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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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62
modules/audio_coding/acm2/audio_coding_module.cc
View File

@ -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 @@ class AudioCodingModuleImpl final : public AudioCodingModule {
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 @@ class AudioCodingModuleImpl final : public AudioCodingModule {
};
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 @@ class AudioCodingModuleImpl final : public AudioCodingModule {
// 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 @@ int32_t AudioCodingModuleImpl::Encode(
}
{
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 @@ int32_t AudioCodingModuleImpl::Encode(
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 @@ void AudioCodingModuleImpl::ModifyEncoder(
// 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::PreprocessToAddData(const AudioFrame& in_frame,
//
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::SetPacketLossRate(int loss_rate) {
//
int AudioCodingModuleImpl::InitializeReceiver() {
rtc::CritScope lock(&acm_crit_sect_);
MutexLock lock(&acm_mutex_);
return InitializeReceiverSafe();
}
@ -550,7 +550,7 @@ int AudioCodingModuleImpl::InitializeReceiverSafe() {
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 @@ bool AudioCodingModuleImpl::HaveValidEncoder(const char* caller_name) const {
}
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.