Switch CriticalSectionWrapper->rtc::CriticalSection in modules/audio_coding.
This is a part of cleaning up CriticalSectionWrapper in general.
BUG=
R=henrik.lundin@webrtc.org
Review URL: https://codereview.webrtc.org/1610073003 .
Cr-Commit-Position: refs/heads/master@{#11319}
diff --git a/webrtc/modules/audio_coding/acm2/acm_receiver.cc b/webrtc/modules/audio_coding/acm2/acm_receiver.cc
index f45d5d3..53ec884 100644
--- a/webrtc/modules/audio_coding/acm2/acm_receiver.cc
+++ b/webrtc/modules/audio_coding/acm2/acm_receiver.cc
@@ -25,7 +25,6 @@
#include "webrtc/modules/audio_coding/acm2/call_statistics.h"
#include "webrtc/modules/audio_coding/neteq/include/neteq.h"
#include "webrtc/system_wrappers/include/clock.h"
-#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
#include "webrtc/system_wrappers/include/tick_util.h"
#include "webrtc/system_wrappers/include/trace.h"
@@ -119,8 +118,7 @@
} // namespace
AcmReceiver::AcmReceiver(const AudioCodingModule::Config& config)
- : crit_sect_(CriticalSectionWrapper::CreateCriticalSection()),
- id_(config.id),
+ : id_(config.id),
last_audio_decoder_(nullptr),
previous_audio_activity_(AudioFrame::kVadPassive),
audio_buffer_(new int16_t[AudioFrame::kMaxDataSizeSamples]),
@@ -157,7 +155,7 @@
}
rtc::Optional<int> AcmReceiver::last_packet_sample_rate_hz() const {
- CriticalSectionScoped lock(crit_sect_.get());
+ rtc::CritScope lock(&crit_sect_);
return last_packet_sample_rate_hz_;
}
@@ -171,7 +169,7 @@
const RTPHeader* header = &rtp_header.header; // Just a shorthand.
{
- CriticalSectionScoped lock(crit_sect_.get());
+ rtc::CritScope lock(&crit_sect_);
const Decoder* decoder = RtpHeaderToDecoder(*header, incoming_payload[0]);
if (!decoder) {
@@ -216,7 +214,7 @@
size_t num_channels;
// Accessing members, take the lock.
- CriticalSectionScoped lock(crit_sect_.get());
+ rtc::CritScope lock(&crit_sect_);
// Always write the output to |audio_buffer_| first.
if (neteq_->GetAudio(AudioFrame::kMaxDataSizeSamples,
@@ -317,7 +315,7 @@
return *ned;
}();
- CriticalSectionScoped lock(crit_sect_.get());
+ rtc::CritScope lock(&crit_sect_);
// The corresponding NetEq decoder ID.
// If this codec has been registered before.
@@ -366,13 +364,13 @@
void AcmReceiver::EnableVad() {
neteq_->EnableVad();
- CriticalSectionScoped lock(crit_sect_.get());
+ rtc::CritScope lock(&crit_sect_);
vad_enabled_ = true;
}
void AcmReceiver::DisableVad() {
neteq_->DisableVad();
- CriticalSectionScoped lock(crit_sect_.get());
+ rtc::CritScope lock(&crit_sect_);
vad_enabled_ = false;
}
@@ -384,7 +382,7 @@
// many as it can.
int AcmReceiver::RemoveAllCodecs() {
int ret_val = 0;
- CriticalSectionScoped lock(crit_sect_.get());
+ rtc::CritScope lock(&crit_sect_);
for (auto it = decoders_.begin(); it != decoders_.end(); ) {
auto cur = it;
++it; // it will be valid even if we erase cur
@@ -404,7 +402,7 @@
}
int AcmReceiver::RemoveCodec(uint8_t payload_type) {
- CriticalSectionScoped lock(crit_sect_.get());
+ rtc::CritScope lock(&crit_sect_);
auto it = decoders_.find(payload_type);
if (it == decoders_.end()) { // Such a payload-type is not registered.
return 0;
@@ -422,7 +420,7 @@
}
void AcmReceiver::set_id(int id) {
- CriticalSectionScoped lock(crit_sect_.get());
+ rtc::CritScope lock(&crit_sect_);
id_ = id;
}
@@ -431,7 +429,7 @@
}
int AcmReceiver::LastAudioCodec(CodecInst* codec) const {
- CriticalSectionScoped lock(crit_sect_.get());
+ rtc::CritScope lock(&crit_sect_);
if (!last_audio_decoder_) {
return -1;
}
@@ -468,7 +466,7 @@
int AcmReceiver::DecoderByPayloadType(uint8_t payload_type,
CodecInst* codec) const {
- CriticalSectionScoped lock(crit_sect_.get());
+ rtc::CritScope lock(&crit_sect_);
auto it = decoders_.find(payload_type);
if (it == decoders_.end()) {
LOG(LERROR) << "AcmReceiver::DecoderByPayloadType "
@@ -532,7 +530,7 @@
void AcmReceiver::GetDecodingCallStatistics(
AudioDecodingCallStats* stats) const {
- CriticalSectionScoped lock(crit_sect_.get());
+ rtc::CritScope lock(&crit_sect_);
*stats = call_stats_.GetDecodingStatistics();
}
diff --git a/webrtc/modules/audio_coding/acm2/acm_receiver.h b/webrtc/modules/audio_coding/acm2/acm_receiver.h
index b150612..826cb45 100644
--- a/webrtc/modules/audio_coding/acm2/acm_receiver.h
+++ b/webrtc/modules/audio_coding/acm2/acm_receiver.h
@@ -16,15 +16,16 @@
#include <vector>
#include "webrtc/base/array_view.h"
+#include "webrtc/base/criticalsection.h"
#include "webrtc/base/optional.h"
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/base/thread_annotations.h"
#include "webrtc/common_audio/vad/include/webrtc_vad.h"
#include "webrtc/engine_configurations.h"
-#include "webrtc/modules/audio_coding/include/audio_coding_module.h"
#include "webrtc/modules/audio_coding/acm2/acm_resampler.h"
#include "webrtc/modules/audio_coding/acm2/call_statistics.h"
#include "webrtc/modules/audio_coding/acm2/initial_delay_manager.h"
+#include "webrtc/modules/audio_coding/include/audio_coding_module.h"
#include "webrtc/modules/audio_coding/neteq/include/neteq.h"
#include "webrtc/modules/include/module_common_types.h"
#include "webrtc/typedefs.h"
@@ -32,7 +33,6 @@
namespace webrtc {
struct CodecInst;
-class CriticalSectionWrapper;
class NetEq;
namespace acm2 {
@@ -281,7 +281,7 @@
uint32_t NowInTimestamp(int decoder_sampling_rate) const;
- rtc::scoped_ptr<CriticalSectionWrapper> crit_sect_;
+ mutable rtc::CriticalSection crit_sect_;
int id_; // TODO(henrik.lundin) Make const.
const Decoder* last_audio_decoder_ GUARDED_BY(crit_sect_);
AudioFrame::VADActivity previous_audio_activity_ GUARDED_BY(crit_sect_);
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 ac302f0..d0e02ea 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 @@
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 @@
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 @@
}
{
- 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 @@
// 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 @@
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 @@
// 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 @@
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 @@
}
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 @@
// 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 @@
// 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 @@
//
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 @@
//
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::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 @@
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 @@
// 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::InitializeReceiver() {
- CriticalSectionScoped lock(acm_crit_sect_.get());
+ rtc::CritScope lock(&acm_crit_sect_);
return InitializeReceiverSafe();
}
@@ -587,7 +584,7 @@
// 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 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 @@
// 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::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::SetOpusApplication(OpusApplicationMode application) {
- CriticalSectionScoped lock(acm_crit_sect_.get());
+ rtc::CritScope lock(&acm_crit_sect_);
if (!HaveValidEncoder("SetOpusApplication")) {
return -1;
}
@@ -761,7 +758,7 @@
// 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::EnableOpusDtx() {
- CriticalSectionScoped lock(acm_crit_sect_.get());
+ rtc::CritScope lock(&acm_crit_sect_);
if (!HaveValidEncoder("EnableOpusDtx")) {
return -1;
}
@@ -778,7 +775,7 @@
}
int AudioCodingModuleImpl::DisableOpusDtx() {
- CriticalSectionScoped lock(acm_crit_sect_.get());
+ rtc::CritScope lock(&acm_crit_sect_);
if (!HaveValidEncoder("DisableOpusDtx")) {
return -1;
}
diff --git a/webrtc/modules/audio_coding/acm2/audio_coding_module_impl.h b/webrtc/modules/audio_coding/acm2/audio_coding_module_impl.h
index 926671f..19776d1 100644
--- a/webrtc/modules/audio_coding/acm2/audio_coding_module_impl.h
+++ b/webrtc/modules/audio_coding/acm2/audio_coding_module_impl.h
@@ -15,6 +15,7 @@
#include <vector>
#include "webrtc/base/buffer.h"
+#include "webrtc/base/criticalsection.h"
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/base/thread_annotations.h"
#include "webrtc/common_types.h"
@@ -25,7 +26,6 @@
namespace webrtc {
-class CriticalSectionWrapper;
class AudioCodingImpl;
namespace acm2 {
@@ -240,7 +240,7 @@
// to |index|.
int UpdateUponReceivingCodec(int index);
- const rtc::scoped_ptr<CriticalSectionWrapper> acm_crit_sect_;
+ mutable rtc::CriticalSection acm_crit_sect_;
rtc::Buffer encode_buffer_ GUARDED_BY(acm_crit_sect_);
int id_; // TODO(henrik.lundin) Make const.
uint32_t expected_codec_ts_ GUARDED_BY(acm_crit_sect_);
@@ -271,7 +271,7 @@
uint32_t last_timestamp_ GUARDED_BY(acm_crit_sect_);
uint32_t last_rtp_timestamp_ GUARDED_BY(acm_crit_sect_);
- const rtc::scoped_ptr<CriticalSectionWrapper> callback_crit_sect_;
+ mutable rtc::CriticalSection callback_crit_sect_;
AudioPacketizationCallback* packetization_callback_
GUARDED_BY(callback_crit_sect_);
ACMVADCallback* vad_callback_ GUARDED_BY(callback_crit_sect_);
diff --git a/webrtc/modules/audio_coding/acm2/audio_coding_module_unittest_oldapi.cc b/webrtc/modules/audio_coding/acm2/audio_coding_module_unittest_oldapi.cc
index 6f82a96..c738d0f 100644
--- a/webrtc/modules/audio_coding/acm2/audio_coding_module_unittest_oldapi.cc
+++ b/webrtc/modules/audio_coding/acm2/audio_coding_module_unittest_oldapi.cc
@@ -13,17 +13,18 @@
#include <vector>
#include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/base/criticalsection.h"
#include "webrtc/base/md5digest.h"
#include "webrtc/base/platform_thread.h"
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/base/thread_annotations.h"
+#include "webrtc/modules/audio_coding/acm2/acm_receive_test_oldapi.h"
+#include "webrtc/modules/audio_coding/acm2/acm_send_test_oldapi.h"
#include "webrtc/modules/audio_coding/codecs/audio_encoder.h"
#include "webrtc/modules/audio_coding/codecs/g711/audio_decoder_pcm.h"
#include "webrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.h"
#include "webrtc/modules/audio_coding/codecs/isac/main/include/audio_encoder_isac.h"
#include "webrtc/modules/audio_coding/codecs/mock/mock_audio_encoder.h"
-#include "webrtc/modules/audio_coding/acm2/acm_receive_test_oldapi.h"
-#include "webrtc/modules/audio_coding/acm2/acm_send_test_oldapi.h"
#include "webrtc/modules/audio_coding/include/audio_coding_module.h"
#include "webrtc/modules/audio_coding/include/audio_coding_module_typedefs.h"
#include "webrtc/modules/audio_coding/neteq/audio_decoder_impl.h"
@@ -37,7 +38,6 @@
#include "webrtc/modules/audio_coding/neteq/tools/rtp_file_source.h"
#include "webrtc/modules/include/module_common_types.h"
#include "webrtc/system_wrappers/include/clock.h"
-#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
#include "webrtc/system_wrappers/include/event_wrapper.h"
#include "webrtc/system_wrappers/include/sleep.h"
#include "webrtc/test/testsupport/fileutils.h"
@@ -94,8 +94,7 @@
: num_calls_(0),
last_frame_type_(kEmptyFrame),
last_payload_type_(-1),
- last_timestamp_(0),
- crit_sect_(CriticalSectionWrapper::CreateCriticalSection()) {}
+ last_timestamp_(0) {}
int32_t SendData(FrameType frame_type,
uint8_t payload_type,
@@ -103,7 +102,7 @@
const uint8_t* payload_data,
size_t payload_len_bytes,
const RTPFragmentationHeader* fragmentation) override {
- CriticalSectionScoped lock(crit_sect_.get());
+ rtc::CritScope lock(&crit_sect_);
++num_calls_;
last_frame_type_ = frame_type;
last_payload_type_ = payload_type;
@@ -113,32 +112,32 @@
}
int num_calls() const {
- CriticalSectionScoped lock(crit_sect_.get());
+ rtc::CritScope lock(&crit_sect_);
return num_calls_;
}
int last_payload_len_bytes() const {
- CriticalSectionScoped lock(crit_sect_.get());
+ rtc::CritScope lock(&crit_sect_);
return last_payload_vec_.size();
}
FrameType last_frame_type() const {
- CriticalSectionScoped lock(crit_sect_.get());
+ rtc::CritScope lock(&crit_sect_);
return last_frame_type_;
}
int last_payload_type() const {
- CriticalSectionScoped lock(crit_sect_.get());
+ rtc::CritScope lock(&crit_sect_);
return last_payload_type_;
}
uint32_t last_timestamp() const {
- CriticalSectionScoped lock(crit_sect_.get());
+ rtc::CritScope lock(&crit_sect_);
return last_timestamp_;
}
void SwapBuffers(std::vector<uint8_t>* payload) {
- CriticalSectionScoped lock(crit_sect_.get());
+ rtc::CritScope lock(&crit_sect_);
last_payload_vec_.swap(*payload);
}
@@ -148,7 +147,7 @@
int last_payload_type_ GUARDED_BY(crit_sect_);
uint32_t last_timestamp_ GUARDED_BY(crit_sect_);
std::vector<uint8_t> last_payload_vec_ GUARDED_BY(crit_sect_);
- const rtc::scoped_ptr<CriticalSectionWrapper> crit_sect_;
+ mutable rtc::CriticalSection crit_sect_;
};
class AudioCodingModuleTestOldApi : public ::testing::Test {
@@ -469,7 +468,6 @@
send_count_(0),
insert_packet_count_(0),
pull_audio_count_(0),
- crit_sect_(CriticalSectionWrapper::CreateCriticalSection()),
next_insert_packet_time_ms_(0),
fake_clock_(new SimulatedClock(0)) {
clock_ = fake_clock_.get();
@@ -503,7 +501,7 @@
virtual bool TestDone() {
if (packet_cb_.num_calls() > kNumPackets) {
- CriticalSectionScoped lock(crit_sect_.get());
+ rtc::CritScope lock(&crit_sect_);
if (pull_audio_count_ > kNumPullCalls) {
// Both conditions for completion are met. End the test.
return true;
@@ -541,7 +539,7 @@
bool CbInsertPacketImpl() {
SleepMs(1);
{
- CriticalSectionScoped lock(crit_sect_.get());
+ rtc::CritScope lock(&crit_sect_);
if (clock_->TimeInMilliseconds() < next_insert_packet_time_ms_) {
return true;
}
@@ -561,7 +559,7 @@
bool CbPullAudioImpl() {
SleepMs(1);
{
- CriticalSectionScoped lock(crit_sect_.get());
+ rtc::CritScope lock(&crit_sect_);
// Don't let the insert thread fall behind.
if (next_insert_packet_time_ms_ < clock_->TimeInMilliseconds()) {
return true;
@@ -581,7 +579,7 @@
int send_count_;
int insert_packet_count_;
int pull_audio_count_ GUARDED_BY(crit_sect_);
- const rtc::scoped_ptr<CriticalSectionWrapper> crit_sect_;
+ mutable rtc::CriticalSection crit_sect_;
int64_t next_insert_packet_time_ms_ GUARDED_BY(crit_sect_);
rtc::scoped_ptr<SimulatedClock> fake_clock_;
};
@@ -681,7 +679,7 @@
// run).
virtual bool TestDone() {
if (packet_cb_.num_calls() > kNumPackets) {
- CriticalSectionScoped lock(crit_sect_.get());
+ rtc::CritScope lock(&crit_sect_);
if (pull_audio_count_ > kNumPullCalls) {
// Both conditions for completion are met. End the test.
return true;
@@ -720,7 +718,6 @@
this,
"codec_registration"),
test_complete_(EventWrapper::Create()),
- crit_sect_(CriticalSectionWrapper::CreateCriticalSection()),
codec_registered_(false),
receive_packet_count_(0),
next_insert_packet_time_ms_(0),
@@ -781,7 +778,7 @@
rtc::scoped_ptr<uint8_t[]> encoded(new uint8_t[max_encoded_bytes]);
AudioEncoder::EncodedInfo info;
{
- CriticalSectionScoped lock(crit_sect_.get());
+ rtc::CritScope lock(&crit_sect_);
if (clock_->TimeInMilliseconds() < next_insert_packet_time_ms_) {
return true;
}
@@ -829,7 +826,7 @@
// End the test early if a fatal failure (ASSERT_*) has occurred.
test_complete_->Set();
}
- CriticalSectionScoped lock(crit_sect_.get());
+ rtc::CritScope lock(&crit_sect_);
if (!codec_registered_ &&
receive_packet_count_ > kRegisterAfterNumPackets) {
// Register the iSAC encoder.
@@ -845,7 +842,7 @@
rtc::PlatformThread receive_thread_;
rtc::PlatformThread codec_registration_thread_;
const rtc::scoped_ptr<EventWrapper> test_complete_;
- const rtc::scoped_ptr<CriticalSectionWrapper> crit_sect_;
+ mutable rtc::CriticalSection crit_sect_;
bool codec_registered_ GUARDED_BY(crit_sect_);
int receive_packet_count_ GUARDED_BY(crit_sect_);
int64_t next_insert_packet_time_ms_ GUARDED_BY(crit_sect_);