diff options
Diffstat (limited to 'chromium/third_party/webrtc/modules/audio_coding/main/acm2/acm_isac.cc')
| -rw-r--r-- | chromium/third_party/webrtc/modules/audio_coding/main/acm2/acm_isac.cc | 269 |
1 files changed, 136 insertions, 133 deletions
diff --git a/chromium/third_party/webrtc/modules/audio_coding/main/acm2/acm_isac.cc b/chromium/third_party/webrtc/modules/audio_coding/main/acm2/acm_isac.cc index e27284212fd..9fbcdd4cd8b 100644 --- a/chromium/third_party/webrtc/modules/audio_coding/main/acm2/acm_isac.cc +++ b/chromium/third_party/webrtc/modules/audio_coding/main/acm2/acm_isac.cc @@ -14,7 +14,8 @@ #include "webrtc/modules/audio_coding/main/interface/audio_coding_module_typedefs.h" #include "webrtc/modules/audio_coding/main/acm2/acm_codec_database.h" #include "webrtc/modules/audio_coding/main/acm2/acm_common_defs.h" -#include "webrtc/modules/audio_coding/neteq4/interface/audio_decoder.h" +#include "webrtc/modules/audio_coding/neteq/interface/audio_decoder.h" +#include "webrtc/system_wrappers/interface/critical_section_wrapper.h" #include "webrtc/system_wrappers/interface/trace.h" #ifdef WEBRTC_CODEC_ISAC @@ -59,14 +60,15 @@ static const int32_t kIsacRatesSwb[NR_ISAC_BANDWIDTHS] = { #if (!defined(WEBRTC_CODEC_ISAC) && !defined(WEBRTC_CODEC_ISACFX)) ACMISAC::ACMISAC(int16_t /* codec_id */) - : codec_inst_ptr_(NULL), + : codec_inst_crit_sect_(CriticalSectionWrapper::CreateCriticalSection()), + codec_inst_ptr_(NULL), is_enc_initialized_(false), isac_coding_mode_(CHANNEL_INDEPENDENT), enforce_frame_size_(false), isac_currentBN_(32000), samples_in10MsAudio_(160), // Initiates to 16 kHz mode. - audio_decoder_(NULL), - decoder_initialized_(false) {} + decoder_initialized_(false) { +} ACMISAC::~ACMISAC() { return; @@ -261,81 +263,14 @@ static uint16_t ACMISACFixGetDecSampRate(ACM_ISAC_STRUCT* /* inst */) { #endif -// Decoder class to be injected into NetEq. -class AcmAudioDecoderIsac : public AudioDecoder { - public: - AcmAudioDecoderIsac(int codec_id, void* state) - : AudioDecoder(ACMCodecDB::neteq_decoders_[codec_id]) { - state_ = state; - } - - // ACMISAC is the owner of the object where |state_| is pointing to. - // Therefore, it should not be deleted in this destructor. - virtual ~AcmAudioDecoderIsac() {} - - virtual int Decode(const uint8_t* encoded, size_t encoded_len, - int16_t* decoded, SpeechType* speech_type) { - int16_t temp_type; - int ret = ACM_ISAC_DECODE_B(static_cast<ACM_ISAC_STRUCT*>(state_), - reinterpret_cast<const uint16_t*>(encoded), - static_cast<int16_t>(encoded_len), decoded, - &temp_type); - *speech_type = ConvertSpeechType(temp_type); - return ret; - } - - virtual bool HasDecodePlc() const { return true; } - - virtual int DecodePlc(int num_frames, int16_t* decoded) { - return ACM_ISAC_DECODEPLC(static_cast<ACM_ISAC_STRUCT*>(state_), - decoded, static_cast<int16_t>(num_frames)); - } - - virtual int Init() { - return 0; // We expect that the initialized instance is injected in the - // constructor. - } - - virtual int IncomingPacket(const uint8_t* payload, - size_t payload_len, - uint16_t rtp_sequence_number, - uint32_t rtp_timestamp, - uint32_t arrival_timestamp) { - return ACM_ISAC_DECODE_BWE(static_cast<ACM_ISAC_STRUCT*>(state_), - reinterpret_cast<const uint16_t*>(payload), - static_cast<uint32_t>(payload_len), - rtp_sequence_number, - rtp_timestamp, - arrival_timestamp); - } - - virtual int DecodeRedundant(const uint8_t* encoded, - size_t encoded_len, int16_t* decoded, - SpeechType* speech_type) { - int16_t temp_type = 1; // Default is speech. - int16_t ret = ACM_ISAC_DECODERCU(static_cast<ACM_ISAC_STRUCT*>(state_), - reinterpret_cast<const uint16_t*>(encoded), - static_cast<int16_t>(encoded_len), decoded, - &temp_type); - *speech_type = ConvertSpeechType(temp_type); - return ret; - } - - virtual int ErrorCode() { - return ACM_ISAC_GETERRORCODE(static_cast<ACM_ISAC_STRUCT*>(state_)); - } - - private: - DISALLOW_COPY_AND_ASSIGN(AcmAudioDecoderIsac); -}; - ACMISAC::ACMISAC(int16_t codec_id) - : is_enc_initialized_(false), + : AudioDecoder(ACMCodecDB::neteq_decoders_[codec_id]), + codec_inst_crit_sect_(CriticalSectionWrapper::CreateCriticalSection()), + is_enc_initialized_(false), isac_coding_mode_(CHANNEL_INDEPENDENT), enforce_frame_size_(false), isac_current_bn_(32000), samples_in_10ms_audio_(160), // Initiates to 16 kHz mode. - audio_decoder_(NULL), decoder_initialized_(false) { codec_id_ = codec_id; @@ -345,14 +280,10 @@ ACMISAC::ACMISAC(int16_t codec_id) return; } codec_inst_ptr_->inst = NULL; + state_ = codec_inst_ptr_; } ACMISAC::~ACMISAC() { - if (audio_decoder_ != NULL) { - delete audio_decoder_; - audio_decoder_ = NULL; - } - if (codec_inst_ptr_ != NULL) { if (codec_inst_ptr_->inst != NULL) { ACM_ISAC_FREE(codec_inst_ptr_->inst); @@ -364,6 +295,34 @@ ACMISAC::~ACMISAC() { return; } +int16_t ACMISAC::InternalInitDecoder(WebRtcACMCodecParams* codec_params) { + // set decoder sampling frequency. + if (codec_params->codec_inst.plfreq == 32000 || + codec_params->codec_inst.plfreq == 48000) { + UpdateDecoderSampFreq(ACMCodecDB::kISACSWB); + } else { + UpdateDecoderSampFreq(ACMCodecDB::kISAC); + } + + // in a one-way communication we may never register send-codec. + // However we like that the BWE to work properly so it has to + // be initialized. The BWE is initialized when iSAC encoder is initialized. + // Therefore, we need this. + if (!encoder_initialized_) { + // Since we don't require a valid rate or a valid packet size when + // initializing the decoder, we set valid values before initializing encoder + codec_params->codec_inst.rate = kIsacWbDefaultRate; + codec_params->codec_inst.pacsize = kIsacPacSize960; + if (InternalInitEncoder(codec_params) < 0) { + return -1; + } + encoder_initialized_ = true; + } + + CriticalSectionScoped lock(codec_inst_crit_sect_.get()); + return ACM_ISAC_DECODERINIT(codec_inst_ptr_->inst); +} + ACMGenericCodec* ACMISAC::CreateInstance(void) { return NULL; } int16_t ACMISAC::InternalEncode(uint8_t* bitstream, @@ -375,6 +334,7 @@ int16_t ACMISAC::InternalEncode(uint8_t* bitstream, // at the first 10ms pushed in to iSAC if the bit-rate is low, this is // sort of a bug in iSAC. to address this we treat iSAC as the // following. + CriticalSectionScoped lock(codec_inst_crit_sect_.get()); if (codec_inst_ptr_ == NULL) { return -1; } @@ -428,6 +388,7 @@ int16_t ACMISAC::InternalInitEncoder(WebRtcACMCodecParams* codec_params) { if (UpdateEncoderSampFreq((uint16_t)codec_params->codec_inst.plfreq) < 0) { return -1; } + CriticalSectionScoped lock(codec_inst_crit_sect_.get()); if (ACM_ISAC_ENCODERINIT(codec_inst_ptr_->inst, isac_coding_mode_) < 0) { return -1; } @@ -450,38 +411,8 @@ int16_t ACMISAC::InternalInitEncoder(WebRtcACMCodecParams* codec_params) { return 0; } -int16_t ACMISAC::InternalInitDecoder(WebRtcACMCodecParams* codec_params) { - if (codec_inst_ptr_ == NULL) { - return -1; - } - - // set decoder sampling frequency. - if (codec_params->codec_inst.plfreq == 32000 || - codec_params->codec_inst.plfreq == 48000) { - UpdateDecoderSampFreq(ACMCodecDB::kISACSWB); - } else { - UpdateDecoderSampFreq(ACMCodecDB::kISAC); - } - - // in a one-way communication we may never register send-codec. - // However we like that the BWE to work properly so it has to - // be initialized. The BWE is initialized when iSAC encoder is initialized. - // Therefore, we need this. - if (!encoder_initialized_) { - // Since we don't require a valid rate or a valid packet size when - // initializing the decoder, we set valid values before initializing encoder - codec_params->codec_inst.rate = kIsacWbDefaultRate; - codec_params->codec_inst.pacsize = kIsacPacSize960; - if (InternalInitEncoder(codec_params) < 0) { - return -1; - } - encoder_initialized_ = true; - } - - return ACM_ISAC_DECODERINIT(codec_inst_ptr_->inst); -} - int16_t ACMISAC::InternalCreateEncoder() { + CriticalSectionScoped lock(codec_inst_crit_sect_.get()); if (codec_inst_ptr_ == NULL) { return -1; } @@ -493,19 +424,6 @@ int16_t ACMISAC::InternalCreateEncoder() { return status; } -void ACMISAC::DestructEncoderSafe() { - // codec with shared instance cannot delete. - encoder_initialized_ = false; - return; -} - -void ACMISAC::InternalDestructEncoderInst(void* ptr_inst) { - if (ptr_inst != NULL) { - ACM_ISAC_FREE(static_cast<ACM_ISAC_STRUCT *>(ptr_inst)); - } - return; -} - int16_t ACMISAC::Transcode(uint8_t* bitstream, int16_t* bitstream_len_byte, int16_t q_bwe, @@ -513,6 +431,7 @@ int16_t ACMISAC::Transcode(uint8_t* bitstream, bool is_red) { int16_t jitter_info = 0; // transcode from a higher rate to lower rate sanity check + CriticalSectionScoped lock(codec_inst_crit_sect_.get()); if (codec_inst_ptr_ == NULL) { return -1; } @@ -530,7 +449,27 @@ int16_t ACMISAC::Transcode(uint8_t* bitstream, } } +void ACMISAC::UpdateFrameLen() { + CriticalSectionScoped lock(codec_inst_crit_sect_.get()); + frame_len_smpl_ = ACM_ISAC_GETNEWFRAMELEN(codec_inst_ptr_->inst); + encoder_params_.codec_inst.pacsize = frame_len_smpl_; +} + +void ACMISAC::DestructEncoderSafe() { + // codec with shared instance cannot delete. + encoder_initialized_ = false; + return; +} + +void ACMISAC::InternalDestructEncoderInst(void* ptr_inst) { + if (ptr_inst != NULL) { + ACM_ISAC_FREE(static_cast<ACM_ISAC_STRUCT *>(ptr_inst)); + } + return; +} + int16_t ACMISAC::SetBitRateSafe(int32_t bit_rate) { + CriticalSectionScoped lock(codec_inst_crit_sect_.get()); if (codec_inst_ptr_ == NULL) { return -1; } @@ -594,6 +533,7 @@ int32_t ACMISAC::GetEstimatedBandwidthSafe() { int samp_rate; // Get bandwidth information + CriticalSectionScoped lock(codec_inst_crit_sect_.get()); ACM_ISAC_GETSENDBWE(codec_inst_ptr_->inst, &bandwidth_index, &delay_index); // Validy check of index @@ -615,6 +555,7 @@ int32_t ACMISAC::SetEstimatedBandwidthSafe(int32_t estimated_bandwidth) { int16_t bandwidth_index; // Check sample frequency and choose appropriate table + CriticalSectionScoped lock(codec_inst_crit_sect_.get()); samp_rate = ACM_ISAC_GETENCSAMPRATE(codec_inst_ptr_->inst); if (samp_rate == 16000) { @@ -657,6 +598,7 @@ int32_t ACMISAC::GetRedPayloadSafe( return -1; #else uint8_t* red_payload, int16_t* payload_bytes) { + CriticalSectionScoped lock(codec_inst_crit_sect_.get()); int16_t bytes = WebRtcIsac_GetRedPayload( codec_inst_ptr_->inst, reinterpret_cast<int16_t*>(red_payload)); @@ -672,6 +614,7 @@ int16_t ACMISAC::UpdateDecoderSampFreq( #ifdef WEBRTC_CODEC_ISAC int16_t codec_id) { // The decoder supports only wideband and super-wideband. + CriticalSectionScoped lock(codec_inst_crit_sect_.get()); if (ACMCodecDB::kISAC == codec_id) { return WebRtcIsac_SetDecSampRate(codec_inst_ptr_->inst, 16000); } else if (ACMCodecDB::kISACSWB == codec_id || @@ -700,6 +643,7 @@ int16_t ACMISAC::UpdateEncoderSampFreq( in_audio_ix_read_ = 0; in_audio_ix_write_ = 0; in_timestamp_ix_write_ = 0; + CriticalSectionScoped lock(codec_inst_crit_sect_.get()); if (WebRtcIsac_SetEncSampRate(codec_inst_ptr_->inst, encoder_samp_freq_hz) < 0) { return -1; @@ -718,6 +662,7 @@ int16_t ACMISAC::UpdateEncoderSampFreq( } int16_t ACMISAC::EncoderSampFreq(uint16_t* samp_freq_hz) { + CriticalSectionScoped lock(codec_inst_crit_sect_.get()); *samp_freq_hz = ACM_ISAC_GETENCSAMPRATE(codec_inst_ptr_->inst); return 0; } @@ -730,6 +675,7 @@ int32_t ACMISAC::ConfigISACBandwidthEstimator( { uint16_t samp_freq_hz; EncoderSampFreq(&samp_freq_hz); + CriticalSectionScoped lock(codec_inst_crit_sect_.get()); // TODO(turajs): at 32kHz we hardcode calling with 30ms and enforce // the frame-size otherwise we might get error. Revise if // control-bwe is changed. @@ -749,26 +695,25 @@ int32_t ACMISAC::ConfigISACBandwidthEstimator( return -1; } UpdateFrameLen(); + CriticalSectionScoped lock(codec_inst_crit_sect_.get()); ACM_ISAC_GETSENDBITRATE(codec_inst_ptr_->inst, &isac_current_bn_); return 0; } int32_t ACMISAC::SetISACMaxPayloadSize(const uint16_t max_payload_len_bytes) { + CriticalSectionScoped lock(codec_inst_crit_sect_.get()); return ACM_ISAC_SETMAXPAYLOADSIZE(codec_inst_ptr_->inst, max_payload_len_bytes); } int32_t ACMISAC::SetISACMaxRate(const uint32_t max_rate_bit_per_sec) { + CriticalSectionScoped lock(codec_inst_crit_sect_.get()); return ACM_ISAC_SETMAXRATE(codec_inst_ptr_->inst, max_rate_bit_per_sec); } -void ACMISAC::UpdateFrameLen() { - frame_len_smpl_ = ACM_ISAC_GETNEWFRAMELEN(codec_inst_ptr_->inst); - encoder_params_.codec_inst.pacsize = frame_len_smpl_; -} - void ACMISAC::CurrentRate(int32_t* rate_bit_per_sec) { if (isac_coding_mode_ == ADAPTIVE) { + CriticalSectionScoped lock(codec_inst_crit_sect_.get()); ACM_ISAC_GETSENDBITRATE(codec_inst_ptr_->inst, rate_bit_per_sec); } } @@ -784,12 +729,71 @@ int16_t ACMISAC::REDPayloadISAC(const int32_t isac_rate, return status; } -AudioDecoder* ACMISAC::Decoder(int codec_id) { - if (audio_decoder_) - return audio_decoder_; +int ACMISAC::Decode(const uint8_t* encoded, + size_t encoded_len, + int16_t* decoded, + SpeechType* speech_type) { + int16_t temp_type; + CriticalSectionScoped lock(codec_inst_crit_sect_.get()); + int ret = + ACM_ISAC_DECODE_B(static_cast<ACM_ISAC_STRUCT*>(codec_inst_ptr_->inst), + reinterpret_cast<const uint16_t*>(encoded), + static_cast<int16_t>(encoded_len), + decoded, + &temp_type); + *speech_type = ConvertSpeechType(temp_type); + return ret; +} + +int ACMISAC::DecodePlc(int num_frames, int16_t* decoded) { + CriticalSectionScoped lock(codec_inst_crit_sect_.get()); + return ACM_ISAC_DECODEPLC( + static_cast<ACM_ISAC_STRUCT*>(codec_inst_ptr_->inst), + decoded, + static_cast<int16_t>(num_frames)); +} + +int ACMISAC::IncomingPacket(const uint8_t* payload, + size_t payload_len, + uint16_t rtp_sequence_number, + uint32_t rtp_timestamp, + uint32_t arrival_timestamp) { + CriticalSectionScoped lock(codec_inst_crit_sect_.get()); + return ACM_ISAC_DECODE_BWE( + static_cast<ACM_ISAC_STRUCT*>(codec_inst_ptr_->inst), + reinterpret_cast<const uint16_t*>(payload), + static_cast<uint32_t>(payload_len), + rtp_sequence_number, + rtp_timestamp, + arrival_timestamp); +} + +int ACMISAC::DecodeRedundant(const uint8_t* encoded, + size_t encoded_len, + int16_t* decoded, + SpeechType* speech_type) { + int16_t temp_type = 1; // Default is speech. + CriticalSectionScoped lock(codec_inst_crit_sect_.get()); + int16_t ret = + ACM_ISAC_DECODERCU(static_cast<ACM_ISAC_STRUCT*>(codec_inst_ptr_->inst), + reinterpret_cast<const uint16_t*>(encoded), + static_cast<int16_t>(encoded_len), + decoded, + &temp_type); + *speech_type = ConvertSpeechType(temp_type); + return ret; +} + +int ACMISAC::ErrorCode() { + CriticalSectionScoped lock(codec_inst_crit_sect_.get()); + return ACM_ISAC_GETERRORCODE( + static_cast<ACM_ISAC_STRUCT*>(codec_inst_ptr_->inst)); +} +AudioDecoder* ACMISAC::Decoder(int codec_id) { // Create iSAC instance if it does not exist. if (!encoder_exist_) { + CriticalSectionScoped lock(codec_inst_crit_sect_.get()); assert(codec_inst_ptr_->inst == NULL); encoder_initialized_ = false; decoder_initialized_ = false; @@ -822,8 +826,7 @@ AudioDecoder* ACMISAC::Decoder(int codec_id) { decoder_initialized_ = true; } - audio_decoder_ = new AcmAudioDecoderIsac(codec_id, codec_inst_ptr_->inst); - return audio_decoder_; + return this; } #endif |