Switch CriticalSectionWrapper->rtc::CriticalSection in modules/audio_coding.

webrtc/modules/audio_coding/acm2/audio_coding_module_impl.cc

 /*
  *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
 #include "webrtc/modules/audio_coding/acm2/audio_coding_module_impl.h"
 
 #include <assert.h>
 #include <stdlib.h>
 #include <vector>
 
 #include "webrtc/base/checks.h"
 #include "webrtc/base/safe_conversions.h"
 #include "webrtc/engine_configurations.h"
 #include "webrtc/modules/audio_coding/include/audio_coding_module_typedefs.h"
 #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"
 #include "webrtc/system_wrappers/include/trace.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
 
 namespace acm2 {
 
@@ skipping 61 matching lines @@
 void AudioCodingModuleImpl::ChangeLogger::MaybeLog(int value) {
   if (value != last_value_ || first_time_) {
     first_time_ = false;
     last_value_ = value;
     RTC_HISTOGRAM_COUNTS_SPARSE_100(histogram_name_, 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),
       bitrate_logger_("WebRTC.Audio.TargetBitrateInKbps"),
       previous_pltype_(255),
       receiver_initialized_(false),
       first_10ms_data_(false),
       first_frame_(true),
-      callback_crit_sect_(CriticalSectionWrapper::CreateCriticalSection()),
       packetization_callback_(NULL),
       vad_callback_(NULL) {
   if (InitializeReceiverSafe() < 0) {
     WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_,
                  "Cannot initialize receiver");
   }
   WEBRTC_TRACE(webrtc::kTraceMemory, webrtc::kTraceAudioCoding, id_, "Created");
 }
 
 AudioCodingModuleImpl::~AudioCodingModuleImpl() = default;
@@ skipping 39 matching lines @@
   FrameType frame_type;
   if (encode_buffer_.size() == 0 && encoded_info.send_even_if_empty) {
     frame_type = kEmptyFrame;
     encoded_info.payload_type = previous_pltype;
   } else {
     RTC_DCHECK_GT(encode_buffer_.size(), 0u);
     frame_type = encoded_info.speech ? kAudioFrameSpeech : kAudioFrameCN;
   }
 
   {
-    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,
           encode_buffer_.data(), encode_buffer_.size(),
           my_fragmentation.fragmentationVectorSize > 0 ? &my_fragmentation
                                                        : nullptr);
     }
 
     if (vad_callback_) {
       // Callback with VAD decision.
       vad_callback_->InFrameType(frame_type);
     }
   }
   previous_pltype_ = encoded_info.payload_type;
   return static_cast<int32_t>(encode_buffer_.size());
 }
 
 /////////////////////////////////////////
 //   Sender
 //
 
 // 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;
   }
   auto* sp = codec_manager_.GetStackParams();
   if (!sp->speech_encoder && codec_manager_.GetCodecInst()) {
     // We have no speech encoder, but we have a specification for making one.
     AudioEncoder* enc =
         rent_a_codec_.RentEncoder(*codec_manager_.GetCodecInst());
     if (!enc)
       return -1;
     sp->speech_encoder = enc;
   }
   if (sp->speech_encoder)
     rent_a_codec_.RentEncoderStack(sp);
   return 0;
 }
 
 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);
 }
 
 // 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);
   }
   auto* enc = codec_manager_.GetStackParams()->speech_encoder;
   if (enc) {
     return rtc::Optional<CodecInst>(CodecManager::ForgeCodecInst(enc));
   }
   return rtc::Optional<CodecInst>();
 }
 
 // Get current send frequency.
 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) {
     WEBRTC_TRACE(webrtc::kTraceStream, webrtc::kTraceAudioCoding, id_,
                  "SendFrequency Failed, no codec is registered");
     return -1;
   }
 
   return enc->SampleRateHz();
 }
 
 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);
   }
 }
 
 // Register a transport callback which will be called to deliver
 // 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;
 }
 
 // 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);
 }
 
 int AudioCodingModuleImpl::Add10MsDataInternal(const AudioFrame& audio_frame,
                                                InputData* input_data) {
   if (audio_frame.samples_per_channel_ == 0) {
     assert(false);
     WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_,
                  "Cannot Add 10 ms audio, payload length is zero");
@@ skipping 151 matching lines @@
   expected_in_ts_ += static_cast<uint32_t>(in_frame.samples_per_channel_);
 
   return 0;
 }
 
 /////////////////////////////////////////
 //   (RED) Redundant Coding
 //
 
 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;
   }
   auto* sp = codec_manager_.GetStackParams();
   if (sp->speech_encoder)
     rent_a_codec_.RentEncoderStack(sp);
   return 0;
 #else
   WEBRTC_TRACE(webrtc::kTraceWarning, webrtc::kTraceAudioCoding, id_,
                "  WEBRTC_CODEC_RED is undefined");
   return -1;
 #endif
 }
 
 /////////////////////////////////////////
 //   (FEC) Forward Error Correction (codec internal)
 //
 
 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;
   }
   auto* sp = codec_manager_.GetStackParams();
   if (sp->speech_encoder)
     rent_a_codec_.RentEncoderStack(sp);
   if (enable_codec_fec) {
     return sp->use_codec_fec ? 0 : -1;
   } else {
     RTC_DCHECK(!sp->use_codec_fec);
     return 0;
   }
 }
 
 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);
   }
   return 0;
 }
 
 /////////////////////////////////////////
 //   (VAD) Voice Activity Detection
 //
 int AudioCodingModuleImpl::SetVAD(bool enable_dtx,
                                   bool enable_vad,
                                   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;
   }
   auto* sp = codec_manager_.GetStackParams();
   if (sp->speech_encoder)
     rent_a_codec_.RentEncoderStack(sp);
   return 0;
 }
 
 // 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;
   return 0;
 }
 
 /////////////////////////////////////////
 //   Receiver
 //
 
 int AudioCodingModuleImpl::InitializeReceiver() {
-  CriticalSectionScoped lock(acm_crit_sect_.get());
+  rtc::CritScope lock(&acm_crit_sect_);
   return InitializeReceiverSafe();
 }
 
 // Initialize receiver, resets codec database etc.
 int AudioCodingModuleImpl::InitializeReceiverSafe() {
   // If the receiver is already initialized then we want to destroy any
   // existing decoders. After a call to this function, we should have a clean
   // start-up.
   if (receiver_initialized_) {
     if (receiver_.RemoveAllCodecs() < 0)
@@ skipping 32 matching lines @@
 // Get current playout frequency.
 int AudioCodingModuleImpl::PlayoutFrequency() const {
   WEBRTC_TRACE(webrtc::kTraceStream, webrtc::kTraceAudioCoding, id_,
                "PlayoutFrequency()");
   return receiver_.last_output_sample_rate_hz();
 }
 
 // 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;
     return -1;
   }
 
   auto codec_id =
       RentACodec::CodecIdByParams(codec.plname, codec.plfreq, codec.channels);
   if (!codec_id) {
     LOG_F(LS_ERROR) << "Wrong codec params to be registered as receive codec";
@@ skipping 17 matching lines @@
                                               : nullptr,
       codec.plname);
 }
 
 int AudioCodingModuleImpl::RegisterExternalReceiveCodec(
     int rtp_payload_type,
     AudioDecoder* external_decoder,
     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;
     return -1;
   }
 
   // Check if the payload-type is valid.
   if (!RentACodec::IsPayloadTypeValid(rtp_payload_type)) {
     LOG_F(LS_ERROR) << "Invalid payload-type " << rtp_payload_type
                     << " for external decoder.";
     return -1;
   }
 
   return receiver_.AddCodec(-1 /* external */, rtp_payload_type, num_channels,
                             sample_rate_hz, external_decoder, name);
 }
 
 // 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);
 }
 
 // Incoming packet from network parsed and ready for decode.
 int AudioCodingModuleImpl::IncomingPacket(const uint8_t* incoming_payload,
                                           const size_t payload_length,
                                           const WebRtcRTPHeader& rtp_header) {
   return receiver_.InsertPacket(
       rtp_header,
       rtc::ArrayView<const uint8_t>(incoming_payload, payload_length));
@@ skipping 39 matching lines @@
 // TODO(turajs) change the return value to void. Also change the corresponding
 // NetEq function.
 int AudioCodingModuleImpl::GetNetworkStatistics(NetworkStatistics* statistics) {
   receiver_.GetNetworkStatistics(statistics);
   return 0;
 }
 
 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;
 }
 
 // TODO(kwiberg): Remove this method, and have callers call IncomingPacket
 // instead. The translation logic and state belong with them, not with
 // AudioCodingModuleImpl.
 int AudioCodingModuleImpl::IncomingPayload(const uint8_t* incoming_payload,
                                            size_t payload_length,
                                            uint8_t payload_type,
@@ skipping 14 matching lines @@
   }
 
   aux_rtp_header_->header.timestamp = timestamp;
   IncomingPacket(incoming_payload, payload_length, *aux_rtp_header_);
   // Get ready for the next payload.
   aux_rtp_header_->header.sequenceNumber++;
   return 0;
 }
 
 int AudioCodingModuleImpl::SetOpusApplication(OpusApplicationMode application) {
-  CriticalSectionScoped lock(acm_crit_sect_.get());
+  rtc::CritScope lock(&acm_crit_sect_);
   if (!HaveValidEncoder("SetOpusApplication")) {
     return -1;
   }
   AudioEncoder::Application app;
   switch (application) {
     case kVoip:
       app = AudioEncoder::Application::kSpeech;
       break;
     case kAudio:
       app = AudioEncoder::Application::kAudio;
       break;
     default:
       FATAL();
       return 0;
   }
   return rent_a_codec_.GetEncoderStack()->SetApplication(app) ? 0 : -1;
 }
 
 // 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;
   }
   rent_a_codec_.GetEncoderStack()->SetMaxPlaybackRate(frequency_hz);
   return 0;
 }
 
 int AudioCodingModuleImpl::EnableOpusDtx() {
-  CriticalSectionScoped lock(acm_crit_sect_.get());
+  rtc::CritScope lock(&acm_crit_sect_);
   if (!HaveValidEncoder("EnableOpusDtx")) {
     return -1;
   }
   return rent_a_codec_.GetEncoderStack()->SetDtx(true) ? 0 : -1;
 }
 
 int AudioCodingModuleImpl::DisableOpusDtx() {
-  CriticalSectionScoped lock(acm_crit_sect_.get());
+  rtc::CritScope lock(&acm_crit_sect_);
   if (!HaveValidEncoder("DisableOpusDtx")) {
     return -1;
   }
   return rent_a_codec_.GetEncoderStack()->SetDtx(false) ? 0 : -1;
 }
 
 int AudioCodingModuleImpl::PlayoutTimestamp(uint32_t* timestamp) {
   return receiver_.GetPlayoutTimestamp(timestamp) ? 0 : -1;
 }
 
@@ skipping 27 matching lines @@
   return receiver_.LeastRequiredDelayMs();
 }
 
 void AudioCodingModuleImpl::GetDecodingCallStatistics(
       AudioDecodingCallStats* call_stats) const {
   receiver_.GetDecodingCallStatistics(call_stats);
 }
 
 }  // namespace acm2
 }  // namespace webrtc