537,7 +537,7 @@
void CbPullAudioImpl() {
SleepMs(1);
{
- rtc::CritScope lock(&crit_sect_);
+ MutexLock lock(&mutex_);
// Don't let the insert thread fall behind.
if (next_insert_packet_time_ms_ < clock_->TimeInMilliseconds()) {
return;
@@ -558,9 +558,9 @@
rtc::Event test_complete_;
int send_count_;
int insert_packet_count_;
- int pull_audio_count_ RTC_GUARDED_BY(crit_sect_);
- rtc::CriticalSection crit_sect_;
- int64_t next_insert_packet_time_ms_ RTC_GUARDED_BY(crit_sect_);
+ int pull_audio_count_ RTC_GUARDED_BY(mutex_);
+ Mutex mutex_;
+ int64_t next_insert_packet_time_ms_ RTC_GUARDED_BY(mutex_);
std::unique_ptr<SimulatedClock> fake_clock_;
};
@@ -658,7 +658,7 @@
// run).
bool TestDone() override {
if (packet_cb_.num_calls() > kNumPackets) {
- rtc::CritScope lock(&crit_sect_);
+ MutexLock lock(&mutex_);
if (pull_audio_count_ > kNumPullCalls) {
// Both conditions for completion are met. End the test.
return true;
@@ -758,7 +758,7 @@
rtc::Buffer encoded;
AudioEncoder::EncodedInfo info;
{
- rtc::CritScope lock(&crit_sect_);
+ MutexLock lock(&mutex_);
if (clock_->TimeInMilliseconds() < next_insert_packet_time_ms_) {
return true;
}
@@ -812,7 +812,7 @@
// End the test early if a fatal failure (ASSERT_*) has occurred.
test_complete_.Set();
}
- rtc::CritScope lock(&crit_sect_);
+ MutexLock lock(&mutex_);
if (!codec_registered_ &&
receive_packet_count_ > kRegisterAfterNumPackets) {
// Register the iSAC encoder.
@@ -831,10 +831,10 @@
std::atomic<bool> quit_;
rtc::Event test_complete_;
- rtc::CriticalSection crit_sect_;
- bool codec_registered_ RTC_GUARDED_BY(crit_sect_);
- int receive_packet_count_ RTC_GUARDED_BY(crit_sect_);
- int64_t next_insert_packet_time_ms_ RTC_GUARDED_BY(crit_sect_);
+ Mutex mutex_;
+ bool codec_registered_ RTC_GUARDED_BY(mutex_);
+ int receive_packet_count_ RTC_GUARDED_BY(mutex_);
+ int64_t next_insert_packet_time_ms_ RTC_GUARDED_BY(mutex_);
std::unique_ptr<AudioEncoderIsacFloatImpl> isac_encoder_;
std::unique_ptr<SimulatedClock> fake_clock_;
test::AudioLoop audio_loop_;