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Unified Diff: webrtc/modules/audio_coding/acm2/audio_coding_module_impl.cc

Issue 1610073003: Switch CriticalSectionWrapper->rtc::CriticalSection in modules/audio_coding. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Created 4 years, 11 months ago
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Index: webrtc/modules/audio_coding/acm2/audio_coding_module_impl.cc
diff --git a/webrtc/modules/audio_coding/acm2/audio_coding_module_impl.cc b/webrtc/modules/audio_coding/acm2/audio_coding_module_impl.cc
index ac302f0fe3c304f035c54ed2f20062ea1b139040..d0e02eaf6c465b134a1c52ee2ec712bfd362aa62 100644
--- a/webrtc/modules/audio_coding/acm2/audio_coding_module_impl.cc
+++ b/webrtc/modules/audio_coding/acm2/audio_coding_module_impl.cc
@@ -21,7 +21,6 @@
#include "webrtc/modules/audio_coding/acm2/acm_common_defs.h"
#include "webrtc/modules/audio_coding/acm2/acm_resampler.h"
#include "webrtc/modules/audio_coding/acm2/call_statistics.h"
-#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
#include "webrtc/system_wrappers/include/logging.h"
#include "webrtc/system_wrappers/include/metrics.h"
#include "webrtc/system_wrappers/include/rw_lock_wrapper.h"
@@ -103,8 +102,7 @@ void AudioCodingModuleImpl::ChangeLogger::MaybeLog(int value) {
AudioCodingModuleImpl::AudioCodingModuleImpl(
const AudioCodingModule::Config& config)
- : acm_crit_sect_(CriticalSectionWrapper::CreateCriticalSection()),
- id_(config.id),
+ : id_(config.id),
expected_codec_ts_(0xD87F3F9F),
expected_in_ts_(0xD87F3F9F),
receiver_(config),
@@ -113,7 +111,6 @@ AudioCodingModuleImpl::AudioCodingModuleImpl(
receiver_initialized_(false),
first_10ms_data_(false),
first_frame_(true),
- callback_crit_sect_(CriticalSectionWrapper::CreateCriticalSection()),
packetization_callback_(NULL),
vad_callback_(NULL) {
if (InitializeReceiverSafe() < 0) {
@@ -173,7 +170,7 @@ int32_t AudioCodingModuleImpl::Encode(const InputData& input_data) {
}
{
- CriticalSectionScoped lock(callback_crit_sect_.get());
+ rtc::CritScope lock(&callback_crit_sect_);
if (packetization_callback_) {
packetization_callback_->SendData(
frame_type, encoded_info.payload_type, encoded_info.encoded_timestamp,
@@ -197,7 +194,7 @@ int32_t AudioCodingModuleImpl::Encode(const InputData& input_data) {
// Can be called multiple times for Codec, CNG, RED.
int AudioCodingModuleImpl::RegisterSendCodec(const CodecInst& send_codec) {
- CriticalSectionScoped lock(acm_crit_sect_.get());
+ rtc::CritScope lock(&acm_crit_sect_);
if (!codec_manager_.RegisterEncoder(send_codec)) {
return -1;
}
@@ -217,7 +214,7 @@ int AudioCodingModuleImpl::RegisterSendCodec(const CodecInst& send_codec) {
void AudioCodingModuleImpl::RegisterExternalSendCodec(
AudioEncoder* external_speech_encoder) {
- CriticalSectionScoped lock(acm_crit_sect_.get());
+ rtc::CritScope lock(&acm_crit_sect_);
auto* sp = codec_manager_.GetStackParams();
sp->speech_encoder = external_speech_encoder;
rent_a_codec_.RentEncoderStack(sp);
@@ -225,7 +222,7 @@ void AudioCodingModuleImpl::RegisterExternalSendCodec(
// Get current send codec.
rtc::Optional<CodecInst> AudioCodingModuleImpl::SendCodec() const {
- CriticalSectionScoped lock(acm_crit_sect_.get());
+ rtc::CritScope lock(&acm_crit_sect_);
auto* ci = codec_manager_.GetCodecInst();
if (ci) {
return rtc::Optional<CodecInst>(*ci);
@@ -241,7 +238,7 @@ rtc::Optional<CodecInst> AudioCodingModuleImpl::SendCodec() const {
int AudioCodingModuleImpl::SendFrequency() const {
WEBRTC_TRACE(webrtc::kTraceStream, webrtc::kTraceAudioCoding, id_,
"SendFrequency()");
- CriticalSectionScoped lock(acm_crit_sect_.get());
+ rtc::CritScope lock(&acm_crit_sect_);
const auto* enc = rent_a_codec_.GetEncoderStack();
if (!enc) {
@@ -254,7 +251,7 @@ int AudioCodingModuleImpl::SendFrequency() const {
}
void AudioCodingModuleImpl::SetBitRate(int bitrate_bps) {
- CriticalSectionScoped lock(acm_crit_sect_.get());
+ rtc::CritScope lock(&acm_crit_sect_);
auto* enc = rent_a_codec_.GetEncoderStack();
if (enc) {
enc->SetTargetBitrate(bitrate_bps);
@@ -265,7 +262,7 @@ void AudioCodingModuleImpl::SetBitRate(int bitrate_bps) {
// the encoded buffers.
int AudioCodingModuleImpl::RegisterTransportCallback(
AudioPacketizationCallback* transport) {
- CriticalSectionScoped lock(callback_crit_sect_.get());
+ rtc::CritScope lock(&callback_crit_sect_);
packetization_callback_ = transport;
return 0;
}
@@ -273,7 +270,7 @@ int AudioCodingModuleImpl::RegisterTransportCallback(
// Add 10MS of raw (PCM) audio data to the encoder.
int AudioCodingModuleImpl::Add10MsData(const AudioFrame& audio_frame) {
InputData input_data;
- CriticalSectionScoped lock(acm_crit_sect_.get());
+ rtc::CritScope lock(&acm_crit_sect_);
int r = Add10MsDataInternal(audio_frame, &input_data);
return r < 0 ? r : Encode(input_data);
}
@@ -445,14 +442,14 @@ int AudioCodingModuleImpl::PreprocessToAddData(const AudioFrame& in_frame,
//
bool AudioCodingModuleImpl::REDStatus() const {
- CriticalSectionScoped lock(acm_crit_sect_.get());
+ rtc::CritScope lock(&acm_crit_sect_);
return codec_manager_.GetStackParams()->use_red;
}
// Configure RED status i.e on/off.
int AudioCodingModuleImpl::SetREDStatus(bool enable_red) {
#ifdef WEBRTC_CODEC_RED
- CriticalSectionScoped lock(acm_crit_sect_.get());
+ rtc::CritScope lock(&acm_crit_sect_);
if (!codec_manager_.SetCopyRed(enable_red)) {
return -1;
}
@@ -472,12 +469,12 @@ int AudioCodingModuleImpl::SetREDStatus(bool enable_red) {
//
bool AudioCodingModuleImpl::CodecFEC() const {
- CriticalSectionScoped lock(acm_crit_sect_.get());
+ rtc::CritScope lock(&acm_crit_sect_);
return codec_manager_.GetStackParams()->use_codec_fec;
}
int AudioCodingModuleImpl::SetCodecFEC(bool enable_codec_fec) {
- CriticalSectionScoped lock(acm_crit_sect_.get());
+ rtc::CritScope lock(&acm_crit_sect_);
if (!codec_manager_.SetCodecFEC(enable_codec_fec)) {
return -1;
}
@@ -493,7 +490,7 @@ int AudioCodingModuleImpl::SetCodecFEC(bool enable_codec_fec) {
}
int AudioCodingModuleImpl::SetPacketLossRate(int loss_rate) {
- CriticalSectionScoped lock(acm_crit_sect_.get());
+ rtc::CritScope lock(&acm_crit_sect_);
if (HaveValidEncoder("SetPacketLossRate")) {
rent_a_codec_.GetEncoderStack()->SetProjectedPacketLossRate(loss_rate /
100.0);
@@ -509,7 +506,7 @@ int AudioCodingModuleImpl::SetVAD(bool enable_dtx,
ACMVADMode mode) {
// Note: |enable_vad| is not used; VAD is enabled based on the DTX setting.
RTC_DCHECK_EQ(enable_dtx, enable_vad);
- CriticalSectionScoped lock(acm_crit_sect_.get());
+ rtc::CritScope lock(&acm_crit_sect_);
if (!codec_manager_.SetVAD(enable_dtx, mode)) {
return -1;
}
@@ -522,7 +519,7 @@ int AudioCodingModuleImpl::SetVAD(bool enable_dtx,
// Get VAD/DTX settings.
int AudioCodingModuleImpl::VAD(bool* dtx_enabled, bool* vad_enabled,
ACMVADMode* mode) const {
- CriticalSectionScoped lock(acm_crit_sect_.get());
+ rtc::CritScope lock(&acm_crit_sect_);
const auto* sp = codec_manager_.GetStackParams();
*dtx_enabled = *vad_enabled = sp->use_cng;
*mode = sp->vad_mode;
@@ -534,7 +531,7 @@ int AudioCodingModuleImpl::VAD(bool* dtx_enabled, bool* vad_enabled,
//
int AudioCodingModuleImpl::InitializeReceiver() {
- CriticalSectionScoped lock(acm_crit_sect_.get());
+ rtc::CritScope lock(&acm_crit_sect_);
return InitializeReceiverSafe();
}
@@ -587,7 +584,7 @@ int AudioCodingModuleImpl::PlayoutFrequency() const {
// Register possible receive codecs, can be called multiple times,
// for codecs, CNG (NB, WB and SWB), DTMF, RED.
int AudioCodingModuleImpl::RegisterReceiveCodec(const CodecInst& codec) {
- CriticalSectionScoped lock(acm_crit_sect_.get());
+ rtc::CritScope lock(&acm_crit_sect_);
RTC_DCHECK(receiver_initialized_);
if (codec.channels > 2) {
LOG_F(LS_ERROR) << "Unsupported number of channels: " << codec.channels;
@@ -625,7 +622,7 @@ int AudioCodingModuleImpl::RegisterExternalReceiveCodec(
int sample_rate_hz,
int num_channels,
const std::string& name) {
- CriticalSectionScoped lock(acm_crit_sect_.get());
+ rtc::CritScope lock(&acm_crit_sect_);
RTC_DCHECK(receiver_initialized_);
if (num_channels > 2 || num_channels < 0) {
LOG_F(LS_ERROR) << "Unsupported number of channels: " << num_channels;
@@ -645,7 +642,7 @@ int AudioCodingModuleImpl::RegisterExternalReceiveCodec(
// Get current received codec.
int AudioCodingModuleImpl::ReceiveCodec(CodecInst* current_codec) const {
- CriticalSectionScoped lock(acm_crit_sect_.get());
+ rtc::CritScope lock(&acm_crit_sect_);
return receiver_.LastAudioCodec(current_codec);
}
@@ -705,7 +702,7 @@ int AudioCodingModuleImpl::GetNetworkStatistics(NetworkStatistics* statistics) {
int AudioCodingModuleImpl::RegisterVADCallback(ACMVADCallback* vad_callback) {
WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceAudioCoding, id_,
"RegisterVADCallback()");
- CriticalSectionScoped lock(callback_crit_sect_.get());
+ rtc::CritScope lock(&callback_crit_sect_);
vad_callback_ = vad_callback;
return 0;
}
@@ -740,7 +737,7 @@ int AudioCodingModuleImpl::IncomingPayload(const uint8_t* incoming_payload,
}
int AudioCodingModuleImpl::SetOpusApplication(OpusApplicationMode application) {
- CriticalSectionScoped lock(acm_crit_sect_.get());
+ rtc::CritScope lock(&acm_crit_sect_);
if (!HaveValidEncoder("SetOpusApplication")) {
return -1;
}
@@ -761,7 +758,7 @@ int AudioCodingModuleImpl::SetOpusApplication(OpusApplicationMode application) {
// Informs Opus encoder of the maximum playback rate the receiver will render.
int AudioCodingModuleImpl::SetOpusMaxPlaybackRate(int frequency_hz) {
- CriticalSectionScoped lock(acm_crit_sect_.get());
+ rtc::CritScope lock(&acm_crit_sect_);
if (!HaveValidEncoder("SetOpusMaxPlaybackRate")) {
return -1;
}
@@ -770,7 +767,7 @@ int AudioCodingModuleImpl::SetOpusMaxPlaybackRate(int frequency_hz) {
}
int AudioCodingModuleImpl::EnableOpusDtx() {
- CriticalSectionScoped lock(acm_crit_sect_.get());
+ rtc::CritScope lock(&acm_crit_sect_);
if (!HaveValidEncoder("EnableOpusDtx")) {
return -1;
}
@@ -778,7 +775,7 @@ int AudioCodingModuleImpl::EnableOpusDtx() {
}
int AudioCodingModuleImpl::DisableOpusDtx() {
- CriticalSectionScoped lock(acm_crit_sect_.get());
+ rtc::CritScope lock(&acm_crit_sect_);
if (!HaveValidEncoder("DisableOpusDtx")) {
return -1;
}
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