| /* |
| * 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/main/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/main/interface/audio_coding_module_typedefs.h" |
| #include "webrtc/modules/audio_coding/main/acm2/acm_common_defs.h" |
| #include "webrtc/modules/audio_coding/main/acm2/acm_generic_codec.h" |
| #include "webrtc/modules/audio_coding/main/acm2/acm_resampler.h" |
| #include "webrtc/modules/audio_coding/main/acm2/call_statistics.h" |
| #include "webrtc/system_wrappers/interface/critical_section_wrapper.h" |
| #include "webrtc/system_wrappers/interface/rw_lock_wrapper.h" |
| #include "webrtc/system_wrappers/interface/trace.h" |
| #include "webrtc/typedefs.h" |
| |
| namespace webrtc { |
| |
| namespace acm2 { |
| |
| enum { |
| kACMToneEnd = 999 |
| }; |
| |
| // Maximum number of bytes in one packet (PCM16B, 20 ms packets, stereo). |
| enum { |
| kMaxPacketSize = 2560 |
| }; |
| |
| // Maximum number of payloads that can be packed in one RED packet. For |
| // regular RED, we only pack two payloads. In case of dual-streaming, in worst |
| // case we might pack 3 payloads in one RED packet. |
| enum { |
| kNumRedFragmentationVectors = 2, |
| kMaxNumFragmentationVectors = 3 |
| }; |
| |
| // If packet N is arrived all packets prior to N - |kNackThresholdPackets| which |
| // are not received are considered as lost, and appear in NACK list. |
| enum { |
| kNackThresholdPackets = 2 |
| }; |
| |
| namespace { |
| |
| // TODO(turajs): the same functionality is used in NetEq. If both classes |
| // need them, make it a static function in ACMCodecDB. |
| bool IsCodecRED(const CodecInst* codec) { |
| return (STR_CASE_CMP(codec->plname, "RED") == 0); |
| } |
| |
| bool IsCodecRED(int index) { |
| return (IsCodecRED(&ACMCodecDB::database_[index])); |
| } |
| |
| bool IsCodecCN(const CodecInst* codec) { |
| return (STR_CASE_CMP(codec->plname, "CN") == 0); |
| } |
| |
| bool IsCodecCN(int index) { |
| return (IsCodecCN(&ACMCodecDB::database_[index])); |
| } |
| |
| // Stereo-to-mono can be used as in-place. |
| int DownMix(const AudioFrame& frame, int length_out_buff, int16_t* out_buff) { |
| if (length_out_buff < frame.samples_per_channel_) { |
| return -1; |
| } |
| for (int n = 0; n < frame.samples_per_channel_; ++n) |
| out_buff[n] = (frame.data_[2 * n] + frame.data_[2 * n + 1]) >> 1; |
| return 0; |
| } |
| |
| // Mono-to-stereo can be used as in-place. |
| int UpMix(const AudioFrame& frame, int length_out_buff, int16_t* out_buff) { |
| if (length_out_buff < frame.samples_per_channel_) { |
| return -1; |
| } |
| for (int n = frame.samples_per_channel_ - 1; n >= 0; --n) { |
| out_buff[2 * n + 1] = frame.data_[n]; |
| out_buff[2 * n] = frame.data_[n]; |
| } |
| return 0; |
| } |
| |
| void ConvertEncodedInfoToFragmentationHeader( |
| const AudioEncoder::EncodedInfo& info, |
| RTPFragmentationHeader* frag) { |
| if (info.redundant.empty()) { |
| frag->fragmentationVectorSize = 0; |
| return; |
| } |
| |
| frag->VerifyAndAllocateFragmentationHeader( |
| static_cast<uint16_t>(info.redundant.size())); |
| frag->fragmentationVectorSize = static_cast<uint16_t>(info.redundant.size()); |
| size_t offset = 0; |
| for (size_t i = 0; i < info.redundant.size(); ++i) { |
| frag->fragmentationOffset[i] = offset; |
| offset += info.redundant[i].encoded_bytes; |
| frag->fragmentationLength[i] = info.redundant[i].encoded_bytes; |
| frag->fragmentationTimeDiff[i] = rtc::checked_cast<uint16_t>( |
| info.encoded_timestamp - info.redundant[i].encoded_timestamp); |
| frag->fragmentationPlType[i] = info.redundant[i].payload_type; |
| } |
| } |
| } // namespace |
| |
| AudioCodingModuleImpl::AudioCodingModuleImpl( |
| const AudioCodingModule::Config& config) |
| : acm_crit_sect_(CriticalSectionWrapper::CreateCriticalSection()), |
| id_(config.id), |
| expected_codec_ts_(0xD87F3F9F), |
| expected_in_ts_(0xD87F3F9F), |
| receiver_(config), |
| codec_manager_(this), |
| previous_pltype_(255), |
| aux_rtp_header_(NULL), |
| 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() { |
| if (aux_rtp_header_ != NULL) { |
| delete aux_rtp_header_; |
| aux_rtp_header_ = NULL; |
| } |
| |
| delete callback_crit_sect_; |
| callback_crit_sect_ = NULL; |
| |
| delete acm_crit_sect_; |
| acm_crit_sect_ = NULL; |
| WEBRTC_TRACE(webrtc::kTraceMemory, webrtc::kTraceAudioCoding, id_, |
| "Destroyed"); |
| } |
| |
| int32_t AudioCodingModuleImpl::Encode(const InputData& input_data) { |
| uint8_t stream[2 * MAX_PAYLOAD_SIZE_BYTE]; // Make room for 1 RED payload. |
| AudioEncoder::EncodedInfo encoded_info; |
| uint8_t previous_pltype; |
| |
| // Keep the scope of the ACM critical section limited. |
| { |
| CriticalSectionScoped lock(acm_crit_sect_); |
| // Check if there is an encoder before. |
| if (!HaveValidEncoder("Process")) { |
| return -1; |
| } |
| |
| AudioEncoder* audio_encoder = |
| codec_manager_.current_encoder()->GetAudioEncoder(); |
| // Scale the timestamp to the codec's RTP timestamp rate. |
| uint32_t rtp_timestamp = |
| first_frame_ ? input_data.input_timestamp |
| : last_rtp_timestamp_ + |
| rtc::CheckedDivExact( |
| input_data.input_timestamp - last_timestamp_, |
| static_cast<uint32_t>(rtc::CheckedDivExact( |
| audio_encoder->SampleRateHz(), |
| audio_encoder->RtpTimestampRateHz()))); |
| last_timestamp_ = input_data.input_timestamp; |
| last_rtp_timestamp_ = rtp_timestamp; |
| first_frame_ = false; |
| |
| encoded_info = audio_encoder->Encode(rtp_timestamp, input_data.audio, |
| input_data.length_per_channel, |
| sizeof(stream), stream); |
| if (encoded_info.encoded_bytes == 0 && !encoded_info.send_even_if_empty) { |
| // Not enough data. |
| return 0; |
| } |
| previous_pltype = previous_pltype_; // Read it while we have the critsect. |
| } |
| |
| RTPFragmentationHeader my_fragmentation; |
| ConvertEncodedInfoToFragmentationHeader(encoded_info, &my_fragmentation); |
| FrameType frame_type; |
| if (encoded_info.encoded_bytes == 0 && encoded_info.send_even_if_empty) { |
| frame_type = kFrameEmpty; |
| encoded_info.payload_type = previous_pltype; |
| } else { |
| DCHECK_GT(encoded_info.encoded_bytes, 0u); |
| frame_type = encoded_info.speech ? kAudioFrameSpeech : kAudioFrameCN; |
| } |
| |
| { |
| CriticalSectionScoped lock(callback_crit_sect_); |
| if (packetization_callback_) { |
| packetization_callback_->SendData( |
| frame_type, encoded_info.payload_type, encoded_info.encoded_timestamp, |
| stream, encoded_info.encoded_bytes, |
| my_fragmentation.fragmentationVectorSize > 0 ? &my_fragmentation |
| : nullptr); |
| } |
| |
| if (vad_callback_) { |
| // Callback with VAD decision. |
| vad_callback_->InFrameType(frame_type); |
| } |
| } |
| { |
| CriticalSectionScoped lock(acm_crit_sect_); |
| previous_pltype_ = encoded_info.payload_type; |
| } |
| return static_cast<int32_t>(encoded_info.encoded_bytes); |
| } |
| |
| ///////////////////////////////////////// |
| // Sender |
| // |
| |
| // TODO(henrik.lundin): Remove this method; only used in tests. |
| int AudioCodingModuleImpl::ResetEncoder() { |
| CriticalSectionScoped lock(acm_crit_sect_); |
| if (!HaveValidEncoder("ResetEncoder")) { |
| return -1; |
| } |
| return 0; |
| } |
| |
| // Can be called multiple times for Codec, CNG, RED. |
| int AudioCodingModuleImpl::RegisterSendCodec(const CodecInst& send_codec) { |
| CriticalSectionScoped lock(acm_crit_sect_); |
| return codec_manager_.RegisterSendCodec(send_codec); |
| } |
| |
| // Get current send codec. |
| int AudioCodingModuleImpl::SendCodec(CodecInst* current_codec) const { |
| CriticalSectionScoped lock(acm_crit_sect_); |
| return codec_manager_.SendCodec(current_codec); |
| } |
| |
| // Get current send frequency. |
| int AudioCodingModuleImpl::SendFrequency() const { |
| WEBRTC_TRACE(webrtc::kTraceStream, webrtc::kTraceAudioCoding, id_, |
| "SendFrequency()"); |
| CriticalSectionScoped lock(acm_crit_sect_); |
| |
| if (!codec_manager_.current_encoder()) { |
| WEBRTC_TRACE(webrtc::kTraceStream, webrtc::kTraceAudioCoding, id_, |
| "SendFrequency Failed, no codec is registered"); |
| return -1; |
| } |
| |
| return codec_manager_.current_encoder()->GetAudioEncoder()->SampleRateHz(); |
| } |
| |
| // Get encode bitrate. |
| // Adaptive rate codecs return their current encode target rate, while other |
| // codecs return there longterm avarage or their fixed rate. |
| // TODO(henrik.lundin): Remove; not used. |
| int AudioCodingModuleImpl::SendBitrate() const { |
| CriticalSectionScoped lock(acm_crit_sect_); |
| |
| if (!codec_manager_.current_encoder()) { |
| WEBRTC_TRACE(webrtc::kTraceStream, webrtc::kTraceAudioCoding, id_, |
| "SendBitrate Failed, no codec is registered"); |
| return -1; |
| } |
| |
| WebRtcACMCodecParams encoder_param; |
| codec_manager_.current_encoder()->EncoderParams(&encoder_param); |
| |
| return encoder_param.codec_inst.rate; |
| } |
| |
| // Set available bandwidth, inform the encoder about the estimated bandwidth |
| // received from the remote party. |
| // TODO(henrik.lundin): Remove; not used. |
| int AudioCodingModuleImpl::SetReceivedEstimatedBandwidth(int bw) { |
| CriticalSectionScoped lock(acm_crit_sect_); |
| FATAL() << "Dead code?"; |
| return -1; |
| // return codecs_[current_send_codec_idx_]->SetEstimatedBandwidth(bw); |
| } |
| |
| // Register a transport callback which will be called to deliver |
| // the encoded buffers. |
| int AudioCodingModuleImpl::RegisterTransportCallback( |
| AudioPacketizationCallback* transport) { |
| CriticalSectionScoped 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; |
| 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 negative or " |
| "zero"); |
| return -1; |
| } |
| |
| if (audio_frame.sample_rate_hz_ > 48000) { |
| assert(false); |
| WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_, |
| "Cannot Add 10 ms audio, input frequency not valid"); |
| return -1; |
| } |
| |
| // If the length and frequency matches. We currently just support raw PCM. |
| if ((audio_frame.sample_rate_hz_ / 100) |
| != audio_frame.samples_per_channel_) { |
| WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_, |
| "Cannot Add 10 ms audio, input frequency and length doesn't" |
| " match"); |
| return -1; |
| } |
| |
| if (audio_frame.num_channels_ != 1 && audio_frame.num_channels_ != 2) { |
| WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_, |
| "Cannot Add 10 ms audio, invalid number of channels."); |
| return -1; |
| } |
| |
| CriticalSectionScoped lock(acm_crit_sect_); |
| // Do we have a codec registered? |
| if (!HaveValidEncoder("Add10MsData")) { |
| return -1; |
| } |
| |
| const AudioFrame* ptr_frame; |
| // Perform a resampling, also down-mix if it is required and can be |
| // performed before resampling (a down mix prior to resampling will take |
| // place if both primary and secondary encoders are mono and input is in |
| // stereo). |
| if (PreprocessToAddData(audio_frame, &ptr_frame) < 0) { |
| return -1; |
| } |
| |
| // Check whether we need an up-mix or down-mix? |
| bool remix = |
| ptr_frame->num_channels_ != |
| codec_manager_.current_encoder()->GetAudioEncoder()->NumChannels(); |
| |
| if (remix) { |
| if (ptr_frame->num_channels_ == 1) { |
| if (UpMix(*ptr_frame, WEBRTC_10MS_PCM_AUDIO, input_data->buffer) < 0) |
| return -1; |
| } else { |
| if (DownMix(*ptr_frame, WEBRTC_10MS_PCM_AUDIO, input_data->buffer) < 0) |
| return -1; |
| } |
| } |
| |
| // When adding data to encoders this pointer is pointing to an audio buffer |
| // with correct number of channels. |
| const int16_t* ptr_audio = ptr_frame->data_; |
| |
| // For pushing data to primary, point the |ptr_audio| to correct buffer. |
| if (codec_manager_.current_encoder()->GetAudioEncoder()->NumChannels() != |
| ptr_frame->num_channels_) |
| ptr_audio = input_data->buffer; |
| |
| input_data->input_timestamp = ptr_frame->timestamp_; |
| input_data->audio = ptr_audio; |
| input_data->length_per_channel = ptr_frame->samples_per_channel_; |
| input_data->audio_channel = |
| codec_manager_.current_encoder()->GetAudioEncoder()->NumChannels(); |
| |
| return 0; |
| } |
| |
| // Perform a resampling and down-mix if required. We down-mix only if |
| // encoder is mono and input is stereo. In case of dual-streaming, both |
| // encoders has to be mono for down-mix to take place. |
| // |*ptr_out| will point to the pre-processed audio-frame. If no pre-processing |
| // is required, |*ptr_out| points to |in_frame|. |
| int AudioCodingModuleImpl::PreprocessToAddData(const AudioFrame& in_frame, |
| const AudioFrame** ptr_out) { |
| bool resample = |
| (in_frame.sample_rate_hz_ != |
| codec_manager_.current_encoder()->GetAudioEncoder()->SampleRateHz()); |
| |
| // This variable is true if primary codec and secondary codec (if exists) |
| // are both mono and input is stereo. |
| bool down_mix = |
| (in_frame.num_channels_ == 2) && |
| (codec_manager_.current_encoder()->GetAudioEncoder()->NumChannels() == 1); |
| |
| if (!first_10ms_data_) { |
| expected_in_ts_ = in_frame.timestamp_; |
| expected_codec_ts_ = in_frame.timestamp_; |
| first_10ms_data_ = true; |
| } else if (in_frame.timestamp_ != expected_in_ts_) { |
| // TODO(turajs): Do we need a warning here. |
| expected_codec_ts_ += |
| (in_frame.timestamp_ - expected_in_ts_) * |
| static_cast<uint32_t>( |
| (static_cast<double>(codec_manager_.current_encoder() |
| ->GetAudioEncoder() |
| ->SampleRateHz()) / |
| static_cast<double>(in_frame.sample_rate_hz_))); |
| expected_in_ts_ = in_frame.timestamp_; |
| } |
| |
| |
| if (!down_mix && !resample) { |
| // No pre-processing is required. |
| expected_in_ts_ += in_frame.samples_per_channel_; |
| expected_codec_ts_ += in_frame.samples_per_channel_; |
| *ptr_out = &in_frame; |
| return 0; |
| } |
| |
| *ptr_out = &preprocess_frame_; |
| preprocess_frame_.num_channels_ = in_frame.num_channels_; |
| int16_t audio[WEBRTC_10MS_PCM_AUDIO]; |
| const int16_t* src_ptr_audio = in_frame.data_; |
| int16_t* dest_ptr_audio = preprocess_frame_.data_; |
| if (down_mix) { |
| // If a resampling is required the output of a down-mix is written into a |
| // local buffer, otherwise, it will be written to the output frame. |
| if (resample) |
| dest_ptr_audio = audio; |
| if (DownMix(in_frame, WEBRTC_10MS_PCM_AUDIO, dest_ptr_audio) < 0) |
| return -1; |
| preprocess_frame_.num_channels_ = 1; |
| // Set the input of the resampler is the down-mixed signal. |
| src_ptr_audio = audio; |
| } |
| |
| preprocess_frame_.timestamp_ = expected_codec_ts_; |
| preprocess_frame_.samples_per_channel_ = in_frame.samples_per_channel_; |
| preprocess_frame_.sample_rate_hz_ = in_frame.sample_rate_hz_; |
| // If it is required, we have to do a resampling. |
| if (resample) { |
| // The result of the resampler is written to output frame. |
| dest_ptr_audio = preprocess_frame_.data_; |
| |
| preprocess_frame_.samples_per_channel_ = resampler_.Resample10Msec( |
| src_ptr_audio, in_frame.sample_rate_hz_, |
| codec_manager_.current_encoder()->GetAudioEncoder()->SampleRateHz(), |
| preprocess_frame_.num_channels_, AudioFrame::kMaxDataSizeSamples, |
| dest_ptr_audio); |
| |
| if (preprocess_frame_.samples_per_channel_ < 0) { |
| WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_, |
| "Cannot add 10 ms audio, resampling failed"); |
| return -1; |
| } |
| preprocess_frame_.sample_rate_hz_ = |
| codec_manager_.current_encoder()->GetAudioEncoder()->SampleRateHz(); |
| } |
| |
| expected_codec_ts_ += preprocess_frame_.samples_per_channel_; |
| expected_in_ts_ += in_frame.samples_per_channel_; |
| |
| return 0; |
| } |
| |
| ///////////////////////////////////////// |
| // (RED) Redundant Coding |
| // |
| |
| bool AudioCodingModuleImpl::REDStatus() const { |
| CriticalSectionScoped lock(acm_crit_sect_); |
| return codec_manager_.red_enabled(); |
| } |
| |
| // Configure RED status i.e on/off. |
| int AudioCodingModuleImpl::SetREDStatus( |
| #ifdef WEBRTC_CODEC_RED |
| bool enable_red) { |
| CriticalSectionScoped lock(acm_crit_sect_); |
| return codec_manager_.SetCopyRed(enable_red) ? 0 : -1; |
| #else |
| bool /* enable_red */) { |
| 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_); |
| return codec_manager_.codec_fec_enabled(); |
| } |
| |
| int AudioCodingModuleImpl::SetCodecFEC(bool enable_codec_fec) { |
| CriticalSectionScoped lock(acm_crit_sect_); |
| return codec_manager_.SetCodecFEC(enable_codec_fec); |
| } |
| |
| int AudioCodingModuleImpl::SetPacketLossRate(int loss_rate) { |
| CriticalSectionScoped lock(acm_crit_sect_); |
| if (HaveValidEncoder("SetPacketLossRate") && |
| codec_manager_.current_encoder()->SetPacketLossRate(loss_rate) < 0) { |
| WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_, |
| "Set packet loss rate failed."); |
| return -1; |
| } |
| return 0; |
| } |
| |
| ///////////////////////////////////////// |
| // (VAD) Voice Activity Detection |
| // |
| int AudioCodingModuleImpl::SetVAD(bool enable_dtx, |
| bool enable_vad, |
| ACMVADMode mode) { |
| CriticalSectionScoped lock(acm_crit_sect_); |
| return codec_manager_.SetVAD(enable_dtx, enable_vad, mode); |
| } |
| |
| // Get VAD/DTX settings. |
| int AudioCodingModuleImpl::VAD(bool* dtx_enabled, bool* vad_enabled, |
| ACMVADMode* mode) const { |
| CriticalSectionScoped lock(acm_crit_sect_); |
| codec_manager_.VAD(dtx_enabled, vad_enabled, mode); |
| return 0; |
| } |
| |
| ///////////////////////////////////////// |
| // Receiver |
| // |
| |
| int AudioCodingModuleImpl::InitializeReceiver() { |
| CriticalSectionScoped 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) |
| return -1; |
| } |
| receiver_.set_id(id_); |
| receiver_.ResetInitialDelay(); |
| receiver_.SetMinimumDelay(0); |
| receiver_.SetMaximumDelay(0); |
| receiver_.FlushBuffers(); |
| |
| // Register RED and CN. |
| for (int i = 0; i < ACMCodecDB::kNumCodecs; i++) { |
| if (IsCodecRED(i) || IsCodecCN(i)) { |
| uint8_t pl_type = static_cast<uint8_t>(ACMCodecDB::database_[i].pltype); |
| if (receiver_.AddCodec(i, pl_type, 1, NULL) < 0) { |
| WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_, |
| "Cannot register master codec."); |
| return -1; |
| } |
| } |
| } |
| receiver_initialized_ = true; |
| return 0; |
| } |
| |
| // TODO(turajs): If NetEq opens an API for reseting the state of decoders then |
| // implement this method. Otherwise it should be removed. I might be that by |
| // removing and registering a decoder we can achieve the effect of resetting. |
| // Reset the decoder state. |
| // TODO(henrik.lundin): Remove; only used in one test, and does nothing. |
| int AudioCodingModuleImpl::ResetDecoder() { |
| return 0; |
| } |
| |
| // Get current receive frequency. |
| int AudioCodingModuleImpl::ReceiveFrequency() const { |
| WEBRTC_TRACE(webrtc::kTraceStream, webrtc::kTraceAudioCoding, id_, |
| "ReceiveFrequency()"); |
| |
| CriticalSectionScoped lock(acm_crit_sect_); |
| |
| int codec_id = receiver_.last_audio_codec_id(); |
| |
| return codec_id < 0 ? receiver_.current_sample_rate_hz() : |
| ACMCodecDB::database_[codec_id].plfreq; |
| } |
| |
| // Get current playout frequency. |
| int AudioCodingModuleImpl::PlayoutFrequency() const { |
| WEBRTC_TRACE(webrtc::kTraceStream, webrtc::kTraceAudioCoding, id_, |
| "PlayoutFrequency()"); |
| |
| CriticalSectionScoped lock(acm_crit_sect_); |
| |
| return receiver_.current_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_); |
| DCHECK(receiver_initialized_); |
| return codec_manager_.RegisterReceiveCodec(codec); |
| } |
| |
| // Get current received codec. |
| int AudioCodingModuleImpl::ReceiveCodec(CodecInst* current_codec) const { |
| CriticalSectionScoped lock(acm_crit_sect_); |
| return receiver_.LastAudioCodec(current_codec); |
| } |
| |
| int AudioCodingModuleImpl::RegisterDecoder(int acm_codec_id, |
| uint8_t payload_type, |
| int channels, |
| AudioDecoder* audio_decoder) { |
| return receiver_.AddCodec(acm_codec_id, payload_type, channels, |
| audio_decoder); |
| } |
| |
| // 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, incoming_payload, payload_length); |
| } |
| |
| // Minimum playout delay (Used for lip-sync). |
| int AudioCodingModuleImpl::SetMinimumPlayoutDelay(int time_ms) { |
| if ((time_ms < 0) || (time_ms > 10000)) { |
| WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_, |
| "Delay must be in the range of 0-1000 milliseconds."); |
| return -1; |
| } |
| return receiver_.SetMinimumDelay(time_ms); |
| } |
| |
| int AudioCodingModuleImpl::SetMaximumPlayoutDelay(int time_ms) { |
| if ((time_ms < 0) || (time_ms > 10000)) { |
| WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_, |
| "Delay must be in the range of 0-1000 milliseconds."); |
| return -1; |
| } |
| return receiver_.SetMaximumDelay(time_ms); |
| } |
| |
| // Estimate the Bandwidth based on the incoming stream, needed for one way |
| // audio where the RTCP send the BW estimate. |
| // This is also done in the RTP module. |
| int AudioCodingModuleImpl::DecoderEstimatedBandwidth() const { |
| // We can estimate far-end to near-end bandwidth if the iSAC are sent. Check |
| // if the last received packets were iSAC packet then retrieve the bandwidth. |
| int last_audio_codec_id = receiver_.last_audio_codec_id(); |
| if (last_audio_codec_id >= 0 && |
| STR_CASE_CMP("ISAC", ACMCodecDB::database_[last_audio_codec_id].plname)) { |
| CriticalSectionScoped lock(acm_crit_sect_); |
| FATAL() << "Dead code?"; |
| // return codecs_[last_audio_codec_id]->GetEstimatedBandwidth(); |
| } |
| return -1; |
| } |
| |
| // Set playout mode for: voice, fax, streaming or off. |
| int AudioCodingModuleImpl::SetPlayoutMode(AudioPlayoutMode mode) { |
| receiver_.SetPlayoutMode(mode); |
| return 0; // TODO(turajs): return value is for backward compatibility. |
| } |
| |
| // Get playout mode voice, fax, streaming or off. |
| AudioPlayoutMode AudioCodingModuleImpl::PlayoutMode() const { |
| return receiver_.PlayoutMode(); |
| } |
| |
| // Get 10 milliseconds of raw audio data to play out. |
| // Automatic resample to the requested frequency. |
| int AudioCodingModuleImpl::PlayoutData10Ms(int desired_freq_hz, |
| AudioFrame* audio_frame) { |
| // GetAudio always returns 10 ms, at the requested sample rate. |
| if (receiver_.GetAudio(desired_freq_hz, audio_frame) != 0) { |
| WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_, |
| "PlayoutData failed, RecOut Failed"); |
| return -1; |
| } |
| |
| audio_frame->id_ = id_; |
| return 0; |
| } |
| |
| ///////////////////////////////////////// |
| // Statistics |
| // |
| |
| // 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_); |
| vad_callback_ = vad_callback; |
| return 0; |
| } |
| |
| // TODO(tlegrand): Modify this function to work for stereo, and add tests. |
| int AudioCodingModuleImpl::IncomingPayload(const uint8_t* incoming_payload, |
| size_t payload_length, |
| uint8_t payload_type, |
| uint32_t timestamp) { |
| // We are not acquiring any lock when interacting with |aux_rtp_header_| no |
| // other method uses this member variable. |
| if (aux_rtp_header_ == NULL) { |
| // This is the first time that we are using |dummy_rtp_header_| |
| // so we have to create it. |
| aux_rtp_header_ = new WebRtcRTPHeader; |
| aux_rtp_header_->header.payloadType = payload_type; |
| // Don't matter in this case. |
| aux_rtp_header_->header.ssrc = 0; |
| aux_rtp_header_->header.markerBit = false; |
| // Start with random numbers. |
| aux_rtp_header_->header.sequenceNumber = 0x1234; // Arbitrary. |
| aux_rtp_header_->type.Audio.channel = 1; |
| } |
| |
| 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::ReplaceInternalDTXWithWebRtc(bool use_webrtc_dtx) { |
| CriticalSectionScoped lock(acm_crit_sect_); |
| |
| if (!HaveValidEncoder("ReplaceInternalDTXWithWebRtc")) { |
| WEBRTC_TRACE( |
| webrtc::kTraceError, webrtc::kTraceAudioCoding, id_, |
| "Cannot replace codec internal DTX when no send codec is registered."); |
| return -1; |
| } |
| |
| FATAL() << "Dead code?"; |
| // int res = codecs_[current_send_codec_idx_]->ReplaceInternalDTX( |
| // use_webrtc_dtx); |
| // Check if VAD is turned on, or if there is any error. |
| // if (res == 1) { |
| // vad_enabled_ = true; |
| // } else if (res < 0) { |
| // WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_, |
| // "Failed to set ReplaceInternalDTXWithWebRtc(%d)", |
| // use_webrtc_dtx); |
| // return res; |
| // } |
| |
| return 0; |
| } |
| |
| int AudioCodingModuleImpl::IsInternalDTXReplacedWithWebRtc( |
| bool* uses_webrtc_dtx) { |
| *uses_webrtc_dtx = true; |
| return 0; |
| } |
| |
| // TODO(henrik.lundin): Remove? Only used in tests. Deprecated in VoiceEngine. |
| int AudioCodingModuleImpl::SetISACMaxRate(int max_bit_per_sec) { |
| CriticalSectionScoped lock(acm_crit_sect_); |
| |
| if (!HaveValidEncoder("SetISACMaxRate")) { |
| return -1; |
| } |
| |
| return codec_manager_.current_encoder()->SetISACMaxRate(max_bit_per_sec); |
| } |
| |
| // TODO(henrik.lundin): Remove? Only used in tests. Deprecated in VoiceEngine. |
| int AudioCodingModuleImpl::SetISACMaxPayloadSize(int max_size_bytes) { |
| CriticalSectionScoped lock(acm_crit_sect_); |
| |
| if (!HaveValidEncoder("SetISACMaxPayloadSize")) { |
| return -1; |
| } |
| |
| return codec_manager_.current_encoder()->SetISACMaxPayloadSize( |
| max_size_bytes); |
| } |
| |
| // TODO(henrik.lundin): Remove? Only used in tests. |
| int AudioCodingModuleImpl::ConfigISACBandwidthEstimator( |
| int frame_size_ms, |
| int rate_bit_per_sec, |
| bool enforce_frame_size) { |
| CriticalSectionScoped lock(acm_crit_sect_); |
| |
| if (!HaveValidEncoder("ConfigISACBandwidthEstimator")) { |
| return -1; |
| } |
| |
| FATAL() << "Dead code?"; |
| return -1; |
| // return codecs_[current_send_codec_idx_]->ConfigISACBandwidthEstimator( |
| // frame_size_ms, rate_bit_per_sec, enforce_frame_size); |
| } |
| |
| int AudioCodingModuleImpl::SetOpusApplication(OpusApplicationMode application, |
| bool disable_dtx_if_needed) { |
| CriticalSectionScoped lock(acm_crit_sect_); |
| if (!HaveValidEncoder("SetOpusApplication")) { |
| return -1; |
| } |
| return codec_manager_.current_encoder()->SetOpusApplication( |
| application, disable_dtx_if_needed); |
| } |
| |
| // Informs Opus encoder of the maximum playback rate the receiver will render. |
| int AudioCodingModuleImpl::SetOpusMaxPlaybackRate(int frequency_hz) { |
| CriticalSectionScoped lock(acm_crit_sect_); |
| if (!HaveValidEncoder("SetOpusMaxPlaybackRate")) { |
| return -1; |
| } |
| return codec_manager_.current_encoder()->SetOpusMaxPlaybackRate(frequency_hz); |
| } |
| |
| int AudioCodingModuleImpl::EnableOpusDtx(bool force_voip) { |
| CriticalSectionScoped lock(acm_crit_sect_); |
| if (!HaveValidEncoder("EnableOpusDtx")) { |
| return -1; |
| } |
| return codec_manager_.current_encoder()->EnableOpusDtx(force_voip); |
| } |
| |
| int AudioCodingModuleImpl::DisableOpusDtx() { |
| CriticalSectionScoped lock(acm_crit_sect_); |
| if (!HaveValidEncoder("DisableOpusDtx")) { |
| return -1; |
| } |
| return codec_manager_.current_encoder()->DisableOpusDtx(); |
| } |
| |
| int AudioCodingModuleImpl::PlayoutTimestamp(uint32_t* timestamp) { |
| return receiver_.GetPlayoutTimestamp(timestamp) ? 0 : -1; |
| } |
| |
| bool AudioCodingModuleImpl::HaveValidEncoder(const char* caller_name) const { |
| if (!codec_manager_.current_encoder()) { |
| WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_, |
| "%s failed: No send codec is registered.", caller_name); |
| return false; |
| } |
| return true; |
| } |
| |
| int AudioCodingModuleImpl::UnregisterReceiveCodec(uint8_t payload_type) { |
| return receiver_.RemoveCodec(payload_type); |
| } |
| |
| // TODO(turajs): correct the type of |length_bytes| when it is corrected in |
| // GenericCodec. |
| int AudioCodingModuleImpl::REDPayloadISAC(int isac_rate, |
| int isac_bw_estimate, |
| uint8_t* payload, |
| int16_t* length_bytes) { |
| CriticalSectionScoped lock(acm_crit_sect_); |
| if (!HaveValidEncoder("EncodeData")) { |
| return -1; |
| } |
| FATAL() << "Dead code?"; |
| return -1; |
| // int status; |
| // status = codecs_[current_send_codec_idx_]->REDPayloadISAC(isac_rate, |
| // isac_bw_estimate, |
| // payload, |
| // length_bytes); |
| // return status; |
| } |
| |
| int AudioCodingModuleImpl::SetInitialPlayoutDelay(int delay_ms) { |
| { |
| CriticalSectionScoped lock(acm_crit_sect_); |
| // Initialize receiver, if it is not initialized. Otherwise, initial delay |
| // is reset upon initialization of the receiver. |
| if (!receiver_initialized_) |
| InitializeReceiverSafe(); |
| } |
| return receiver_.SetInitialDelay(delay_ms); |
| } |
| |
| int AudioCodingModuleImpl::SetDtmfPlayoutStatus(bool enable) { |
| return 0; |
| } |
| |
| bool AudioCodingModuleImpl::DtmfPlayoutStatus() const { |
| return true; |
| } |
| |
| int AudioCodingModuleImpl::EnableNack(size_t max_nack_list_size) { |
| return receiver_.EnableNack(max_nack_list_size); |
| } |
| |
| void AudioCodingModuleImpl::DisableNack() { |
| receiver_.DisableNack(); |
| } |
| |
| std::vector<uint16_t> AudioCodingModuleImpl::GetNackList( |
| int64_t round_trip_time_ms) const { |
| return receiver_.GetNackList(round_trip_time_ms); |
| } |
| |
| int AudioCodingModuleImpl::LeastRequiredDelayMs() const { |
| return receiver_.LeastRequiredDelayMs(); |
| } |
| |
| void AudioCodingModuleImpl::GetDecodingCallStatistics( |
| AudioDecodingCallStats* call_stats) const { |
| receiver_.GetDecodingCallStatistics(call_stats); |
| } |
| |
| } // namespace acm2 |
| |
| AudioCodingImpl::AudioCodingImpl(const Config& config) { |
| AudioCodingModule::Config config_old = config.ToOldConfig(); |
| acm_old_.reset(new acm2::AudioCodingModuleImpl(config_old)); |
| acm_old_->RegisterTransportCallback(config.transport); |
| acm_old_->RegisterVADCallback(config.vad_callback); |
| acm_old_->SetDtmfPlayoutStatus(config.play_dtmf); |
| if (config.initial_playout_delay_ms > 0) { |
| acm_old_->SetInitialPlayoutDelay(config.initial_playout_delay_ms); |
| } |
| playout_frequency_hz_ = config.playout_frequency_hz; |
| } |
| |
| AudioCodingImpl::~AudioCodingImpl() = default; |
| |
| bool AudioCodingImpl::RegisterSendCodec(AudioEncoder* send_codec) { |
| FATAL() << "Not implemented yet."; |
| return false; |
| } |
| |
| bool AudioCodingImpl::RegisterSendCodec(int encoder_type, |
| uint8_t payload_type, |
| int frame_size_samples) { |
| std::string codec_name; |
| int sample_rate_hz; |
| int channels; |
| if (!MapCodecTypeToParameters( |
| encoder_type, &codec_name, &sample_rate_hz, &channels)) { |
| return false; |
| } |
| webrtc::CodecInst codec; |
| AudioCodingModule::Codec( |
| codec_name.c_str(), &codec, sample_rate_hz, channels); |
| codec.pltype = payload_type; |
| if (frame_size_samples > 0) { |
| codec.pacsize = frame_size_samples; |
| } |
| return acm_old_->RegisterSendCodec(codec) == 0; |
| } |
| |
| const AudioEncoder* AudioCodingImpl::GetSenderInfo() const { |
| FATAL() << "Not implemented yet."; |
| return reinterpret_cast<const AudioEncoder*>(NULL); |
| } |
| |
| const CodecInst* AudioCodingImpl::GetSenderCodecInst() { |
| if (acm_old_->SendCodec(¤t_send_codec_) != 0) { |
| return NULL; |
| } |
| return ¤t_send_codec_; |
| } |
| |
| int AudioCodingImpl::Add10MsAudio(const AudioFrame& audio_frame) { |
| acm2::AudioCodingModuleImpl::InputData input_data; |
| if (acm_old_->Add10MsDataInternal(audio_frame, &input_data) != 0) |
| return -1; |
| return acm_old_->Encode(input_data); |
| } |
| |
| const ReceiverInfo* AudioCodingImpl::GetReceiverInfo() const { |
| FATAL() << "Not implemented yet."; |
| return reinterpret_cast<const ReceiverInfo*>(NULL); |
| } |
| |
| bool AudioCodingImpl::RegisterReceiveCodec(AudioDecoder* receive_codec) { |
| FATAL() << "Not implemented yet."; |
| return false; |
| } |
| |
| bool AudioCodingImpl::RegisterReceiveCodec(int decoder_type, |
| uint8_t payload_type) { |
| std::string codec_name; |
| int sample_rate_hz; |
| int channels; |
| if (!MapCodecTypeToParameters( |
| decoder_type, &codec_name, &sample_rate_hz, &channels)) { |
| return false; |
| } |
| webrtc::CodecInst codec; |
| AudioCodingModule::Codec( |
| codec_name.c_str(), &codec, sample_rate_hz, channels); |
| codec.pltype = payload_type; |
| return acm_old_->RegisterReceiveCodec(codec) == 0; |
| } |
| |
| bool AudioCodingImpl::InsertPacket(const uint8_t* incoming_payload, |
| size_t payload_len_bytes, |
| const WebRtcRTPHeader& rtp_info) { |
| return acm_old_->IncomingPacket( |
| incoming_payload, payload_len_bytes, rtp_info) == 0; |
| } |
| |
| bool AudioCodingImpl::InsertPayload(const uint8_t* incoming_payload, |
| size_t payload_len_byte, |
| uint8_t payload_type, |
| uint32_t timestamp) { |
| FATAL() << "Not implemented yet."; |
| return false; |
| } |
| |
| bool AudioCodingImpl::SetMinimumPlayoutDelay(int time_ms) { |
| FATAL() << "Not implemented yet."; |
| return false; |
| } |
| |
| bool AudioCodingImpl::SetMaximumPlayoutDelay(int time_ms) { |
| FATAL() << "Not implemented yet."; |
| return false; |
| } |
| |
| int AudioCodingImpl::LeastRequiredDelayMs() const { |
| FATAL() << "Not implemented yet."; |
| return -1; |
| } |
| |
| bool AudioCodingImpl::PlayoutTimestamp(uint32_t* timestamp) { |
| FATAL() << "Not implemented yet."; |
| return false; |
| } |
| |
| bool AudioCodingImpl::Get10MsAudio(AudioFrame* audio_frame) { |
| return acm_old_->PlayoutData10Ms(playout_frequency_hz_, audio_frame) == 0; |
| } |
| |
| bool AudioCodingImpl::GetNetworkStatistics( |
| NetworkStatistics* network_statistics) { |
| FATAL() << "Not implemented yet."; |
| return false; |
| } |
| |
| bool AudioCodingImpl::EnableNack(size_t max_nack_list_size) { |
| FATAL() << "Not implemented yet."; |
| return false; |
| } |
| |
| void AudioCodingImpl::DisableNack() { |
| // A bug in the linker of Visual Studio 2013 Update 3 prevent us from using |
| // FATAL() here, if we do so then the linker hang when the WPO is turned on. |
| // TODO(sebmarchand): Re-evaluate this when we upgrade the toolchain. |
| } |
| |
| bool AudioCodingImpl::SetVad(bool enable_dtx, |
| bool enable_vad, |
| ACMVADMode vad_mode) { |
| return acm_old_->SetVAD(enable_dtx, enable_vad, vad_mode) == 0; |
| } |
| |
| std::vector<uint16_t> AudioCodingImpl::GetNackList( |
| int round_trip_time_ms) const { |
| return acm_old_->GetNackList(round_trip_time_ms); |
| } |
| |
| void AudioCodingImpl::GetDecodingCallStatistics( |
| AudioDecodingCallStats* call_stats) const { |
| acm_old_->GetDecodingCallStatistics(call_stats); |
| } |
| |
| bool AudioCodingImpl::MapCodecTypeToParameters(int codec_type, |
| std::string* codec_name, |
| int* sample_rate_hz, |
| int* channels) { |
| switch (codec_type) { |
| #ifdef WEBRTC_CODEC_PCM16 |
| case acm2::ACMCodecDB::kPCM16B: |
| *codec_name = "L16"; |
| *sample_rate_hz = 8000; |
| *channels = 1; |
| break; |
| case acm2::ACMCodecDB::kPCM16Bwb: |
| *codec_name = "L16"; |
| *sample_rate_hz = 16000; |
| *channels = 1; |
| break; |
| case acm2::ACMCodecDB::kPCM16Bswb32kHz: |
| *codec_name = "L16"; |
| *sample_rate_hz = 32000; |
| *channels = 1; |
| break; |
| case acm2::ACMCodecDB::kPCM16B_2ch: |
| *codec_name = "L16"; |
| *sample_rate_hz = 8000; |
| *channels = 2; |
| break; |
| case acm2::ACMCodecDB::kPCM16Bwb_2ch: |
| *codec_name = "L16"; |
| *sample_rate_hz = 16000; |
| *channels = 2; |
| break; |
| case acm2::ACMCodecDB::kPCM16Bswb32kHz_2ch: |
| *codec_name = "L16"; |
| *sample_rate_hz = 32000; |
| *channels = 2; |
| break; |
| #endif |
| #if (defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX)) |
| case acm2::ACMCodecDB::kISAC: |
| *codec_name = "ISAC"; |
| *sample_rate_hz = 16000; |
| *channels = 1; |
| break; |
| #endif |
| #ifdef WEBRTC_CODEC_ISAC |
| case acm2::ACMCodecDB::kISACSWB: |
| *codec_name = "ISAC"; |
| *sample_rate_hz = 32000; |
| *channels = 1; |
| break; |
| case acm2::ACMCodecDB::kISACFB: |
| *codec_name = "ISAC"; |
| *sample_rate_hz = 48000; |
| *channels = 1; |
| break; |
| #endif |
| #ifdef WEBRTC_CODEC_ILBC |
| case acm2::ACMCodecDB::kILBC: |
| *codec_name = "ILBC"; |
| *sample_rate_hz = 8000; |
| *channels = 1; |
| break; |
| #endif |
| case acm2::ACMCodecDB::kPCMA: |
| *codec_name = "PCMA"; |
| *sample_rate_hz = 8000; |
| *channels = 1; |
| break; |
| case acm2::ACMCodecDB::kPCMA_2ch: |
| *codec_name = "PCMA"; |
| *sample_rate_hz = 8000; |
| *channels = 2; |
| break; |
| case acm2::ACMCodecDB::kPCMU: |
| *codec_name = "PCMU"; |
| *sample_rate_hz = 8000; |
| *channels = 1; |
| break; |
| case acm2::ACMCodecDB::kPCMU_2ch: |
| *codec_name = "PCMU"; |
| *sample_rate_hz = 8000; |
| *channels = 2; |
| break; |
| #ifdef WEBRTC_CODEC_G722 |
| case acm2::ACMCodecDB::kG722: |
| *codec_name = "G722"; |
| *sample_rate_hz = 16000; |
| *channels = 1; |
| break; |
| case acm2::ACMCodecDB::kG722_2ch: |
| *codec_name = "G722"; |
| *sample_rate_hz = 16000; |
| *channels = 2; |
| break; |
| #endif |
| #ifdef WEBRTC_CODEC_OPUS |
| case acm2::ACMCodecDB::kOpus: |
| *codec_name = "opus"; |
| *sample_rate_hz = 48000; |
| *channels = 2; |
| break; |
| #endif |
| case acm2::ACMCodecDB::kCNNB: |
| *codec_name = "CN"; |
| *sample_rate_hz = 8000; |
| *channels = 1; |
| break; |
| case acm2::ACMCodecDB::kCNWB: |
| *codec_name = "CN"; |
| *sample_rate_hz = 16000; |
| *channels = 1; |
| break; |
| case acm2::ACMCodecDB::kCNSWB: |
| *codec_name = "CN"; |
| *sample_rate_hz = 32000; |
| *channels = 1; |
| break; |
| case acm2::ACMCodecDB::kRED: |
| *codec_name = "red"; |
| *sample_rate_hz = 8000; |
| *channels = 1; |
| break; |
| #ifdef WEBRTC_CODEC_AVT |
| case acm2::ACMCodecDB::kAVT: |
| *codec_name = "telephone-event"; |
| *sample_rate_hz = 8000; |
| *channels = 1; |
| break; |
| #endif |
| default: |
| FATAL() << "Codec type " << codec_type << " not supported."; |
| } |
| return true; |
| } |
| |
| } // namespace webrtc |