| Index: webrtc/modules/audio_coding/acm2/audio_coding_module.cc
|
| diff --git a/webrtc/modules/audio_coding/acm2/audio_coding_module.cc b/webrtc/modules/audio_coding/acm2/audio_coding_module.cc
|
| index 14958181859501459181f56380e79bf885e40376..f7607c6521c2f434effb96ad1eaec1393bb2bc6d 100644
|
| --- a/webrtc/modules/audio_coding/acm2/audio_coding_module.cc
|
| +++ b/webrtc/modules/audio_coding/acm2/audio_coding_module.cc
|
| @@ -234,17 +234,17 @@ class AudioCodingModuleImpl final : public AudioCodingModule {
|
| int RegisterReceiveCodecUnlocked(
|
| const CodecInst& codec,
|
| rtc::FunctionView<std::unique_ptr<AudioDecoder>()> isac_factory)
|
| - EXCLUSIVE_LOCKS_REQUIRED(acm_crit_sect_);
|
| + RTC_EXCLUSIVE_LOCKS_REQUIRED(acm_crit_sect_);
|
|
|
| int Add10MsDataInternal(const AudioFrame& audio_frame, InputData* input_data)
|
| - EXCLUSIVE_LOCKS_REQUIRED(acm_crit_sect_);
|
| + RTC_EXCLUSIVE_LOCKS_REQUIRED(acm_crit_sect_);
|
| int Encode(const InputData& input_data)
|
| - EXCLUSIVE_LOCKS_REQUIRED(acm_crit_sect_);
|
| + RTC_EXCLUSIVE_LOCKS_REQUIRED(acm_crit_sect_);
|
|
|
| - int InitializeReceiverSafe() EXCLUSIVE_LOCKS_REQUIRED(acm_crit_sect_);
|
| + int InitializeReceiverSafe() RTC_EXCLUSIVE_LOCKS_REQUIRED(acm_crit_sect_);
|
|
|
| bool HaveValidEncoder(const char* caller_name) const
|
| - EXCLUSIVE_LOCKS_REQUIRED(acm_crit_sect_);
|
| + RTC_EXCLUSIVE_LOCKS_REQUIRED(acm_crit_sect_);
|
|
|
| // Preprocessing of input audio, including resampling and down-mixing if
|
| // required, before pushing audio into encoder's buffer.
|
| @@ -259,33 +259,36 @@ class AudioCodingModuleImpl final : public AudioCodingModule {
|
| // 0: otherwise.
|
| int PreprocessToAddData(const AudioFrame& in_frame,
|
| const AudioFrame** ptr_out)
|
| - EXCLUSIVE_LOCKS_REQUIRED(acm_crit_sect_);
|
| + RTC_EXCLUSIVE_LOCKS_REQUIRED(acm_crit_sect_);
|
|
|
| // 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_ GUARDED_BY(acm_crit_sect_);
|
| + rtc::Buffer encode_buffer_ RTC_GUARDED_BY(acm_crit_sect_);
|
| int id_; // TODO(henrik.lundin) Make const.
|
| - uint32_t expected_codec_ts_ GUARDED_BY(acm_crit_sect_);
|
| - uint32_t expected_in_ts_ GUARDED_BY(acm_crit_sect_);
|
| - acm2::ACMResampler resampler_ 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_);
|
| acm2::AcmReceiver receiver_; // AcmReceiver has it's own internal lock.
|
| - ChangeLogger bitrate_logger_ GUARDED_BY(acm_crit_sect_);
|
| + ChangeLogger bitrate_logger_ RTC_GUARDED_BY(acm_crit_sect_);
|
|
|
| - std::unique_ptr<EncoderFactory> encoder_factory_ GUARDED_BY(acm_crit_sect_);
|
| + std::unique_ptr<EncoderFactory> encoder_factory_
|
| + RTC_GUARDED_BY(acm_crit_sect_);
|
|
|
| // Current encoder stack, either obtained from
|
| // encoder_factory_->rent_a_codec.RentEncoderStack or provided by a call to
|
| // RegisterEncoder.
|
| - std::unique_ptr<AudioEncoder> encoder_stack_ GUARDED_BY(acm_crit_sect_);
|
| + std::unique_ptr<AudioEncoder> encoder_stack_ RTC_GUARDED_BY(acm_crit_sect_);
|
|
|
| - std::unique_ptr<AudioDecoder> isac_decoder_16k_ GUARDED_BY(acm_crit_sect_);
|
| - std::unique_ptr<AudioDecoder> isac_decoder_32k_ GUARDED_BY(acm_crit_sect_);
|
| + std::unique_ptr<AudioDecoder> isac_decoder_16k_
|
| + RTC_GUARDED_BY(acm_crit_sect_);
|
| + std::unique_ptr<AudioDecoder> isac_decoder_32k_
|
| + RTC_GUARDED_BY(acm_crit_sect_);
|
|
|
| // This is to keep track of CN instances where we can send DTMFs.
|
| - uint8_t previous_pltype_ GUARDED_BY(acm_crit_sect_);
|
| + uint8_t previous_pltype_ RTC_GUARDED_BY(acm_crit_sect_);
|
|
|
| // Used when payloads are pushed into ACM without any RTP info
|
| // One example is when pre-encoded bit-stream is pushed from
|
| @@ -295,19 +298,19 @@ class AudioCodingModuleImpl final : public AudioCodingModule {
|
| // be used in other methods, locks need to be taken.
|
| std::unique_ptr<WebRtcRTPHeader> aux_rtp_header_;
|
|
|
| - bool receiver_initialized_ GUARDED_BY(acm_crit_sect_);
|
| + bool receiver_initialized_ RTC_GUARDED_BY(acm_crit_sect_);
|
|
|
| - AudioFrame preprocess_frame_ GUARDED_BY(acm_crit_sect_);
|
| - bool first_10ms_data_ GUARDED_BY(acm_crit_sect_);
|
| + AudioFrame preprocess_frame_ RTC_GUARDED_BY(acm_crit_sect_);
|
| + bool first_10ms_data_ RTC_GUARDED_BY(acm_crit_sect_);
|
|
|
| - bool first_frame_ GUARDED_BY(acm_crit_sect_);
|
| - uint32_t last_timestamp_ GUARDED_BY(acm_crit_sect_);
|
| - uint32_t last_rtp_timestamp_ GUARDED_BY(acm_crit_sect_);
|
| + 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_);
|
|
|
| rtc::CriticalSection callback_crit_sect_;
|
| AudioPacketizationCallback* packetization_callback_
|
| - GUARDED_BY(callback_crit_sect_);
|
| - ACMVADCallback* vad_callback_ GUARDED_BY(callback_crit_sect_);
|
| + RTC_GUARDED_BY(callback_crit_sect_);
|
| + ACMVADCallback* vad_callback_ RTC_GUARDED_BY(callback_crit_sect_);
|
|
|
| int codec_histogram_bins_log_[static_cast<size_t>(
|
| AudioEncoder::CodecType::kMaxLoggedAudioCodecTypes)];
|
|
|