| /* |
| * 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/neteq/neteq_impl.h" |
| |
| #include <assert.h> |
| #include <memory.h> // memset |
| |
| #include <algorithm> |
| #include <vector> |
| |
| #include "webrtc/base/checks.h" |
| #include "webrtc/base/logging.h" |
| #include "webrtc/base/safe_conversions.h" |
| #include "webrtc/base/trace_event.h" |
| #include "webrtc/common_audio/signal_processing/include/signal_processing_library.h" |
| #include "webrtc/modules/audio_coding/codecs/audio_decoder.h" |
| #include "webrtc/modules/audio_coding/neteq/accelerate.h" |
| #include "webrtc/modules/audio_coding/neteq/background_noise.h" |
| #include "webrtc/modules/audio_coding/neteq/buffer_level_filter.h" |
| #include "webrtc/modules/audio_coding/neteq/comfort_noise.h" |
| #include "webrtc/modules/audio_coding/neteq/decision_logic.h" |
| #include "webrtc/modules/audio_coding/neteq/decoder_database.h" |
| #include "webrtc/modules/audio_coding/neteq/defines.h" |
| #include "webrtc/modules/audio_coding/neteq/delay_manager.h" |
| #include "webrtc/modules/audio_coding/neteq/delay_peak_detector.h" |
| #include "webrtc/modules/audio_coding/neteq/dtmf_buffer.h" |
| #include "webrtc/modules/audio_coding/neteq/dtmf_tone_generator.h" |
| #include "webrtc/modules/audio_coding/neteq/expand.h" |
| #include "webrtc/modules/audio_coding/neteq/merge.h" |
| #include "webrtc/modules/audio_coding/neteq/nack_tracker.h" |
| #include "webrtc/modules/audio_coding/neteq/normal.h" |
| #include "webrtc/modules/audio_coding/neteq/packet_buffer.h" |
| #include "webrtc/modules/audio_coding/neteq/packet.h" |
| #include "webrtc/modules/audio_coding/neteq/payload_splitter.h" |
| #include "webrtc/modules/audio_coding/neteq/post_decode_vad.h" |
| #include "webrtc/modules/audio_coding/neteq/preemptive_expand.h" |
| #include "webrtc/modules/audio_coding/neteq/sync_buffer.h" |
| #include "webrtc/modules/audio_coding/neteq/tick_timer.h" |
| #include "webrtc/modules/audio_coding/neteq/timestamp_scaler.h" |
| #include "webrtc/modules/include/module_common_types.h" |
| |
| namespace webrtc { |
| |
| NetEqImpl::Dependencies::Dependencies( |
| const NetEq::Config& config, |
| const rtc::scoped_refptr<AudioDecoderFactory>& decoder_factory) |
| : tick_timer(new TickTimer), |
| buffer_level_filter(new BufferLevelFilter), |
| decoder_database(new DecoderDatabase(decoder_factory)), |
| delay_peak_detector(new DelayPeakDetector(tick_timer.get())), |
| delay_manager(new DelayManager(config.max_packets_in_buffer, |
| delay_peak_detector.get(), |
| tick_timer.get())), |
| dtmf_buffer(new DtmfBuffer(config.sample_rate_hz)), |
| dtmf_tone_generator(new DtmfToneGenerator), |
| packet_buffer( |
| new PacketBuffer(config.max_packets_in_buffer, tick_timer.get())), |
| payload_splitter(new PayloadSplitter), |
| timestamp_scaler(new TimestampScaler(*decoder_database)), |
| accelerate_factory(new AccelerateFactory), |
| expand_factory(new ExpandFactory), |
| preemptive_expand_factory(new PreemptiveExpandFactory) {} |
| |
| NetEqImpl::Dependencies::~Dependencies() = default; |
| |
| NetEqImpl::NetEqImpl(const NetEq::Config& config, |
| Dependencies&& deps, |
| bool create_components) |
| : tick_timer_(std::move(deps.tick_timer)), |
| buffer_level_filter_(std::move(deps.buffer_level_filter)), |
| decoder_database_(std::move(deps.decoder_database)), |
| delay_manager_(std::move(deps.delay_manager)), |
| delay_peak_detector_(std::move(deps.delay_peak_detector)), |
| dtmf_buffer_(std::move(deps.dtmf_buffer)), |
| dtmf_tone_generator_(std::move(deps.dtmf_tone_generator)), |
| packet_buffer_(std::move(deps.packet_buffer)), |
| payload_splitter_(std::move(deps.payload_splitter)), |
| timestamp_scaler_(std::move(deps.timestamp_scaler)), |
| vad_(new PostDecodeVad()), |
| expand_factory_(std::move(deps.expand_factory)), |
| accelerate_factory_(std::move(deps.accelerate_factory)), |
| preemptive_expand_factory_(std::move(deps.preemptive_expand_factory)), |
| last_mode_(kModeNormal), |
| decoded_buffer_length_(kMaxFrameSize), |
| decoded_buffer_(new int16_t[decoded_buffer_length_]), |
| playout_timestamp_(0), |
| new_codec_(false), |
| timestamp_(0), |
| reset_decoder_(false), |
| ssrc_(0), |
| first_packet_(true), |
| error_code_(0), |
| decoder_error_code_(0), |
| background_noise_mode_(config.background_noise_mode), |
| playout_mode_(config.playout_mode), |
| enable_fast_accelerate_(config.enable_fast_accelerate), |
| nack_enabled_(false), |
| enable_muted_state_(config.enable_muted_state) { |
| LOG(LS_INFO) << "NetEq config: " << config.ToString(); |
| int fs = config.sample_rate_hz; |
| if (fs != 8000 && fs != 16000 && fs != 32000 && fs != 48000) { |
| LOG(LS_ERROR) << "Sample rate " << fs << " Hz not supported. " << |
| "Changing to 8000 Hz."; |
| fs = 8000; |
| } |
| delay_manager_->SetMaximumDelay(config.max_delay_ms); |
| fs_hz_ = fs; |
| fs_mult_ = fs / 8000; |
| last_output_sample_rate_hz_ = fs; |
| output_size_samples_ = static_cast<size_t>(kOutputSizeMs * 8 * fs_mult_); |
| decoder_frame_length_ = 3 * output_size_samples_; |
| WebRtcSpl_Init(); |
| if (create_components) { |
| SetSampleRateAndChannels(fs, 1); // Default is 1 channel. |
| } |
| RTC_DCHECK(!vad_->enabled()); |
| if (config.enable_post_decode_vad) { |
| vad_->Enable(); |
| } |
| } |
| |
| NetEqImpl::~NetEqImpl() = default; |
| |
| int NetEqImpl::InsertPacket(const WebRtcRTPHeader& rtp_header, |
| rtc::ArrayView<const uint8_t> payload, |
| uint32_t receive_timestamp) { |
| TRACE_EVENT0("webrtc", "NetEqImpl::InsertPacket"); |
| rtc::CritScope lock(&crit_sect_); |
| int error = |
| InsertPacketInternal(rtp_header, payload, receive_timestamp, false); |
| if (error != 0) { |
| error_code_ = error; |
| return kFail; |
| } |
| return kOK; |
| } |
| |
| int NetEqImpl::InsertSyncPacket(const WebRtcRTPHeader& rtp_header, |
| uint32_t receive_timestamp) { |
| rtc::CritScope lock(&crit_sect_); |
| const uint8_t kSyncPayload[] = { 's', 'y', 'n', 'c' }; |
| int error = |
| InsertPacketInternal(rtp_header, kSyncPayload, receive_timestamp, true); |
| |
| if (error != 0) { |
| error_code_ = error; |
| return kFail; |
| } |
| return kOK; |
| } |
| |
| namespace { |
| void SetAudioFrameActivityAndType(bool vad_enabled, |
| NetEqImpl::OutputType type, |
| AudioFrame::VADActivity last_vad_activity, |
| AudioFrame* audio_frame) { |
| switch (type) { |
| case NetEqImpl::OutputType::kNormalSpeech: { |
| audio_frame->speech_type_ = AudioFrame::kNormalSpeech; |
| audio_frame->vad_activity_ = AudioFrame::kVadActive; |
| break; |
| } |
| case NetEqImpl::OutputType::kVadPassive: { |
| // This should only be reached if the VAD is enabled. |
| RTC_DCHECK(vad_enabled); |
| audio_frame->speech_type_ = AudioFrame::kNormalSpeech; |
| audio_frame->vad_activity_ = AudioFrame::kVadPassive; |
| break; |
| } |
| case NetEqImpl::OutputType::kCNG: { |
| audio_frame->speech_type_ = AudioFrame::kCNG; |
| audio_frame->vad_activity_ = AudioFrame::kVadPassive; |
| break; |
| } |
| case NetEqImpl::OutputType::kPLC: { |
| audio_frame->speech_type_ = AudioFrame::kPLC; |
| audio_frame->vad_activity_ = last_vad_activity; |
| break; |
| } |
| case NetEqImpl::OutputType::kPLCCNG: { |
| audio_frame->speech_type_ = AudioFrame::kPLCCNG; |
| audio_frame->vad_activity_ = AudioFrame::kVadPassive; |
| break; |
| } |
| default: |
| RTC_NOTREACHED(); |
| } |
| if (!vad_enabled) { |
| // Always set kVadUnknown when receive VAD is inactive. |
| audio_frame->vad_activity_ = AudioFrame::kVadUnknown; |
| } |
| } |
| } // namespace |
| |
| int NetEqImpl::GetAudio(AudioFrame* audio_frame, bool* muted) { |
| TRACE_EVENT0("webrtc", "NetEqImpl::GetAudio"); |
| rtc::CritScope lock(&crit_sect_); |
| int error = GetAudioInternal(audio_frame, muted); |
| if (error != 0) { |
| error_code_ = error; |
| return kFail; |
| } |
| RTC_DCHECK_EQ( |
| audio_frame->sample_rate_hz_, |
| rtc::checked_cast<int>(audio_frame->samples_per_channel_ * 100)); |
| SetAudioFrameActivityAndType(vad_->enabled(), LastOutputType(), |
| last_vad_activity_, audio_frame); |
| last_vad_activity_ = audio_frame->vad_activity_; |
| last_output_sample_rate_hz_ = audio_frame->sample_rate_hz_; |
| RTC_DCHECK(last_output_sample_rate_hz_ == 8000 || |
| last_output_sample_rate_hz_ == 16000 || |
| last_output_sample_rate_hz_ == 32000 || |
| last_output_sample_rate_hz_ == 48000) |
| << "Unexpected sample rate " << last_output_sample_rate_hz_; |
| return kOK; |
| } |
| |
| int NetEqImpl::RegisterPayloadType(NetEqDecoder codec, |
| const std::string& name, |
| uint8_t rtp_payload_type) { |
| rtc::CritScope lock(&crit_sect_); |
| LOG(LS_VERBOSE) << "RegisterPayloadType " |
| << static_cast<int>(rtp_payload_type) << " " |
| << static_cast<int>(codec); |
| int ret = decoder_database_->RegisterPayload(rtp_payload_type, codec, name); |
| if (ret != DecoderDatabase::kOK) { |
| switch (ret) { |
| case DecoderDatabase::kInvalidRtpPayloadType: |
| error_code_ = kInvalidRtpPayloadType; |
| break; |
| case DecoderDatabase::kCodecNotSupported: |
| error_code_ = kCodecNotSupported; |
| break; |
| case DecoderDatabase::kDecoderExists: |
| error_code_ = kDecoderExists; |
| break; |
| default: |
| error_code_ = kOtherError; |
| } |
| return kFail; |
| } |
| return kOK; |
| } |
| |
| int NetEqImpl::RegisterExternalDecoder(AudioDecoder* decoder, |
| NetEqDecoder codec, |
| const std::string& codec_name, |
| uint8_t rtp_payload_type) { |
| rtc::CritScope lock(&crit_sect_); |
| LOG(LS_VERBOSE) << "RegisterExternalDecoder " |
| << static_cast<int>(rtp_payload_type) << " " |
| << static_cast<int>(codec); |
| if (!decoder) { |
| LOG(LS_ERROR) << "Cannot register external decoder with NULL pointer"; |
| assert(false); |
| return kFail; |
| } |
| int ret = decoder_database_->InsertExternal(rtp_payload_type, codec, |
| codec_name, decoder); |
| if (ret != DecoderDatabase::kOK) { |
| switch (ret) { |
| case DecoderDatabase::kInvalidRtpPayloadType: |
| error_code_ = kInvalidRtpPayloadType; |
| break; |
| case DecoderDatabase::kCodecNotSupported: |
| error_code_ = kCodecNotSupported; |
| break; |
| case DecoderDatabase::kDecoderExists: |
| error_code_ = kDecoderExists; |
| break; |
| case DecoderDatabase::kInvalidSampleRate: |
| error_code_ = kInvalidSampleRate; |
| break; |
| case DecoderDatabase::kInvalidPointer: |
| error_code_ = kInvalidPointer; |
| break; |
| default: |
| error_code_ = kOtherError; |
| } |
| return kFail; |
| } |
| return kOK; |
| } |
| |
| int NetEqImpl::RemovePayloadType(uint8_t rtp_payload_type) { |
| rtc::CritScope lock(&crit_sect_); |
| int ret = decoder_database_->Remove(rtp_payload_type); |
| if (ret == DecoderDatabase::kOK) { |
| return kOK; |
| } else if (ret == DecoderDatabase::kDecoderNotFound) { |
| error_code_ = kDecoderNotFound; |
| } else { |
| error_code_ = kOtherError; |
| } |
| return kFail; |
| } |
| |
| bool NetEqImpl::SetMinimumDelay(int delay_ms) { |
| rtc::CritScope lock(&crit_sect_); |
| if (delay_ms >= 0 && delay_ms < 10000) { |
| assert(delay_manager_.get()); |
| return delay_manager_->SetMinimumDelay(delay_ms); |
| } |
| return false; |
| } |
| |
| bool NetEqImpl::SetMaximumDelay(int delay_ms) { |
| rtc::CritScope lock(&crit_sect_); |
| if (delay_ms >= 0 && delay_ms < 10000) { |
| assert(delay_manager_.get()); |
| return delay_manager_->SetMaximumDelay(delay_ms); |
| } |
| return false; |
| } |
| |
| int NetEqImpl::LeastRequiredDelayMs() const { |
| rtc::CritScope lock(&crit_sect_); |
| assert(delay_manager_.get()); |
| return delay_manager_->least_required_delay_ms(); |
| } |
| |
| int NetEqImpl::SetTargetDelay() { |
| return kNotImplemented; |
| } |
| |
| int NetEqImpl::TargetDelay() { |
| return kNotImplemented; |
| } |
| |
| int NetEqImpl::CurrentDelayMs() const { |
| rtc::CritScope lock(&crit_sect_); |
| if (fs_hz_ == 0) |
| return 0; |
| // Sum up the samples in the packet buffer with the future length of the sync |
| // buffer, and divide the sum by the sample rate. |
| const size_t delay_samples = |
| packet_buffer_->NumSamplesInBuffer(decoder_database_.get(), |
| decoder_frame_length_) + |
| sync_buffer_->FutureLength(); |
| // The division below will truncate. |
| const int delay_ms = |
| static_cast<int>(delay_samples) / rtc::CheckedDivExact(fs_hz_, 1000); |
| return delay_ms; |
| } |
| |
| int NetEqImpl::FilteredCurrentDelayMs() const { |
| rtc::CritScope lock(&crit_sect_); |
| // Calculate the filtered packet buffer level in samples. The value from |
| // |buffer_level_filter_| is in number of packets, represented in Q8. |
| const size_t packet_buffer_samples = |
| (buffer_level_filter_->filtered_current_level() * |
| decoder_frame_length_) >> |
| 8; |
| // Sum up the filtered packet buffer level with the future length of the sync |
| // buffer, and divide the sum by the sample rate. |
| const size_t delay_samples = |
| packet_buffer_samples + sync_buffer_->FutureLength(); |
| // The division below will truncate. The return value is in ms. |
| return static_cast<int>(delay_samples) / rtc::CheckedDivExact(fs_hz_, 1000); |
| } |
| |
| // Deprecated. |
| // TODO(henrik.lundin) Delete. |
| void NetEqImpl::SetPlayoutMode(NetEqPlayoutMode mode) { |
| rtc::CritScope lock(&crit_sect_); |
| if (mode != playout_mode_) { |
| playout_mode_ = mode; |
| CreateDecisionLogic(); |
| } |
| } |
| |
| // Deprecated. |
| // TODO(henrik.lundin) Delete. |
| NetEqPlayoutMode NetEqImpl::PlayoutMode() const { |
| rtc::CritScope lock(&crit_sect_); |
| return playout_mode_; |
| } |
| |
| int NetEqImpl::NetworkStatistics(NetEqNetworkStatistics* stats) { |
| rtc::CritScope lock(&crit_sect_); |
| assert(decoder_database_.get()); |
| const size_t total_samples_in_buffers = |
| packet_buffer_->NumSamplesInBuffer(decoder_database_.get(), |
| decoder_frame_length_) + |
| sync_buffer_->FutureLength(); |
| assert(delay_manager_.get()); |
| assert(decision_logic_.get()); |
| stats_.GetNetworkStatistics(fs_hz_, total_samples_in_buffers, |
| decoder_frame_length_, *delay_manager_.get(), |
| *decision_logic_.get(), stats); |
| return 0; |
| } |
| |
| void NetEqImpl::GetRtcpStatistics(RtcpStatistics* stats) { |
| rtc::CritScope lock(&crit_sect_); |
| if (stats) { |
| rtcp_.GetStatistics(false, stats); |
| } |
| } |
| |
| void NetEqImpl::GetRtcpStatisticsNoReset(RtcpStatistics* stats) { |
| rtc::CritScope lock(&crit_sect_); |
| if (stats) { |
| rtcp_.GetStatistics(true, stats); |
| } |
| } |
| |
| void NetEqImpl::EnableVad() { |
| rtc::CritScope lock(&crit_sect_); |
| assert(vad_.get()); |
| vad_->Enable(); |
| } |
| |
| void NetEqImpl::DisableVad() { |
| rtc::CritScope lock(&crit_sect_); |
| assert(vad_.get()); |
| vad_->Disable(); |
| } |
| |
| rtc::Optional<uint32_t> NetEqImpl::GetPlayoutTimestamp() const { |
| rtc::CritScope lock(&crit_sect_); |
| if (first_packet_ || last_mode_ == kModeRfc3389Cng || |
| last_mode_ == kModeCodecInternalCng) { |
| // We don't have a valid RTP timestamp until we have decoded our first |
| // RTP packet. Also, the RTP timestamp is not accurate while playing CNG, |
| // which is indicated by returning an empty value. |
| return rtc::Optional<uint32_t>(); |
| } |
| return rtc::Optional<uint32_t>( |
| timestamp_scaler_->ToExternal(playout_timestamp_)); |
| } |
| |
| int NetEqImpl::last_output_sample_rate_hz() const { |
| rtc::CritScope lock(&crit_sect_); |
| return last_output_sample_rate_hz_; |
| } |
| |
| int NetEqImpl::SetTargetNumberOfChannels() { |
| return kNotImplemented; |
| } |
| |
| int NetEqImpl::SetTargetSampleRate() { |
| return kNotImplemented; |
| } |
| |
| int NetEqImpl::LastError() const { |
| rtc::CritScope lock(&crit_sect_); |
| return error_code_; |
| } |
| |
| int NetEqImpl::LastDecoderError() { |
| rtc::CritScope lock(&crit_sect_); |
| return decoder_error_code_; |
| } |
| |
| void NetEqImpl::FlushBuffers() { |
| rtc::CritScope lock(&crit_sect_); |
| LOG(LS_VERBOSE) << "FlushBuffers"; |
| packet_buffer_->Flush(); |
| assert(sync_buffer_.get()); |
| assert(expand_.get()); |
| sync_buffer_->Flush(); |
| sync_buffer_->set_next_index(sync_buffer_->next_index() - |
| expand_->overlap_length()); |
| // Set to wait for new codec. |
| first_packet_ = true; |
| } |
| |
| void NetEqImpl::PacketBufferStatistics(int* current_num_packets, |
| int* max_num_packets) const { |
| rtc::CritScope lock(&crit_sect_); |
| packet_buffer_->BufferStat(current_num_packets, max_num_packets); |
| } |
| |
| void NetEqImpl::EnableNack(size_t max_nack_list_size) { |
| rtc::CritScope lock(&crit_sect_); |
| if (!nack_enabled_) { |
| const int kNackThresholdPackets = 2; |
| nack_.reset(NackTracker::Create(kNackThresholdPackets)); |
| nack_enabled_ = true; |
| nack_->UpdateSampleRate(fs_hz_); |
| } |
| nack_->SetMaxNackListSize(max_nack_list_size); |
| } |
| |
| void NetEqImpl::DisableNack() { |
| rtc::CritScope lock(&crit_sect_); |
| nack_.reset(); |
| nack_enabled_ = false; |
| } |
| |
| std::vector<uint16_t> NetEqImpl::GetNackList(int64_t round_trip_time_ms) const { |
| rtc::CritScope lock(&crit_sect_); |
| if (!nack_enabled_) { |
| return std::vector<uint16_t>(); |
| } |
| RTC_DCHECK(nack_.get()); |
| return nack_->GetNackList(round_trip_time_ms); |
| } |
| |
| const SyncBuffer* NetEqImpl::sync_buffer_for_test() const { |
| rtc::CritScope lock(&crit_sect_); |
| return sync_buffer_.get(); |
| } |
| |
| Operations NetEqImpl::last_operation_for_test() const { |
| rtc::CritScope lock(&crit_sect_); |
| return last_operation_; |
| } |
| |
| // Methods below this line are private. |
| |
| int NetEqImpl::InsertPacketInternal(const WebRtcRTPHeader& rtp_header, |
| rtc::ArrayView<const uint8_t> payload, |
| uint32_t receive_timestamp, |
| bool is_sync_packet) { |
| if (payload.empty()) { |
| LOG_F(LS_ERROR) << "payload is empty"; |
| return kInvalidPointer; |
| } |
| // Sanity checks for sync-packets. |
| if (is_sync_packet) { |
| if (decoder_database_->IsDtmf(rtp_header.header.payloadType) || |
| decoder_database_->IsRed(rtp_header.header.payloadType) || |
| decoder_database_->IsComfortNoise(rtp_header.header.payloadType)) { |
| LOG_F(LS_ERROR) << "Sync-packet with an unacceptable payload type " |
| << static_cast<int>(rtp_header.header.payloadType); |
| return kSyncPacketNotAccepted; |
| } |
| if (first_packet_ || !current_rtp_payload_type_ || |
| rtp_header.header.payloadType != *current_rtp_payload_type_ || |
| rtp_header.header.ssrc != ssrc_) { |
| // Even if |current_rtp_payload_type_| is empty, sync-packet isn't |
| // accepted. |
| LOG_F(LS_ERROR) |
| << "Changing codec, SSRC or first packet with sync-packet."; |
| return kSyncPacketNotAccepted; |
| } |
| } |
| PacketList packet_list; |
| RTPHeader main_header; |
| { |
| // Convert to Packet. |
| // Create |packet| within this separate scope, since it should not be used |
| // directly once it's been inserted in the packet list. This way, |packet| |
| // is not defined outside of this block. |
| Packet* packet = new Packet; |
| packet->header.markerBit = false; |
| packet->header.payloadType = rtp_header.header.payloadType; |
| packet->header.sequenceNumber = rtp_header.header.sequenceNumber; |
| packet->header.timestamp = rtp_header.header.timestamp; |
| packet->header.ssrc = rtp_header.header.ssrc; |
| packet->header.numCSRCs = 0; |
| packet->payload.SetData(payload.data(), payload.size()); |
| packet->primary = true; |
| // Waiting time will be set upon inserting the packet in the buffer. |
| RTC_DCHECK(!packet->waiting_time); |
| packet->sync_packet = is_sync_packet; |
| // Insert packet in a packet list. |
| packet_list.push_back(packet); |
| // Save main payloads header for later. |
| memcpy(&main_header, &packet->header, sizeof(main_header)); |
| } |
| |
| bool update_sample_rate_and_channels = false; |
| // Reinitialize NetEq if it's needed (changed SSRC or first call). |
| if ((main_header.ssrc != ssrc_) || first_packet_) { |
| // Note: |first_packet_| will be cleared further down in this method, once |
| // the packet has been successfully inserted into the packet buffer. |
| |
| rtcp_.Init(main_header.sequenceNumber); |
| |
| // Flush the packet buffer and DTMF buffer. |
| packet_buffer_->Flush(); |
| dtmf_buffer_->Flush(); |
| |
| // Store new SSRC. |
| ssrc_ = main_header.ssrc; |
| |
| // Update audio buffer timestamp. |
| sync_buffer_->IncreaseEndTimestamp(main_header.timestamp - timestamp_); |
| |
| // Update codecs. |
| timestamp_ = main_header.timestamp; |
| |
| // Reset timestamp scaling. |
| timestamp_scaler_->Reset(); |
| |
| // Trigger an update of sampling rate and the number of channels. |
| update_sample_rate_and_channels = true; |
| } |
| |
| // Update RTCP statistics, only for regular packets. |
| if (!is_sync_packet) |
| rtcp_.Update(main_header, receive_timestamp); |
| |
| // Check for RED payload type, and separate payloads into several packets. |
| if (decoder_database_->IsRed(main_header.payloadType)) { |
| assert(!is_sync_packet); // We had a sanity check for this. |
| if (payload_splitter_->SplitRed(&packet_list) != PayloadSplitter::kOK) { |
| PacketBuffer::DeleteAllPackets(&packet_list); |
| return kRedundancySplitError; |
| } |
| // Only accept a few RED payloads of the same type as the main data, |
| // DTMF events and CNG. |
| payload_splitter_->CheckRedPayloads(&packet_list, *decoder_database_); |
| // Update the stored main payload header since the main payload has now |
| // changed. |
| memcpy(&main_header, &packet_list.front()->header, sizeof(main_header)); |
| } |
| |
| // Check payload types. |
| if (decoder_database_->CheckPayloadTypes(packet_list) == |
| DecoderDatabase::kDecoderNotFound) { |
| PacketBuffer::DeleteAllPackets(&packet_list); |
| return kUnknownRtpPayloadType; |
| } |
| |
| // Scale timestamp to internal domain (only for some codecs). |
| timestamp_scaler_->ToInternal(&packet_list); |
| |
| // Process DTMF payloads. Cycle through the list of packets, and pick out any |
| // DTMF payloads found. |
| PacketList::iterator it = packet_list.begin(); |
| while (it != packet_list.end()) { |
| Packet* current_packet = (*it); |
| assert(current_packet); |
| assert(!current_packet->payload.empty()); |
| if (decoder_database_->IsDtmf(current_packet->header.payloadType)) { |
| assert(!current_packet->sync_packet); // We had a sanity check for this. |
| DtmfEvent event; |
| int ret = DtmfBuffer::ParseEvent(current_packet->header.timestamp, |
| current_packet->payload.data(), |
| current_packet->payload.size(), &event); |
| if (ret != DtmfBuffer::kOK) { |
| PacketBuffer::DeleteAllPackets(&packet_list); |
| return kDtmfParsingError; |
| } |
| if (dtmf_buffer_->InsertEvent(event) != DtmfBuffer::kOK) { |
| PacketBuffer::DeleteAllPackets(&packet_list); |
| return kDtmfInsertError; |
| } |
| delete current_packet; |
| it = packet_list.erase(it); |
| } else { |
| ++it; |
| } |
| } |
| |
| // Check for FEC in packets, and separate payloads into several packets. |
| int ret = payload_splitter_->SplitFec(&packet_list, decoder_database_.get()); |
| if (ret != PayloadSplitter::kOK) { |
| PacketBuffer::DeleteAllPackets(&packet_list); |
| switch (ret) { |
| case PayloadSplitter::kUnknownPayloadType: |
| return kUnknownRtpPayloadType; |
| default: |
| return kOtherError; |
| } |
| } |
| |
| // Split payloads into smaller chunks. This also verifies that all payloads |
| // are of a known payload type. SplitAudio() method is protected against |
| // sync-packets. |
| ret = payload_splitter_->SplitAudio(&packet_list, *decoder_database_); |
| if (ret != PayloadSplitter::kOK) { |
| PacketBuffer::DeleteAllPackets(&packet_list); |
| switch (ret) { |
| case PayloadSplitter::kUnknownPayloadType: |
| return kUnknownRtpPayloadType; |
| case PayloadSplitter::kFrameSplitError: |
| return kFrameSplitError; |
| default: |
| return kOtherError; |
| } |
| } |
| |
| // Update bandwidth estimate, if the packet is not sync-packet nor comfort |
| // noise. |
| if (!packet_list.empty() && !packet_list.front()->sync_packet && |
| !decoder_database_->IsComfortNoise(main_header.payloadType)) { |
| // The list can be empty here if we got nothing but DTMF payloads. |
| AudioDecoder* decoder = |
| decoder_database_->GetDecoder(main_header.payloadType); |
| assert(decoder); // Should always get a valid object, since we have |
| // already checked that the payload types are known. |
| decoder->IncomingPacket(packet_list.front()->payload.data(), |
| packet_list.front()->payload.size(), |
| packet_list.front()->header.sequenceNumber, |
| packet_list.front()->header.timestamp, |
| receive_timestamp); |
| } |
| |
| if (nack_enabled_) { |
| RTC_DCHECK(nack_); |
| if (update_sample_rate_and_channels) { |
| nack_->Reset(); |
| } |
| nack_->UpdateLastReceivedPacket(packet_list.front()->header.sequenceNumber, |
| packet_list.front()->header.timestamp); |
| } |
| |
| // Insert packets in buffer. |
| const size_t buffer_length_before_insert = |
| packet_buffer_->NumPacketsInBuffer(); |
| ret = packet_buffer_->InsertPacketList( |
| &packet_list, |
| *decoder_database_, |
| ¤t_rtp_payload_type_, |
| ¤t_cng_rtp_payload_type_); |
| if (ret == PacketBuffer::kFlushed) { |
| // Reset DSP timestamp etc. if packet buffer flushed. |
| new_codec_ = true; |
| update_sample_rate_and_channels = true; |
| } else if (ret != PacketBuffer::kOK) { |
| PacketBuffer::DeleteAllPackets(&packet_list); |
| return kOtherError; |
| } |
| |
| if (first_packet_) { |
| first_packet_ = false; |
| // Update the codec on the next GetAudio call. |
| new_codec_ = true; |
| } |
| |
| if (current_rtp_payload_type_) { |
| RTC_DCHECK(decoder_database_->GetDecoderInfo(*current_rtp_payload_type_)) |
| << "Payload type " << static_cast<int>(*current_rtp_payload_type_) |
| << " is unknown where it shouldn't be"; |
| } |
| |
| if (update_sample_rate_and_channels && !packet_buffer_->Empty()) { |
| // We do not use |current_rtp_payload_type_| to |set payload_type|, but |
| // get the next RTP header from |packet_buffer_| to obtain the payload type. |
| // The reason for it is the following corner case. If NetEq receives a |
| // CNG packet with a sample rate different than the current CNG then it |
| // flushes its buffer, assuming send codec must have been changed. However, |
| // payload type of the hypothetically new send codec is not known. |
| const RTPHeader* rtp_header = packet_buffer_->NextRtpHeader(); |
| assert(rtp_header); |
| int payload_type = rtp_header->payloadType; |
| size_t channels = 1; |
| if (!decoder_database_->IsComfortNoise(payload_type)) { |
| AudioDecoder* decoder = decoder_database_->GetDecoder(payload_type); |
| assert(decoder); // Payloads are already checked to be valid. |
| channels = decoder->Channels(); |
| } |
| const DecoderDatabase::DecoderInfo* decoder_info = |
| decoder_database_->GetDecoderInfo(payload_type); |
| assert(decoder_info); |
| if (decoder_info->SampleRateHz() != fs_hz_ || |
| channels != algorithm_buffer_->Channels()) { |
| SetSampleRateAndChannels(decoder_info->SampleRateHz(), |
| channels); |
| } |
| if (nack_enabled_) { |
| RTC_DCHECK(nack_); |
| // Update the sample rate even if the rate is not new, because of Reset(). |
| nack_->UpdateSampleRate(fs_hz_); |
| } |
| } |
| |
| // TODO(hlundin): Move this code to DelayManager class. |
| const DecoderDatabase::DecoderInfo* dec_info = |
| decoder_database_->GetDecoderInfo(main_header.payloadType); |
| assert(dec_info); // Already checked that the payload type is known. |
| delay_manager_->LastDecoderType(dec_info->codec_type); |
| if (delay_manager_->last_pack_cng_or_dtmf() == 0) { |
| // Calculate the total speech length carried in each packet. |
| const size_t buffer_length_after_insert = |
| packet_buffer_->NumPacketsInBuffer(); |
| |
| if (buffer_length_after_insert > buffer_length_before_insert) { |
| const size_t packet_length_samples = |
| (buffer_length_after_insert - buffer_length_before_insert) * |
| decoder_frame_length_; |
| if (packet_length_samples != decision_logic_->packet_length_samples()) { |
| decision_logic_->set_packet_length_samples(packet_length_samples); |
| delay_manager_->SetPacketAudioLength( |
| rtc::checked_cast<int>((1000 * packet_length_samples) / fs_hz_)); |
| } |
| } |
| |
| // Update statistics. |
| if ((int32_t) (main_header.timestamp - timestamp_) >= 0 && |
| !new_codec_) { |
| // Only update statistics if incoming packet is not older than last played |
| // out packet, and if new codec flag is not set. |
| delay_manager_->Update(main_header.sequenceNumber, main_header.timestamp, |
| fs_hz_); |
| } |
| } else if (delay_manager_->last_pack_cng_or_dtmf() == -1) { |
| // This is first "normal" packet after CNG or DTMF. |
| // Reset packet time counter and measure time until next packet, |
| // but don't update statistics. |
| delay_manager_->set_last_pack_cng_or_dtmf(0); |
| delay_manager_->ResetPacketIatCount(); |
| } |
| return 0; |
| } |
| |
| int NetEqImpl::GetAudioInternal(AudioFrame* audio_frame, bool* muted) { |
| PacketList packet_list; |
| DtmfEvent dtmf_event; |
| Operations operation; |
| bool play_dtmf; |
| *muted = false; |
| tick_timer_->Increment(); |
| stats_.IncreaseCounter(output_size_samples_, fs_hz_); |
| |
| // Check for muted state. |
| if (enable_muted_state_ && expand_->Muted() && packet_buffer_->Empty()) { |
| RTC_DCHECK_EQ(last_mode_, kModeExpand); |
| playout_timestamp_ += static_cast<uint32_t>(output_size_samples_); |
| audio_frame->sample_rate_hz_ = fs_hz_; |
| audio_frame->samples_per_channel_ = output_size_samples_; |
| audio_frame->timestamp_ = |
| first_packet_ |
| ? 0 |
| : timestamp_scaler_->ToExternal(playout_timestamp_) - |
| static_cast<uint32_t>(audio_frame->samples_per_channel_); |
| audio_frame->num_channels_ = sync_buffer_->Channels(); |
| stats_.ExpandedNoiseSamples(output_size_samples_); |
| *muted = true; |
| return 0; |
| } |
| |
| int return_value = GetDecision(&operation, &packet_list, &dtmf_event, |
| &play_dtmf); |
| if (return_value != 0) { |
| last_mode_ = kModeError; |
| return return_value; |
| } |
| |
| AudioDecoder::SpeechType speech_type; |
| int length = 0; |
| int decode_return_value = Decode(&packet_list, &operation, |
| &length, &speech_type); |
| |
| assert(vad_.get()); |
| bool sid_frame_available = |
| (operation == kRfc3389Cng && !packet_list.empty()); |
| vad_->Update(decoded_buffer_.get(), static_cast<size_t>(length), speech_type, |
| sid_frame_available, fs_hz_); |
| |
| if (sid_frame_available || speech_type == AudioDecoder::kComfortNoise) { |
| // Start a new stopwatch since we are decoding a new CNG packet. |
| generated_noise_stopwatch_ = tick_timer_->GetNewStopwatch(); |
| } |
| |
| algorithm_buffer_->Clear(); |
| switch (operation) { |
| case kNormal: { |
| DoNormal(decoded_buffer_.get(), length, speech_type, play_dtmf); |
| break; |
| } |
| case kMerge: { |
| DoMerge(decoded_buffer_.get(), length, speech_type, play_dtmf); |
| break; |
| } |
| case kExpand: { |
| return_value = DoExpand(play_dtmf); |
| break; |
| } |
| case kAccelerate: |
| case kFastAccelerate: { |
| const bool fast_accelerate = |
| enable_fast_accelerate_ && (operation == kFastAccelerate); |
| return_value = DoAccelerate(decoded_buffer_.get(), length, speech_type, |
| play_dtmf, fast_accelerate); |
| break; |
| } |
| case kPreemptiveExpand: { |
| return_value = DoPreemptiveExpand(decoded_buffer_.get(), length, |
| speech_type, play_dtmf); |
| break; |
| } |
| case kRfc3389Cng: |
| case kRfc3389CngNoPacket: { |
| return_value = DoRfc3389Cng(&packet_list, play_dtmf); |
| break; |
| } |
| case kCodecInternalCng: { |
| // This handles the case when there is no transmission and the decoder |
| // should produce internal comfort noise. |
| // TODO(hlundin): Write test for codec-internal CNG. |
| DoCodecInternalCng(decoded_buffer_.get(), length); |
| break; |
| } |
| case kDtmf: { |
| // TODO(hlundin): Write test for this. |
| return_value = DoDtmf(dtmf_event, &play_dtmf); |
| break; |
| } |
| case kAlternativePlc: { |
| // TODO(hlundin): Write test for this. |
| DoAlternativePlc(false); |
| break; |
| } |
| case kAlternativePlcIncreaseTimestamp: { |
| // TODO(hlundin): Write test for this. |
| DoAlternativePlc(true); |
| break; |
| } |
| case kAudioRepetitionIncreaseTimestamp: { |
| // TODO(hlundin): Write test for this. |
| sync_buffer_->IncreaseEndTimestamp( |
| static_cast<uint32_t>(output_size_samples_)); |
| // Skipping break on purpose. Execution should move on into the |
| // next case. |
| FALLTHROUGH(); |
| } |
| case kAudioRepetition: { |
| // TODO(hlundin): Write test for this. |
| // Copy last |output_size_samples_| from |sync_buffer_| to |
| // |algorithm_buffer|. |
| algorithm_buffer_->PushBackFromIndex( |
| *sync_buffer_, sync_buffer_->Size() - output_size_samples_); |
| expand_->Reset(); |
| break; |
| } |
| case kUndefined: { |
| LOG(LS_ERROR) << "Invalid operation kUndefined."; |
| assert(false); // This should not happen. |
| last_mode_ = kModeError; |
| return kInvalidOperation; |
| } |
| } // End of switch. |
| last_operation_ = operation; |
| if (return_value < 0) { |
| return return_value; |
| } |
| |
| if (last_mode_ != kModeRfc3389Cng) { |
| comfort_noise_->Reset(); |
| } |
| |
| // Copy from |algorithm_buffer| to |sync_buffer_|. |
| sync_buffer_->PushBack(*algorithm_buffer_); |
| |
| // Extract data from |sync_buffer_| to |output|. |
| size_t num_output_samples_per_channel = output_size_samples_; |
| size_t num_output_samples = output_size_samples_ * sync_buffer_->Channels(); |
| if (num_output_samples > AudioFrame::kMaxDataSizeSamples) { |
| LOG(LS_WARNING) << "Output array is too short. " |
| << AudioFrame::kMaxDataSizeSamples << " < " |
| << output_size_samples_ << " * " |
| << sync_buffer_->Channels(); |
| num_output_samples = AudioFrame::kMaxDataSizeSamples; |
| num_output_samples_per_channel = |
| AudioFrame::kMaxDataSizeSamples / sync_buffer_->Channels(); |
| } |
| sync_buffer_->GetNextAudioInterleaved(num_output_samples_per_channel, |
| audio_frame); |
| audio_frame->sample_rate_hz_ = fs_hz_; |
| if (sync_buffer_->FutureLength() < expand_->overlap_length()) { |
| // The sync buffer should always contain |overlap_length| samples, but now |
| // too many samples have been extracted. Reinstall the |overlap_length| |
| // lookahead by moving the index. |
| const size_t missing_lookahead_samples = |
| expand_->overlap_length() - sync_buffer_->FutureLength(); |
| RTC_DCHECK_GE(sync_buffer_->next_index(), missing_lookahead_samples); |
| sync_buffer_->set_next_index(sync_buffer_->next_index() - |
| missing_lookahead_samples); |
| } |
| if (audio_frame->samples_per_channel_ != output_size_samples_) { |
| LOG(LS_ERROR) << "audio_frame->samples_per_channel_ (" |
| << audio_frame->samples_per_channel_ |
| << ") != output_size_samples_ (" << output_size_samples_ |
| << ")"; |
| // TODO(minyue): treatment of under-run, filling zeros |
| memset(audio_frame->data_, 0, num_output_samples * sizeof(int16_t)); |
| return kSampleUnderrun; |
| } |
| |
| // Should always have overlap samples left in the |sync_buffer_|. |
| RTC_DCHECK_GE(sync_buffer_->FutureLength(), expand_->overlap_length()); |
| |
| if (play_dtmf) { |
| return_value = |
| DtmfOverdub(dtmf_event, sync_buffer_->Channels(), audio_frame->data_); |
| } |
| |
| // Update the background noise parameters if last operation wrote data |
| // straight from the decoder to the |sync_buffer_|. That is, none of the |
| // operations that modify the signal can be followed by a parameter update. |
| if ((last_mode_ == kModeNormal) || |
| (last_mode_ == kModeAccelerateFail) || |
| (last_mode_ == kModePreemptiveExpandFail) || |
| (last_mode_ == kModeRfc3389Cng) || |
| (last_mode_ == kModeCodecInternalCng)) { |
| background_noise_->Update(*sync_buffer_, *vad_.get()); |
| } |
| |
| if (operation == kDtmf) { |
| // DTMF data was written the end of |sync_buffer_|. |
| // Update index to end of DTMF data in |sync_buffer_|. |
| sync_buffer_->set_dtmf_index(sync_buffer_->Size()); |
| } |
| |
| if (last_mode_ != kModeExpand) { |
| // If last operation was not expand, calculate the |playout_timestamp_| from |
| // the |sync_buffer_|. However, do not update the |playout_timestamp_| if it |
| // would be moved "backwards". |
| uint32_t temp_timestamp = sync_buffer_->end_timestamp() - |
| static_cast<uint32_t>(sync_buffer_->FutureLength()); |
| if (static_cast<int32_t>(temp_timestamp - playout_timestamp_) > 0) { |
| playout_timestamp_ = temp_timestamp; |
| } |
| } else { |
| // Use dead reckoning to estimate the |playout_timestamp_|. |
| playout_timestamp_ += static_cast<uint32_t>(output_size_samples_); |
| } |
| // Set the timestamp in the audio frame to zero before the first packet has |
| // been inserted. Otherwise, subtract the frame size in samples to get the |
| // timestamp of the first sample in the frame (playout_timestamp_ is the |
| // last + 1). |
| audio_frame->timestamp_ = |
| first_packet_ |
| ? 0 |
| : timestamp_scaler_->ToExternal(playout_timestamp_) - |
| static_cast<uint32_t>(audio_frame->samples_per_channel_); |
| |
| if (!(last_mode_ == kModeRfc3389Cng || |
| last_mode_ == kModeCodecInternalCng || |
| last_mode_ == kModeExpand)) { |
| generated_noise_stopwatch_.reset(); |
| } |
| |
| if (decode_return_value) return decode_return_value; |
| return return_value; |
| } |
| |
| int NetEqImpl::GetDecision(Operations* operation, |
| PacketList* packet_list, |
| DtmfEvent* dtmf_event, |
| bool* play_dtmf) { |
| // Initialize output variables. |
| *play_dtmf = false; |
| *operation = kUndefined; |
| |
| assert(sync_buffer_.get()); |
| uint32_t end_timestamp = sync_buffer_->end_timestamp(); |
| if (!new_codec_) { |
| const uint32_t five_seconds_samples = 5 * fs_hz_; |
| packet_buffer_->DiscardOldPackets(end_timestamp, five_seconds_samples); |
| } |
| const RTPHeader* header = packet_buffer_->NextRtpHeader(); |
| |
| RTC_DCHECK(!generated_noise_stopwatch_ || |
| generated_noise_stopwatch_->ElapsedTicks() >= 1); |
| uint64_t generated_noise_samples = |
| generated_noise_stopwatch_ |
| ? (generated_noise_stopwatch_->ElapsedTicks() - 1) * |
| output_size_samples_ + |
| decision_logic_->noise_fast_forward() |
| : 0; |
| |
| if (decision_logic_->CngRfc3389On() || last_mode_ == kModeRfc3389Cng) { |
| // Because of timestamp peculiarities, we have to "manually" disallow using |
| // a CNG packet with the same timestamp as the one that was last played. |
| // This can happen when using redundancy and will cause the timing to shift. |
| while (header && decoder_database_->IsComfortNoise(header->payloadType) && |
| (end_timestamp >= header->timestamp || |
| end_timestamp + generated_noise_samples > header->timestamp)) { |
| // Don't use this packet, discard it. |
| if (packet_buffer_->DiscardNextPacket() != PacketBuffer::kOK) { |
| assert(false); // Must be ok by design. |
| } |
| // Check buffer again. |
| if (!new_codec_) { |
| packet_buffer_->DiscardOldPackets(end_timestamp, 5 * fs_hz_); |
| } |
| header = packet_buffer_->NextRtpHeader(); |
| } |
| } |
| |
| assert(expand_.get()); |
| const int samples_left = static_cast<int>(sync_buffer_->FutureLength() - |
| expand_->overlap_length()); |
| if (last_mode_ == kModeAccelerateSuccess || |
| last_mode_ == kModeAccelerateLowEnergy || |
| last_mode_ == kModePreemptiveExpandSuccess || |
| last_mode_ == kModePreemptiveExpandLowEnergy) { |
| // Subtract (samples_left + output_size_samples_) from sampleMemory. |
| decision_logic_->AddSampleMemory( |
| -(samples_left + rtc::checked_cast<int>(output_size_samples_))); |
| } |
| |
| // Check if it is time to play a DTMF event. |
| if (dtmf_buffer_->GetEvent( |
| static_cast<uint32_t>( |
| end_timestamp + generated_noise_samples), |
| dtmf_event)) { |
| *play_dtmf = true; |
| } |
| |
| // Get instruction. |
| assert(sync_buffer_.get()); |
| assert(expand_.get()); |
| generated_noise_samples = |
| generated_noise_stopwatch_ |
| ? generated_noise_stopwatch_->ElapsedTicks() * output_size_samples_ + |
| decision_logic_->noise_fast_forward() |
| : 0; |
| *operation = decision_logic_->GetDecision( |
| *sync_buffer_, *expand_, decoder_frame_length_, header, last_mode_, |
| *play_dtmf, generated_noise_samples, &reset_decoder_); |
| |
| // Check if we already have enough samples in the |sync_buffer_|. If so, |
| // change decision to normal, unless the decision was merge, accelerate, or |
| // preemptive expand. |
| if (samples_left >= rtc::checked_cast<int>(output_size_samples_) && |
| *operation != kMerge && |
| *operation != kAccelerate && |
| *operation != kFastAccelerate && |
| *operation != kPreemptiveExpand) { |
| *operation = kNormal; |
| return 0; |
| } |
| |
| decision_logic_->ExpandDecision(*operation); |
| |
| // Check conditions for reset. |
| if (new_codec_ || *operation == kUndefined) { |
| // The only valid reason to get kUndefined is that new_codec_ is set. |
| assert(new_codec_); |
| if (*play_dtmf && !header) { |
| timestamp_ = dtmf_event->timestamp; |
| } else { |
| if (!header) { |
| LOG(LS_ERROR) << "Packet missing where it shouldn't."; |
| return -1; |
| } |
| timestamp_ = header->timestamp; |
| if (*operation == kRfc3389CngNoPacket && |
| decoder_database_->IsComfortNoise(header->payloadType)) { |
| // Change decision to CNG packet, since we do have a CNG packet, but it |
| // was considered too early to use. Now, use it anyway. |
| *operation = kRfc3389Cng; |
| } else if (*operation != kRfc3389Cng) { |
| *operation = kNormal; |
| } |
| } |
| // Adjust |sync_buffer_| timestamp before setting |end_timestamp| to the |
| // new value. |
| sync_buffer_->IncreaseEndTimestamp(timestamp_ - end_timestamp); |
| end_timestamp = timestamp_; |
| new_codec_ = false; |
| decision_logic_->SoftReset(); |
| buffer_level_filter_->Reset(); |
| delay_manager_->Reset(); |
| stats_.ResetMcu(); |
| } |
| |
| size_t required_samples = output_size_samples_; |
| const size_t samples_10_ms = static_cast<size_t>(80 * fs_mult_); |
| const size_t samples_20_ms = 2 * samples_10_ms; |
| const size_t samples_30_ms = 3 * samples_10_ms; |
| |
| switch (*operation) { |
| case kExpand: { |
| timestamp_ = end_timestamp; |
| return 0; |
| } |
| case kRfc3389CngNoPacket: |
| case kCodecInternalCng: { |
| return 0; |
| } |
| case kDtmf: { |
| // TODO(hlundin): Write test for this. |
| // Update timestamp. |
| timestamp_ = end_timestamp; |
| const uint64_t generated_noise_samples = |
| generated_noise_stopwatch_ |
| ? generated_noise_stopwatch_->ElapsedTicks() * |
| output_size_samples_ + |
| decision_logic_->noise_fast_forward() |
| : 0; |
| if (generated_noise_samples > 0 && last_mode_ != kModeDtmf) { |
| // Make a jump in timestamp due to the recently played comfort noise. |
| uint32_t timestamp_jump = |
| static_cast<uint32_t>(generated_noise_samples); |
| sync_buffer_->IncreaseEndTimestamp(timestamp_jump); |
| timestamp_ += timestamp_jump; |
| } |
| return 0; |
| } |
| case kAccelerate: |
| case kFastAccelerate: { |
| // In order to do an accelerate we need at least 30 ms of audio data. |
| if (samples_left >= static_cast<int>(samples_30_ms)) { |
| // Already have enough data, so we do not need to extract any more. |
| decision_logic_->set_sample_memory(samples_left); |
| decision_logic_->set_prev_time_scale(true); |
| return 0; |
| } else if (samples_left >= static_cast<int>(samples_10_ms) && |
| decoder_frame_length_ >= samples_30_ms) { |
| // Avoid decoding more data as it might overflow the playout buffer. |
| *operation = kNormal; |
| return 0; |
| } else if (samples_left < static_cast<int>(samples_20_ms) && |
| decoder_frame_length_ < samples_30_ms) { |
| // Build up decoded data by decoding at least 20 ms of audio data. Do |
| // not perform accelerate yet, but wait until we only need to do one |
| // decoding. |
| required_samples = 2 * output_size_samples_; |
| *operation = kNormal; |
| } |
| // If none of the above is true, we have one of two possible situations: |
| // (1) 20 ms <= samples_left < 30 ms and decoder_frame_length_ < 30 ms; or |
| // (2) samples_left < 10 ms and decoder_frame_length_ >= 30 ms. |
| // In either case, we move on with the accelerate decision, and decode one |
| // frame now. |
| break; |
| } |
| case kPreemptiveExpand: { |
| // In order to do a preemptive expand we need at least 30 ms of decoded |
| // audio data. |
| if ((samples_left >= static_cast<int>(samples_30_ms)) || |
| (samples_left >= static_cast<int>(samples_10_ms) && |
| decoder_frame_length_ >= samples_30_ms)) { |
| // Already have enough data, so we do not need to extract any more. |
| // Or, avoid decoding more data as it might overflow the playout buffer. |
| // Still try preemptive expand, though. |
| decision_logic_->set_sample_memory(samples_left); |
| decision_logic_->set_prev_time_scale(true); |
| return 0; |
| } |
| if (samples_left < static_cast<int>(samples_20_ms) && |
| decoder_frame_length_ < samples_30_ms) { |
| // Build up decoded data by decoding at least 20 ms of audio data. |
| // Still try to perform preemptive expand. |
| required_samples = 2 * output_size_samples_; |
| } |
| // Move on with the preemptive expand decision. |
| break; |
| } |
| case kMerge: { |
| required_samples = |
| std::max(merge_->RequiredFutureSamples(), required_samples); |
| break; |
| } |
| default: { |
| // Do nothing. |
| } |
| } |
| |
| // Get packets from buffer. |
| int extracted_samples = 0; |
| if (header && |
| *operation != kAlternativePlc && |
| *operation != kAlternativePlcIncreaseTimestamp && |
| *operation != kAudioRepetition && |
| *operation != kAudioRepetitionIncreaseTimestamp) { |
| sync_buffer_->IncreaseEndTimestamp(header->timestamp - end_timestamp); |
| if (decision_logic_->CngOff()) { |
| // Adjustment of timestamp only corresponds to an actual packet loss |
| // if comfort noise is not played. If comfort noise was just played, |
| // this adjustment of timestamp is only done to get back in sync with the |
| // stream timestamp; no loss to report. |
| stats_.LostSamples(header->timestamp - end_timestamp); |
| } |
| |
| if (*operation != kRfc3389Cng) { |
| // We are about to decode and use a non-CNG packet. |
| decision_logic_->SetCngOff(); |
| } |
| |
| extracted_samples = ExtractPackets(required_samples, packet_list); |
| if (extracted_samples < 0) { |
| return kPacketBufferCorruption; |
| } |
| } |
| |
| if (*operation == kAccelerate || *operation == kFastAccelerate || |
| *operation == kPreemptiveExpand) { |
| decision_logic_->set_sample_memory(samples_left + extracted_samples); |
| decision_logic_->set_prev_time_scale(true); |
| } |
| |
| if (*operation == kAccelerate || *operation == kFastAccelerate) { |
| // Check that we have enough data (30ms) to do accelerate. |
| if (extracted_samples + samples_left < static_cast<int>(samples_30_ms)) { |
| // TODO(hlundin): Write test for this. |
| // Not enough, do normal operation instead. |
| *operation = kNormal; |
| } |
| } |
| |
| timestamp_ = end_timestamp; |
| return 0; |
| } |
| |
| int NetEqImpl::Decode(PacketList* packet_list, Operations* operation, |
| int* decoded_length, |
| AudioDecoder::SpeechType* speech_type) { |
| *speech_type = AudioDecoder::kSpeech; |
| |
| // When packet_list is empty, we may be in kCodecInternalCng mode, and for |
| // that we use current active decoder. |
| AudioDecoder* decoder = decoder_database_->GetActiveDecoder(); |
| |
| if (!packet_list->empty()) { |
| const Packet* packet = packet_list->front(); |
| uint8_t payload_type = packet->header.payloadType; |
| if (!decoder_database_->IsComfortNoise(payload_type)) { |
| decoder = decoder_database_->GetDecoder(payload_type); |
| assert(decoder); |
| if (!decoder) { |
| LOG(LS_WARNING) << "Unknown payload type " |
| << static_cast<int>(payload_type); |
| PacketBuffer::DeleteAllPackets(packet_list); |
| return kDecoderNotFound; |
| } |
| bool decoder_changed; |
| decoder_database_->SetActiveDecoder(payload_type, &decoder_changed); |
| if (decoder_changed) { |
| // We have a new decoder. Re-init some values. |
| const DecoderDatabase::DecoderInfo* decoder_info = decoder_database_ |
| ->GetDecoderInfo(payload_type); |
| assert(decoder_info); |
| if (!decoder_info) { |
| LOG(LS_WARNING) << "Unknown payload type " |
| << static_cast<int>(payload_type); |
| PacketBuffer::DeleteAllPackets(packet_list); |
| return kDecoderNotFound; |
| } |
| // If sampling rate or number of channels has changed, we need to make |
| // a reset. |
| if (decoder_info->SampleRateHz() != fs_hz_ || |
| decoder->Channels() != algorithm_buffer_->Channels()) { |
| // TODO(tlegrand): Add unittest to cover this event. |
| SetSampleRateAndChannels(decoder_info->SampleRateHz(), |
| decoder->Channels()); |
| } |
| sync_buffer_->set_end_timestamp(timestamp_); |
| playout_timestamp_ = timestamp_; |
| } |
| } |
| } |
| |
| if (reset_decoder_) { |
| // TODO(hlundin): Write test for this. |
| if (decoder) |
| decoder->Reset(); |
| |
| // Reset comfort noise decoder. |
| ComfortNoiseDecoder* cng_decoder = decoder_database_->GetActiveCngDecoder(); |
| if (cng_decoder) |
| cng_decoder->Reset(); |
| |
| reset_decoder_ = false; |
| } |
| |
| *decoded_length = 0; |
| // Update codec-internal PLC state. |
| if ((*operation == kMerge) && decoder && decoder->HasDecodePlc()) { |
| decoder->DecodePlc(1, &decoded_buffer_[*decoded_length]); |
| } |
| |
| int return_value; |
| if (*operation == kCodecInternalCng) { |
| RTC_DCHECK(packet_list->empty()); |
| return_value = DecodeCng(decoder, decoded_length, speech_type); |
| } else { |
| return_value = DecodeLoop(packet_list, *operation, decoder, |
| decoded_length, speech_type); |
| } |
| |
| if (*decoded_length < 0) { |
| // Error returned from the decoder. |
| *decoded_length = 0; |
| sync_buffer_->IncreaseEndTimestamp( |
| static_cast<uint32_t>(decoder_frame_length_)); |
| int error_code = 0; |
| if (decoder) |
| error_code = decoder->ErrorCode(); |
| if (error_code != 0) { |
| // Got some error code from the decoder. |
| decoder_error_code_ = error_code; |
| return_value = kDecoderErrorCode; |
| LOG(LS_WARNING) << "Decoder returned error code: " << error_code; |
| } else { |
| // Decoder does not implement error codes. Return generic error. |
| return_value = kOtherDecoderError; |
| LOG(LS_WARNING) << "Decoder error (no error code)"; |
| } |
| *operation = kExpand; // Do expansion to get data instead. |
| } |
| if (*speech_type != AudioDecoder::kComfortNoise) { |
| // Don't increment timestamp if codec returned CNG speech type |
| // since in this case, the we will increment the CNGplayedTS counter. |
| // Increase with number of samples per channel. |
| assert(*decoded_length == 0 || |
| (decoder && decoder->Channels() == sync_buffer_->Channels())); |
| sync_buffer_->IncreaseEndTimestamp( |
| *decoded_length / static_cast<int>(sync_buffer_->Channels())); |
| } |
| return return_value; |
| } |
| |
| int NetEqImpl::DecodeCng(AudioDecoder* decoder, int* decoded_length, |
| AudioDecoder::SpeechType* speech_type) { |
| if (!decoder) { |
| // This happens when active decoder is not defined. |
| *decoded_length = -1; |
| return 0; |
| } |
| |
| while (*decoded_length < rtc::checked_cast<int>(output_size_samples_)) { |
| const int length = decoder->Decode( |
| nullptr, 0, fs_hz_, |
| (decoded_buffer_length_ - *decoded_length) * sizeof(int16_t), |
| &decoded_buffer_[*decoded_length], speech_type); |
| if (length > 0) { |
| *decoded_length += length; |
| } else { |
| // Error. |
| LOG(LS_WARNING) << "Failed to decode CNG"; |
| *decoded_length = -1; |
| break; |
| } |
| if (*decoded_length > static_cast<int>(decoded_buffer_length_)) { |
| // Guard against overflow. |
| LOG(LS_WARNING) << "Decoded too much CNG."; |
| return kDecodedTooMuch; |
| } |
| } |
| return 0; |
| } |
| |
| int NetEqImpl::DecodeLoop(PacketList* packet_list, const Operations& operation, |
| AudioDecoder* decoder, int* decoded_length, |
| AudioDecoder::SpeechType* speech_type) { |
| Packet* packet = NULL; |
| if (!packet_list->empty()) { |
| packet = packet_list->front(); |
| } |
| |
| // Do decoding. |
| while (packet && |
| !decoder_database_->IsComfortNoise(packet->header.payloadType)) { |
| assert(decoder); // At this point, we must have a decoder object. |
| // The number of channels in the |sync_buffer_| should be the same as the |
| // number decoder channels. |
| assert(sync_buffer_->Channels() == decoder->Channels()); |
| assert(decoded_buffer_length_ >= kMaxFrameSize * decoder->Channels()); |
| assert(operation == kNormal || operation == kAccelerate || |
| operation == kFastAccelerate || operation == kMerge || |
| operation == kPreemptiveExpand); |
| packet_list->pop_front(); |
| const size_t payload_length = packet->payload.size(); |
| int decode_length; |
| if (packet->sync_packet) { |
| // Decode to silence with the same frame size as the last decode. |
| memset(&decoded_buffer_[*decoded_length], 0, |
| decoder_frame_length_ * decoder->Channels() * |
| sizeof(decoded_buffer_[0])); |
| decode_length = rtc::checked_cast<int>(decoder_frame_length_); |
| } else if (!packet->primary) { |
| // This is a redundant payload; call the special decoder method. |
| decode_length = decoder->DecodeRedundant( |
| packet->payload.data(), packet->payload.size(), fs_hz_, |
| (decoded_buffer_length_ - *decoded_length) * sizeof(int16_t), |
| &decoded_buffer_[*decoded_length], speech_type); |
| } else { |
| decode_length = decoder->Decode( |
| packet->payload.data(), packet->payload.size(), fs_hz_, |
| (decoded_buffer_length_ - *decoded_length) * sizeof(int16_t), |
| &decoded_buffer_[*decoded_length], speech_type); |
| } |
| |
| delete packet; |
| packet = NULL; |
| if (decode_length > 0) { |
| *decoded_length += decode_length; |
| // Update |decoder_frame_length_| with number of samples per channel. |
| decoder_frame_length_ = |
| static_cast<size_t>(decode_length) / decoder->Channels(); |
| } else if (decode_length < 0) { |
| // Error. |
| LOG(LS_WARNING) << "Decode " << decode_length << " " << payload_length; |
| *decoded_length = -1; |
| PacketBuffer::DeleteAllPackets(packet_list); |
| break; |
| } |
| if (*decoded_length > static_cast<int>(decoded_buffer_length_)) { |
| // Guard against overflow. |
| LOG(LS_WARNING) << "Decoded too much."; |
| PacketBuffer::DeleteAllPackets(packet_list); |
| return kDecodedTooMuch; |
| } |
| if (!packet_list->empty()) { |
| packet = packet_list->front(); |
| } else { |
| packet = NULL; |
| } |
| } // End of decode loop. |
| |
| // If the list is not empty at this point, either a decoding error terminated |
| // the while-loop, or list must hold exactly one CNG packet. |
| assert(packet_list->empty() || *decoded_length < 0 || |
| (packet_list->size() == 1 && packet && |
| decoder_database_->IsComfortNoise(packet->header.payloadType))); |
| return 0; |
| } |
| |
| void NetEqImpl::DoNormal(const int16_t* decoded_buffer, size_t decoded_length, |
| AudioDecoder::SpeechType speech_type, bool play_dtmf) { |
| assert(normal_.get()); |
| assert(mute_factor_array_.get()); |
| normal_->Process(decoded_buffer, decoded_length, last_mode_, |
| mute_factor_array_.get(), algorithm_buffer_.get()); |
| if (decoded_length != 0) { |
| last_mode_ = kModeNormal; |
| } |
| |
| // If last packet was decoded as an inband CNG, set mode to CNG instead. |
| if ((speech_type == AudioDecoder::kComfortNoise) |
| || ((last_mode_ == kModeCodecInternalCng) |
| && (decoded_length == 0))) { |
| // TODO(hlundin): Remove second part of || statement above. |
| last_mode_ = kModeCodecInternalCng; |
| } |
| |
| if (!play_dtmf) { |
| dtmf_tone_generator_->Reset(); |
| } |
| } |
| |
| void NetEqImpl::DoMerge(int16_t* decoded_buffer, size_t decoded_length, |
| AudioDecoder::SpeechType speech_type, bool play_dtmf) { |
| assert(mute_factor_array_.get()); |
| assert(merge_.get()); |
| size_t new_length = merge_->Process(decoded_buffer, decoded_length, |
| mute_factor_array_.get(), |
| algorithm_buffer_.get()); |
| size_t expand_length_correction = new_length - |
| decoded_length / algorithm_buffer_->Channels(); |
| |
| // Update in-call and post-call statistics. |
| if (expand_->MuteFactor(0) == 0) { |
| // Expand generates only noise. |
| stats_.ExpandedNoiseSamples(expand_length_correction); |
| } else { |
| // Expansion generates more than only noise. |
| stats_.ExpandedVoiceSamples(expand_length_correction); |
| } |
| |
| last_mode_ = kModeMerge; |
| // If last packet was decoded as an inband CNG, set mode to CNG instead. |
| if (speech_type == AudioDecoder::kComfortNoise) { |
| last_mode_ = kModeCodecInternalCng; |
| } |
| expand_->Reset(); |
| if (!play_dtmf) { |
| dtmf_tone_generator_->Reset(); |
| } |
| } |
| |
| int NetEqImpl::DoExpand(bool play_dtmf) { |
| while ((sync_buffer_->FutureLength() - expand_->overlap_length()) < |
| output_size_samples_) { |
| algorithm_buffer_->Clear(); |
| int return_value = expand_->Process(algorithm_buffer_.get()); |
| size_t length = algorithm_buffer_->Size(); |
| |
| // Update in-call and post-call statistics. |
| if (expand_->MuteFactor(0) == 0) { |
| // Expand operation generates only noise. |
| stats_.ExpandedNoiseSamples(length); |
| } else { |
| // Expand operation generates more than only noise. |
| stats_.ExpandedVoiceSamples(length); |
| } |
| |
| last_mode_ = kModeExpand; |
| |
| if (return_value < 0) { |
| return return_value; |
| } |
| |
| sync_buffer_->PushBack(*algorithm_buffer_); |
| algorithm_buffer_->Clear(); |
| } |
| if (!play_dtmf) { |
| dtmf_tone_generator_->Reset(); |
| } |
| |
| if (!generated_noise_stopwatch_) { |
| // Start a new stopwatch since we may be covering for a lost CNG packet. |
| generated_noise_stopwatch_ = tick_timer_->GetNewStopwatch(); |
| } |
| |
| return 0; |
| } |
| |
| int NetEqImpl::DoAccelerate(int16_t* decoded_buffer, |
| size_t decoded_length, |
| AudioDecoder::SpeechType speech_type, |
| bool play_dtmf, |
| bool fast_accelerate) { |
| const size_t required_samples = |
| static_cast<size_t>(240 * fs_mult_); // Must have 30 ms. |
| size_t borrowed_samples_per_channel = 0; |
| size_t num_channels = algorithm_buffer_->Channels(); |
| size_t decoded_length_per_channel = decoded_length / num_channels; |
| if (decoded_length_per_channel < required_samples) { |
| // Must move data from the |sync_buffer_| in order to get 30 ms. |
| borrowed_samples_per_channel = static_cast<int>(required_samples - |
| decoded_length_per_channel); |
| memmove(&decoded_buffer[borrowed_samples_per_channel * num_channels], |
| decoded_buffer, |
| sizeof(int16_t) * decoded_length); |
| sync_buffer_->ReadInterleavedFromEnd(borrowed_samples_per_channel, |
| decoded_buffer); |
| decoded_length = required_samples * num_channels; |
| } |
| |
| size_t samples_removed; |
| Accelerate::ReturnCodes return_code = |
| accelerate_->Process(decoded_buffer, decoded_length, fast_accelerate, |
| algorithm_buffer_.get(), &samples_removed); |
| stats_.AcceleratedSamples(samples_removed); |
| switch (return_code) { |
| case Accelerate::kSuccess: |
| last_mode_ = kModeAccelerateSuccess; |
| break; |
| case Accelerate::kSuccessLowEnergy: |
| last_mode_ = kModeAccelerateLowEnergy; |
| break; |
| case Accelerate::kNoStretch: |
| last_mode_ = kModeAccelerateFail; |
| break; |
| case Accelerate::kError: |
| // TODO(hlundin): Map to kModeError instead? |
| last_mode_ = kModeAccelerateFail; |
| return kAccelerateError; |
| } |
| |
| if (borrowed_samples_per_channel > 0) { |
| // Copy borrowed samples back to the |sync_buffer_|. |
| size_t length = algorithm_buffer_->Size(); |
| if (length < borrowed_samples_per_channel) { |
| // This destroys the beginning of the buffer, but will not cause any |
| // problems. |
| sync_buffer_->ReplaceAtIndex(*algorithm_buffer_, |
| sync_buffer_->Size() - |
| borrowed_samples_per_channel); |
| sync_buffer_->PushFrontZeros(borrowed_samples_per_channel - length); |
| algorithm_buffer_->PopFront(length); |
| assert(algorithm_buffer_->Empty()); |
| } else { |
| sync_buffer_->ReplaceAtIndex(*algorithm_buffer_, |
| borrowed_samples_per_channel, |
| sync_buffer_->Size() - |
| borrowed_samples_per_channel); |
| algorithm_buffer_->PopFront(borrowed_samples_per_channel); |
| } |
| } |
| |
| // If last packet was decoded as an inband CNG, set mode to CNG instead. |
| if (speech_type == AudioDecoder::kComfortNoise) { |
| last_mode_ = kModeCodecInternalCng; |
| } |
| if (!play_dtmf) { |
| dtmf_tone_generator_->Reset(); |
| } |
| expand_->Reset(); |
| return 0; |
| } |
| |
| int NetEqImpl::DoPreemptiveExpand(int16_t* decoded_buffer, |
| size_t decoded_length, |
| AudioDecoder::SpeechType speech_type, |
| bool play_dtmf) { |
| const size_t required_samples = |
| static_cast<size_t>(240 * fs_mult_); // Must have 30 ms. |
| size_t num_channels = algorithm_buffer_->Channels(); |
| size_t borrowed_samples_per_channel = 0; |
| size_t old_borrowed_samples_per_channel = 0; |
| size_t decoded_length_per_channel = decoded_length / num_channels; |
| if (decoded_length_per_channel < required_samples) { |
| // Must move data from the |sync_buffer_| in order to get 30 ms. |
| borrowed_samples_per_channel = |
| required_samples - decoded_length_per_channel; |
| // Calculate how many of these were already played out. |
| old_borrowed_samples_per_channel = |
| (borrowed_samples_per_channel > sync_buffer_->FutureLength()) ? |
| (borrowed_samples_per_channel - sync_buffer_->FutureLength()) : 0; |
| memmove(&decoded_buffer[borrowed_samples_per_channel * num_channels], |
| decoded_buffer, |
| sizeof(int16_t) * decoded_length); |
| sync_buffer_->ReadInterleavedFromEnd(borrowed_samples_per_channel, |
| decoded_buffer); |
| decoded_length = required_samples * num_channels; |
| } |
| |
| size_t samples_added; |
| PreemptiveExpand::ReturnCodes return_code = preemptive_expand_->Process( |
| decoded_buffer, decoded_length, |
| old_borrowed_samples_per_channel, |
| algorithm_buffer_.get(), &samples_added); |
| stats_.PreemptiveExpandedSamples(samples_added); |
| switch (return_code) { |
| case PreemptiveExpand::kSuccess: |
| last_mode_ = kModePreemptiveExpandSuccess; |
| break; |
| case PreemptiveExpand::kSuccessLowEnergy: |
| last_mode_ = kModePreemptiveExpandLowEnergy; |
| break; |
| case PreemptiveExpand::kNoStretch: |
| last_mode_ = kModePreemptiveExpandFail; |
| break; |
| case PreemptiveExpand::kError: |
| // TODO(hlundin): Map to kModeError instead? |
| last_mode_ = kModePreemptiveExpandFail; |
| return kPreemptiveExpandError; |
| } |
| |
| if (borrowed_samples_per_channel > 0) { |
| // Copy borrowed samples back to the |sync_buffer_|. |
| sync_buffer_->ReplaceAtIndex( |
| *algorithm_buffer_, borrowed_samples_per_channel, |
| sync_buffer_->Size() - borrowed_samples_per_channel); |
| algorithm_buffer_->PopFront(borrowed_samples_per_channel); |
| } |
| |
| // If last packet was decoded as an inband CNG, set mode to CNG instead. |
| if (speech_type == AudioDecoder::kComfortNoise) { |
| last_mode_ = kModeCodecInternalCng; |
| } |
| if (!play_dtmf) { |
| dtmf_tone_generator_->Reset(); |
| } |
| expand_->Reset(); |
| return 0; |
| } |
| |
| int NetEqImpl::DoRfc3389Cng(PacketList* packet_list, bool play_dtmf) { |
| if (!packet_list->empty()) { |
| // Must have exactly one SID frame at this point. |
| assert(packet_list->size() == 1); |
| Packet* packet = packet_list->front(); |
| packet_list->pop_front(); |
| if (!decoder_database_->IsComfortNoise(packet->header.payloadType)) { |
| LOG(LS_ERROR) << "Trying to decode non-CNG payload as CNG."; |
| return kOtherError; |
| } |
| // UpdateParameters() deletes |packet|. |
| if (comfort_noise_->UpdateParameters(packet) == |
| ComfortNoise::kInternalError) { |
| algorithm_buffer_->Zeros(output_size_samples_); |
| return -comfort_noise_->internal_error_code(); |
| } |
| } |
| int cn_return = comfort_noise_->Generate(output_size_samples_, |
| algorithm_buffer_.get()); |
| expand_->Reset(); |
| last_mode_ = kModeRfc3389Cng; |
| if (!play_dtmf) { |
| dtmf_tone_generator_->Reset(); |
| } |
| if (cn_return == ComfortNoise::kInternalError) { |
| decoder_error_code_ = comfort_noise_->internal_error_code(); |
| return kComfortNoiseErrorCode; |
| } else if (cn_return == ComfortNoise::kUnknownPayloadType) { |
| return kUnknownRtpPayloadType; |
| } |
| return 0; |
| } |
| |
| void NetEqImpl::DoCodecInternalCng(const int16_t* decoded_buffer, |
| size_t decoded_length) { |
| RTC_DCHECK(normal_.get()); |
| RTC_DCHECK(mute_factor_array_.get()); |
| normal_->Process(decoded_buffer, decoded_length, last_mode_, |
| mute_factor_array_.get(), algorithm_buffer_.get()); |
| last_mode_ = kModeCodecInternalCng; |
| expand_->Reset(); |
| } |
| |
| int NetEqImpl::DoDtmf(const DtmfEvent& dtmf_event, bool* play_dtmf) { |
| // This block of the code and the block further down, handling |dtmf_switch| |
| // are commented out. Otherwise playing out-of-band DTMF would fail in VoE |
| // test, DtmfTest.ManualSuccessfullySendsOutOfBandTelephoneEvents. This is |
| // equivalent to |dtmf_switch| always be false. |
| // |
| // See http://webrtc-codereview.appspot.com/1195004/ for discussion |
| // On this issue. This change might cause some glitches at the point of |
| // switch from audio to DTMF. Issue 1545 is filed to track this. |
| // |
| // bool dtmf_switch = false; |
| // if ((last_mode_ != kModeDtmf) && dtmf_tone_generator_->initialized()) { |
| // // Special case; see below. |
| // // We must catch this before calling Generate, since |initialized| is |
| // // modified in that call. |
| // dtmf_switch = true; |
| // } |
| |
| int dtmf_return_value = 0; |
| if (!dtmf_tone_generator_->initialized()) { |
| // Initialize if not already done. |
| dtmf_return_value = dtmf_tone_generator_->Init(fs_hz_, dtmf_event.event_no, |
| dtmf_event.volume); |
| } |
| |
| if (dtmf_return_value == 0) { |
| // Generate DTMF signal. |
| dtmf_return_value = dtmf_tone_generator_->Generate(output_size_samples_, |
| algorithm_buffer_.get()); |
| } |
| |
| if (dtmf_return_value < 0) { |
| algorithm_buffer_->Zeros(output_size_samples_); |
| return dtmf_return_value; |
| } |
| |
| // if (dtmf_switch) { |
| // // This is the special case where the previous operation was DTMF |
| // // overdub, but the current instruction is "regular" DTMF. We must make |
| // // sure that the DTMF does not have any discontinuities. The first DTMF |
| // // sample that we generate now must be played out immediately, therefore |
| // // it must be copied to the speech buffer. |
| // // TODO(hlundin): This code seems incorrect. (Legacy.) Write test and |
| // // verify correct operation. |
| // assert(false); |
| // // Must generate enough data to replace all of the |sync_buffer_| |
| // // "future". |
| // int required_length = sync_buffer_->FutureLength(); |
| // assert(dtmf_tone_generator_->initialized()); |
| // dtmf_return_value = dtmf_tone_generator_->Generate(required_length, |
| // algorithm_buffer_); |
| // assert((size_t) required_length == algorithm_buffer_->Size()); |
| // if (dtmf_return_value < 0) { |
| // algorithm_buffer_->Zeros(output_size_samples_); |
| // return dtmf_return_value; |
| // } |
| // |
| // // Overwrite the "future" part of the speech buffer with the new DTMF |
| // // data. |
| // // TODO(hlundin): It seems that this overwriting has gone lost. |
| // // Not adapted for multi-channel yet. |
| // assert(algorithm_buffer_->Channels() == 1); |
| // if (algorithm_buffer_->Channels() != 1) { |
| // LOG(LS_WARNING) << "DTMF not supported for more than one channel"; |
| // return kStereoNotSupported; |
| // } |
| // // Shuffle the remaining data to the beginning of algorithm buffer. |
| // algorithm_buffer_->PopFront(sync_buffer_->FutureLength()); |
| // } |
| |
| sync_buffer_->IncreaseEndTimestamp( |
| static_cast<uint32_t>(output_size_samples_)); |
| expand_->Reset(); |
| last_mode_ = kModeDtmf; |
| |
| // Set to false because the DTMF is already in the algorithm buffer. |
| *play_dtmf = false; |
| return 0; |
| } |
| |
| void NetEqImpl::DoAlternativePlc(bool increase_timestamp) { |
| AudioDecoder* decoder = decoder_database_->GetActiveDecoder(); |
| size_t length; |
| if (decoder && decoder->HasDecodePlc()) { |
| // Use the decoder's packet-loss concealment. |
| // TODO(hlundin): Will probably need a longer buffer for multi-channel. |
| int16_t decoded_buffer[kMaxFrameSize]; |
| length = decoder->DecodePlc(1, decoded_buffer); |
| if (length > 0) |
| algorithm_buffer_->PushBackInterleaved(decoded_buffer, length); |
| } else { |
| // Do simple zero-stuffing. |
| length = output_size_samples_; |
| algorithm_buffer_->Zeros(length); |
| // By not advancing the timestamp, NetEq inserts samples. |
| stats_.AddZeros(length); |
| } |
| if (increase_timestamp) { |
| sync_buffer_->IncreaseEndTimestamp(static_cast<uint32_t>(length)); |
| } |
| expand_->Reset(); |
| } |
| |
| int NetEqImpl::DtmfOverdub(const DtmfEvent& dtmf_event, size_t num_channels, |
| int16_t* output) const { |
| size_t out_index = 0; |
| size_t overdub_length = output_size_samples_; // Default value. |
| |
| if (sync_buffer_->dtmf_index() > sync_buffer_->next_index()) { |
| // Special operation for transition from "DTMF only" to "DTMF overdub". |
| out_index = std::min( |
| sync_buffer_->dtmf_index() - sync_buffer_->next_index(), |
| output_size_samples_); |
| overdub_length = output_size_samples_ - out_index; |
| } |
| |
| AudioMultiVector dtmf_output(num_channels); |
| int dtmf_return_value = 0; |
| if (!dtmf_tone_generator_->initialized()) { |
| dtmf_return_value = dtmf_tone_generator_->Init(fs_hz_, dtmf_event.event_no, |
| dtmf_event.volume); |
| } |
| if (dtmf_return_value == 0) { |
| dtmf_return_value = dtmf_tone_generator_->Generate(overdub_length, |
| &dtmf_output); |
| assert(overdub_length == dtmf_output.Size()); |
| } |
| dtmf_output.ReadInterleaved(overdub_length, &output[out_index]); |
| return dtmf_return_value < 0 ? dtmf_return_value : 0; |
| } |
| |
| int NetEqImpl::ExtractPackets(size_t required_samples, |
| PacketList* packet_list) { |
| bool first_packet = true; |
| uint8_t prev_payload_type = 0; |
| uint32_t prev_timestamp = 0; |
| uint16_t prev_sequence_number = 0; |
| bool next_packet_available = false; |
| |
| const RTPHeader* header = packet_buffer_->NextRtpHeader(); |
| assert(header); |
| if (!header) { |
| LOG(LS_ERROR) << "Packet buffer unexpectedly empty."; |
| return -1; |
| } |
| uint32_t first_timestamp = header->timestamp; |
| int extracted_samples = 0; |
| |
| // Packet extraction loop. |
| do { |
| timestamp_ = header->timestamp; |
| size_t discard_count = 0; |
| Packet* packet = packet_buffer_->GetNextPacket(&discard_count); |
| // |header| may be invalid after the |packet_buffer_| operation. |
| header = NULL; |
| if (!packet) { |
| LOG(LS_ERROR) << "Should always be able to extract a packet here"; |
| assert(false); // Should always be able to extract a packet here. |
| return -1; |
| } |
| stats_.PacketsDiscarded(discard_count); |
| stats_.StoreWaitingTime(packet->waiting_time->ElapsedMs()); |
| assert(!packet->payload.empty()); |
| packet_list->push_back(packet); // Store packet in list. |
| |
| if (first_packet) { |
| first_packet = false; |
| if (nack_enabled_) { |
| RTC_DCHECK(nack_); |
| // TODO(henrik.lundin): Should we update this for all decoded packets? |
| nack_->UpdateLastDecodedPacket(packet->header.sequenceNumber, |
| packet->header.timestamp); |
| } |
| prev_sequence_number = packet->header.sequenceNumber; |
| prev_timestamp = packet->header.timestamp; |
| prev_payload_type = packet->header.payloadType; |
| } |
| |
| // Store number of extracted samples. |
| int packet_duration = 0; |
| AudioDecoder* decoder = decoder_database_->GetDecoder( |
| packet->header.payloadType); |
| if (decoder) { |
| if (packet->sync_packet) { |
| packet_duration = rtc::checked_cast<int>(decoder_frame_length_); |
| } else { |
| if (packet->primary) { |
| packet_duration = decoder->PacketDuration(packet->payload.data(), |
| packet->payload.size()); |
| } else { |
| packet_duration = decoder->PacketDurationRedundant( |
| packet->payload.data(), packet->payload.size()); |
| stats_.SecondaryDecodedSamples(packet_duration); |
| } |
| } |
| } else if (!decoder_database_->IsComfortNoise(packet->header.payloadType)) { |
| LOG(LS_WARNING) << "Unknown payload type " |
| << static_cast<int>(packet->header.payloadType); |
| assert(false); |
| } |
| if (packet_duration <= 0) { |
| // Decoder did not return a packet duration. Assume that the packet |
| // contains the same number of samples as the previous one. |
| packet_duration = rtc::checked_cast<int>(decoder_frame_length_); |
| } |
| extracted_samples = packet->header.timestamp - first_timestamp + |
| packet_duration; |
| |
| // Check what packet is available next. |
| header = packet_buffer_->NextRtpHeader(); |
| next_packet_available = false; |
| if (header && prev_payload_type == header->payloadType) { |
| int16_t seq_no_diff = header->sequenceNumber - prev_sequence_number; |
| size_t ts_diff = header->timestamp - prev_timestamp; |
| if (seq_no_diff == 1 || |
| (seq_no_diff == 0 && ts_diff == decoder_frame_length_)) { |
| // The next sequence number is available, or the next part of a packet |
| // that was split into pieces upon insertion. |
| next_packet_available = true; |
| } |
| prev_sequence_number = header->sequenceNumber; |
| } |
| } while (extracted_samples < rtc::checked_cast<int>(required_samples) && |
| next_packet_available); |
| |
| if (extracted_samples > 0) { |
| // Delete old packets only when we are going to decode something. Otherwise, |
| // we could end up in the situation where we never decode anything, since |
| // all incoming packets are considered too old but the buffer will also |
| // never be flooded and flushed. |
| packet_buffer_->DiscardAllOldPackets(timestamp_); |
| } |
| |
| return extracted_samples; |
| } |
| |
| void NetEqImpl::UpdatePlcComponents(int fs_hz, size_t channels) { |
| // Delete objects and create new ones. |
| expand_.reset(expand_factory_->Create(background_noise_.get(), |
| sync_buffer_.get(), &random_vector_, |
| &stats_, fs_hz, channels)); |
| merge_.reset(new Merge(fs_hz, channels, expand_.get(), sync_buffer_.get())); |
| } |
| |
| void NetEqImpl::SetSampleRateAndChannels(int fs_hz, size_t channels) { |
| LOG(LS_VERBOSE) << "SetSampleRateAndChannels " << fs_hz << " " << channels; |
| // TODO(hlundin): Change to an enumerator and skip assert. |
| assert(fs_hz == 8000 || fs_hz == 16000 || fs_hz == 32000 || fs_hz == 48000); |
| assert(channels > 0); |
| |
| fs_hz_ = fs_hz; |
| fs_mult_ = fs_hz / 8000; |
| output_size_samples_ = static_cast<size_t>(kOutputSizeMs * 8 * fs_mult_); |
| decoder_frame_length_ = 3 * output_size_samples_; // Initialize to 30ms. |
| |
| last_mode_ = kModeNormal; |
| |
| // Create a new array of mute factors and set all to 1. |
| mute_factor_array_.reset(new int16_t[channels]); |
| for (size_t i = 0; i < channels; ++i) { |
| mute_factor_array_[i] = 16384; // 1.0 in Q14. |
| } |
| |
| ComfortNoiseDecoder* cng_decoder = decoder_database_->GetActiveCngDecoder(); |
| if (cng_decoder) |
| cng_decoder->Reset(); |
| |
| // Reinit post-decode VAD with new sample rate. |
| assert(vad_.get()); // Cannot be NULL here. |
| vad_->Init(); |
| |
| // Delete algorithm buffer and create a new one. |
| algorithm_buffer_.reset(new AudioMultiVector(channels)); |
| |
| // Delete sync buffer and create a new one. |
| sync_buffer_.reset(new SyncBuffer(channels, kSyncBufferSize * fs_mult_)); |
| |
| // Delete BackgroundNoise object and create a new one. |
| background_noise_.reset(new BackgroundNoise(channels)); |
| background_noise_->set_mode(background_noise_mode_); |
| |
| // Reset random vector. |
| random_vector_.Reset(); |
| |
| UpdatePlcComponents(fs_hz, channels); |
| |
| // Move index so that we create a small set of future samples (all 0). |
| sync_buffer_->set_next_index(sync_buffer_->next_index() - |
| expand_->overlap_length()); |
| |
| normal_.reset(new Normal(fs_hz, decoder_database_.get(), *background_noise_, |
| expand_.get())); |
| accelerate_.reset( |
| accelerate_factory_->Create(fs_hz, channels, *background_noise_)); |
| preemptive_expand_.reset(preemptive_expand_factory_->Create( |
| fs_hz, channels, *background_noise_, expand_->overlap_length())); |
| |
| // Delete ComfortNoise object and create a new one. |
| comfort_noise_.reset(new ComfortNoise(fs_hz, decoder_database_.get(), |
| sync_buffer_.get())); |
| |
| // Verify that |decoded_buffer_| is long enough. |
| if (decoded_buffer_length_ < kMaxFrameSize * channels) { |
| // Reallocate to larger size. |
| decoded_buffer_length_ = kMaxFrameSize * channels; |
| decoded_buffer_.reset(new int16_t[decoded_buffer_length_]); |
| } |
| |
| // Create DecisionLogic if it is not created yet, then communicate new sample |
| // rate and output size to DecisionLogic object. |
| if (!decision_logic_.get()) { |
| CreateDecisionLogic(); |
| } |
| decision_logic_->SetSampleRate(fs_hz_, output_size_samples_); |
| } |
| |
| NetEqImpl::OutputType NetEqImpl::LastOutputType() { |
| assert(vad_.get()); |
| assert(expand_.get()); |
| if (last_mode_ == kModeCodecInternalCng || last_mode_ == kModeRfc3389Cng) { |
| return OutputType::kCNG; |
| } else if (last_mode_ == kModeExpand && expand_->MuteFactor(0) == 0) { |
| // Expand mode has faded down to background noise only (very long expand). |
| return OutputType::kPLCCNG; |
| } else if (last_mode_ == kModeExpand) { |
| return OutputType::kPLC; |
| } else if (vad_->running() && !vad_->active_speech()) { |
| return OutputType::kVadPassive; |
| } else { |
| return OutputType::kNormalSpeech; |
| } |
| } |
| |
| void NetEqImpl::CreateDecisionLogic() { |
| decision_logic_.reset(DecisionLogic::Create( |
| fs_hz_, output_size_samples_, playout_mode_, decoder_database_.get(), |
| *packet_buffer_.get(), delay_manager_.get(), buffer_level_filter_.get(), |
| tick_timer_.get())); |
| } |
| } // namespace webrtc |