| /* |
| * 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 "modules/audio_coding/neteq/neteq_impl.h" |
| |
| #include <assert.h> |
| #include <algorithm> |
| #include <cstdint> |
| #include <cstring> |
| #include <list> |
| #include <map> |
| #include <utility> |
| #include <vector> |
| |
| #include "api/audio_codecs/audio_decoder.h" |
| #include "common_audio/signal_processing/include/signal_processing_library.h" |
| #include "modules/audio_coding/codecs/cng/webrtc_cng.h" |
| #include "modules/audio_coding/neteq/accelerate.h" |
| #include "modules/audio_coding/neteq/background_noise.h" |
| #include "modules/audio_coding/neteq/buffer_level_filter.h" |
| #include "modules/audio_coding/neteq/comfort_noise.h" |
| #include "modules/audio_coding/neteq/decision_logic.h" |
| #include "modules/audio_coding/neteq/decoder_database.h" |
| #include "modules/audio_coding/neteq/defines.h" |
| #include "modules/audio_coding/neteq/delay_manager.h" |
| #include "modules/audio_coding/neteq/delay_peak_detector.h" |
| #include "modules/audio_coding/neteq/dtmf_buffer.h" |
| #include "modules/audio_coding/neteq/dtmf_tone_generator.h" |
| #include "modules/audio_coding/neteq/expand.h" |
| #include "modules/audio_coding/neteq/merge.h" |
| #include "modules/audio_coding/neteq/nack_tracker.h" |
| #include "modules/audio_coding/neteq/normal.h" |
| #include "modules/audio_coding/neteq/packet.h" |
| #include "modules/audio_coding/neteq/packet_buffer.h" |
| #include "modules/audio_coding/neteq/post_decode_vad.h" |
| #include "modules/audio_coding/neteq/preemptive_expand.h" |
| #include "modules/audio_coding/neteq/red_payload_splitter.h" |
| #include "modules/audio_coding/neteq/statistics_calculator.h" |
| #include "modules/audio_coding/neteq/sync_buffer.h" |
| #include "modules/audio_coding/neteq/tick_timer.h" |
| #include "modules/audio_coding/neteq/time_stretch.h" |
| #include "modules/audio_coding/neteq/timestamp_scaler.h" |
| #include "rtc_base/checks.h" |
| #include "rtc_base/logging.h" |
| #include "rtc_base/numerics/safe_conversions.h" |
| #include "rtc_base/sanitizer.h" |
| #include "rtc_base/strings/audio_format_to_string.h" |
| #include "rtc_base/trace_event.h" |
| #include "system_wrappers/include/clock.h" |
| |
| namespace webrtc { |
| |
| NetEqImpl::Dependencies::Dependencies( |
| const NetEq::Config& config, |
| Clock* clock, |
| const rtc::scoped_refptr<AudioDecoderFactory>& decoder_factory) |
| : clock(clock), |
| tick_timer(new TickTimer), |
| stats(new StatisticsCalculator), |
| buffer_level_filter(new BufferLevelFilter), |
| decoder_database( |
| new DecoderDatabase(decoder_factory, config.codec_pair_id)), |
| delay_peak_detector( |
| new DelayPeakDetector(tick_timer.get(), config.enable_rtx_handling)), |
| delay_manager(DelayManager::Create(config.max_packets_in_buffer, |
| config.min_delay_ms, |
| config.enable_rtx_handling, |
| delay_peak_detector.get(), |
| tick_timer.get(), |
| stats.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())), |
| red_payload_splitter(new RedPayloadSplitter), |
| 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) |
| : clock_(deps.clock), |
| 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)), |
| red_payload_splitter_(std::move(deps.red_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)), |
| stats_(std::move(deps.stats)), |
| 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), |
| first_packet_(true), |
| enable_fast_accelerate_(config.enable_fast_accelerate), |
| nack_enabled_(false), |
| enable_muted_state_(config.enable_muted_state), |
| expand_uma_logger_("WebRTC.Audio.ExpandRatePercent", |
| 10, // Report once every 10 s. |
| tick_timer_.get()), |
| speech_expand_uma_logger_("WebRTC.Audio.SpeechExpandRatePercent", |
| 10, // Report once every 10 s. |
| tick_timer_.get()), |
| no_time_stretching_(config.for_test_no_time_stretching), |
| enable_rtx_handling_(config.enable_rtx_handling) { |
| RTC_LOG(LS_INFO) << "NetEq config: " << config.ToString(); |
| int fs = config.sample_rate_hz; |
| if (fs != 8000 && fs != 16000 && fs != 32000 && fs != 48000) { |
| RTC_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_; |
| 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 RTPHeader& rtp_header, |
| rtc::ArrayView<const uint8_t> payload, |
| uint32_t receive_timestamp) { |
| rtc::MsanCheckInitialized(payload); |
| TRACE_EVENT0("webrtc", "NetEqImpl::InsertPacket"); |
| rtc::CritScope lock(&crit_sect_); |
| if (InsertPacketInternal(rtp_header, payload, receive_timestamp) != 0) { |
| return kFail; |
| } |
| return kOK; |
| } |
| |
| void NetEqImpl::InsertEmptyPacket(const RTPHeader& /*rtp_header*/) { |
| // TODO(henrik.lundin) Handle NACK as well. This will make use of the |
| // rtp_header parameter. |
| // https://bugs.chromium.org/p/webrtc/issues/detail?id=7611 |
| rtc::CritScope lock(&crit_sect_); |
| delay_manager_->RegisterEmptyPacket(); |
| } |
| |
| 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, |
| absl::optional<Operations> action_override) { |
| TRACE_EVENT0("webrtc", "NetEqImpl::GetAudio"); |
| rtc::CritScope lock(&crit_sect_); |
| if (GetAudioInternal(audio_frame, muted, action_override) != 0) { |
| return kFail; |
| } |
| RTC_DCHECK_EQ( |
| audio_frame->sample_rate_hz_, |
| rtc::dchecked_cast<int>(audio_frame->samples_per_channel_ * 100)); |
| RTC_DCHECK_EQ(*muted, audio_frame->muted()); |
| 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; |
| } |
| |
| void NetEqImpl::SetCodecs(const std::map<int, SdpAudioFormat>& codecs) { |
| rtc::CritScope lock(&crit_sect_); |
| const std::vector<int> changed_payload_types = |
| decoder_database_->SetCodecs(codecs); |
| for (const int pt : changed_payload_types) { |
| packet_buffer_->DiscardPacketsWithPayloadType(pt, stats_.get()); |
| } |
| } |
| |
| bool NetEqImpl::RegisterPayloadType(int rtp_payload_type, |
| const SdpAudioFormat& audio_format) { |
| RTC_LOG(LS_VERBOSE) << "NetEqImpl::RegisterPayloadType: payload type " |
| << rtp_payload_type << ", codec " |
| << rtc::ToString(audio_format); |
| rtc::CritScope lock(&crit_sect_); |
| return decoder_database_->RegisterPayload(rtp_payload_type, audio_format) == |
| DecoderDatabase::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 || ret == DecoderDatabase::kDecoderNotFound) { |
| packet_buffer_->DiscardPacketsWithPayloadType(rtp_payload_type, |
| stats_.get()); |
| return kOK; |
| } |
| return kFail; |
| } |
| |
| void NetEqImpl::RemoveAllPayloadTypes() { |
| rtc::CritScope lock(&crit_sect_); |
| decoder_database_->RemoveAll(); |
| } |
| |
| 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; |
| } |
| |
| bool NetEqImpl::SetBaseMinimumDelayMs(int delay_ms) { |
| rtc::CritScope lock(&crit_sect_); |
| if (delay_ms >= 0 && delay_ms <= 10000) { |
| return delay_manager_->SetBaseMinimumDelay(delay_ms); |
| } |
| return false; |
| } |
| |
| int NetEqImpl::GetBaseMinimumDelayMs() const { |
| rtc::CritScope lock(&crit_sect_); |
| return delay_manager_->GetBaseMinimumDelay(); |
| } |
| |
| int NetEqImpl::TargetDelayMs() const { |
| rtc::CritScope lock(&crit_sect_); |
| RTC_DCHECK(delay_manager_.get()); |
| // The value from TargetLevel() is in number of packets, represented in Q8. |
| const size_t target_delay_samples = |
| (delay_manager_->TargetLevel() * decoder_frame_length_) >> 8; |
| return static_cast<int>(target_delay_samples) / |
| rtc::CheckedDivExact(fs_hz_, 1000); |
| } |
| |
| int NetEqImpl::FilteredCurrentDelayMs() const { |
| rtc::CritScope lock(&crit_sect_); |
| // Sum up the filtered packet buffer level with the future length of the sync |
| // buffer. |
| const int delay_samples = buffer_level_filter_->filtered_current_level() + |
| sync_buffer_->FutureLength(); |
| // The division below will truncate. The return value is in ms. |
| return delay_samples / rtc::CheckedDivExact(fs_hz_, 1000); |
| } |
| |
| 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_frame_length_) + |
| sync_buffer_->FutureLength(); |
| assert(delay_manager_.get()); |
| assert(decision_logic_.get()); |
| const int ms_per_packet = rtc::dchecked_cast<int>( |
| decision_logic_->packet_length_samples() / (fs_hz_ / 1000)); |
| stats_->PopulateDelayManagerStats(ms_per_packet, *delay_manager_.get(), |
| stats); |
| stats_->GetNetworkStatistics(fs_hz_, total_samples_in_buffers, |
| decoder_frame_length_, stats); |
| return 0; |
| } |
| |
| NetEqLifetimeStatistics NetEqImpl::GetLifetimeStatistics() const { |
| rtc::CritScope lock(&crit_sect_); |
| return stats_->GetLifetimeStatistics(); |
| } |
| |
| NetEqOperationsAndState NetEqImpl::GetOperationsAndState() const { |
| rtc::CritScope lock(&crit_sect_); |
| auto result = stats_->GetOperationsAndState(); |
| result.current_buffer_size_ms = |
| (packet_buffer_->NumSamplesInBuffer(decoder_frame_length_) + |
| sync_buffer_->FutureLength()) * |
| 1000 / fs_hz_; |
| result.current_frame_size_ms = decoder_frame_length_ * 1000 / fs_hz_; |
| result.next_packet_available = packet_buffer_->PeekNextPacket() && |
| packet_buffer_->PeekNextPacket()->timestamp == |
| sync_buffer_->end_timestamp(); |
| return result; |
| } |
| |
| 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(); |
| } |
| |
| absl::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 absl::nullopt; |
| } |
| return timestamp_scaler_->ToExternal(playout_timestamp_); |
| } |
| |
| int NetEqImpl::last_output_sample_rate_hz() const { |
| rtc::CritScope lock(&crit_sect_); |
| return last_output_sample_rate_hz_; |
| } |
| |
| absl::optional<SdpAudioFormat> NetEqImpl::GetDecoderFormat( |
| int payload_type) const { |
| rtc::CritScope lock(&crit_sect_); |
| const DecoderDatabase::DecoderInfo* const di = |
| decoder_database_->GetDecoderInfo(payload_type); |
| if (!di) { |
| return absl::nullopt; // Payload type not registered. |
| } |
| |
| SdpAudioFormat format = di->GetFormat(); |
| // TODO(solenberg): This is legacy but messed up - mixing RTP rate and SR. |
| format.clockrate_hz = di->IsRed() ? 8000 : di->SampleRateHz(); |
| const AudioDecoder* const decoder = di->GetDecoder(); |
| format.num_channels = decoder ? decoder->Channels() : 1; |
| return format; |
| } |
| |
| void NetEqImpl::FlushBuffers() { |
| rtc::CritScope lock(&crit_sect_); |
| RTC_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::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); |
| } |
| |
| std::vector<uint32_t> NetEqImpl::LastDecodedTimestamps() const { |
| rtc::CritScope lock(&crit_sect_); |
| return last_decoded_timestamps_; |
| } |
| |
| int NetEqImpl::SyncBufferSizeMs() const { |
| rtc::CritScope lock(&crit_sect_); |
| return rtc::dchecked_cast<int>(sync_buffer_->FutureLength() / |
| rtc::CheckedDivExact(fs_hz_, 1000)); |
| } |
| |
| 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 RTPHeader& rtp_header, |
| rtc::ArrayView<const uint8_t> payload, |
| uint32_t receive_timestamp) { |
| if (payload.empty()) { |
| RTC_LOG_F(LS_ERROR) << "payload is empty"; |
| return kInvalidPointer; |
| } |
| |
| int64_t receive_time_ms = clock_->TimeInMilliseconds(); |
| stats_->ReceivedPacket(); |
| |
| PacketList packet_list; |
| // Insert packet in a packet list. |
| packet_list.push_back([&rtp_header, &payload, &receive_time_ms] { |
| // Convert to Packet. |
| Packet packet; |
| packet.payload_type = rtp_header.payloadType; |
| packet.sequence_number = rtp_header.sequenceNumber; |
| packet.timestamp = rtp_header.timestamp; |
| packet.payload.SetData(payload.data(), payload.size()); |
| packet.packet_info = RtpPacketInfo(rtp_header, receive_time_ms); |
| // Waiting time will be set upon inserting the packet in the buffer. |
| RTC_DCHECK(!packet.waiting_time); |
| return packet; |
| }()); |
| |
| bool update_sample_rate_and_channels = first_packet_; |
| |
| if (update_sample_rate_and_channels) { |
| // Reset timestamp scaling. |
| timestamp_scaler_->Reset(); |
| } |
| |
| if (!decoder_database_->IsRed(rtp_header.payloadType)) { |
| // Scale timestamp to internal domain (only for some codecs). |
| timestamp_scaler_->ToInternal(&packet_list); |
| } |
| |
| // Store these for later use, since the first packet may very well disappear |
| // before we need these values. |
| uint32_t main_timestamp = packet_list.front().timestamp; |
| uint8_t main_payload_type = packet_list.front().payload_type; |
| uint16_t main_sequence_number = packet_list.front().sequence_number; |
| |
| // Reinitialize NetEq if it's needed (changed SSRC or first call). |
| if (update_sample_rate_and_channels) { |
| // Note: |first_packet_| will be cleared further down in this method, once |
| // the packet has been successfully inserted into the packet buffer. |
| |
| // Flush the packet buffer and DTMF buffer. |
| packet_buffer_->Flush(); |
| dtmf_buffer_->Flush(); |
| |
| // Update audio buffer timestamp. |
| sync_buffer_->IncreaseEndTimestamp(main_timestamp - timestamp_); |
| |
| // Update codecs. |
| timestamp_ = main_timestamp; |
| } |
| |
| if (nack_enabled_) { |
| RTC_DCHECK(nack_); |
| if (update_sample_rate_and_channels) { |
| nack_->Reset(); |
| } |
| nack_->UpdateLastReceivedPacket(rtp_header.sequenceNumber, |
| rtp_header.timestamp); |
| } |
| |
| // Check for RED payload type, and separate payloads into several packets. |
| if (decoder_database_->IsRed(rtp_header.payloadType)) { |
| if (!red_payload_splitter_->SplitRed(&packet_list)) { |
| return kRedundancySplitError; |
| } |
| // Only accept a few RED payloads of the same type as the main data, |
| // DTMF events and CNG. |
| red_payload_splitter_->CheckRedPayloads(&packet_list, *decoder_database_); |
| if (packet_list.empty()) { |
| return kRedundancySplitError; |
| } |
| } |
| |
| // Check payload types. |
| if (decoder_database_->CheckPayloadTypes(packet_list) == |
| DecoderDatabase::kDecoderNotFound) { |
| return kUnknownRtpPayloadType; |
| } |
| |
| RTC_DCHECK(!packet_list.empty()); |
| |
| // Update main_timestamp, if new packets appear in the list |
| // after RED splitting. |
| if (decoder_database_->IsRed(rtp_header.payloadType)) { |
| timestamp_scaler_->ToInternal(&packet_list); |
| main_timestamp = packet_list.front().timestamp; |
| main_payload_type = packet_list.front().payload_type; |
| main_sequence_number = packet_list.front().sequence_number; |
| } |
| |
| // 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()) { |
| const Packet& current_packet = (*it); |
| RTC_DCHECK(!current_packet.payload.empty()); |
| if (decoder_database_->IsDtmf(current_packet.payload_type)) { |
| DtmfEvent event; |
| int ret = DtmfBuffer::ParseEvent(current_packet.timestamp, |
| current_packet.payload.data(), |
| current_packet.payload.size(), &event); |
| if (ret != DtmfBuffer::kOK) { |
| return kDtmfParsingError; |
| } |
| if (dtmf_buffer_->InsertEvent(event) != DtmfBuffer::kOK) { |
| return kDtmfInsertError; |
| } |
| it = packet_list.erase(it); |
| } else { |
| ++it; |
| } |
| } |
| |
| // Update bandwidth estimate, if the packet is not comfort noise. |
| if (!packet_list.empty() && |
| !decoder_database_->IsComfortNoise(main_payload_type)) { |
| // The list can be empty here if we got nothing but DTMF payloads. |
| AudioDecoder* decoder = decoder_database_->GetDecoder(main_payload_type); |
| RTC_DCHECK(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().sequence_number, |
| packet_list.front().timestamp, receive_timestamp); |
| } |
| |
| PacketList parsed_packet_list; |
| while (!packet_list.empty()) { |
| Packet& packet = packet_list.front(); |
| const DecoderDatabase::DecoderInfo* info = |
| decoder_database_->GetDecoderInfo(packet.payload_type); |
| if (!info) { |
| RTC_LOG(LS_WARNING) << "SplitAudio unknown payload type"; |
| return kUnknownRtpPayloadType; |
| } |
| |
| if (info->IsComfortNoise()) { |
| // Carry comfort noise packets along. |
| parsed_packet_list.splice(parsed_packet_list.end(), packet_list, |
| packet_list.begin()); |
| } else { |
| const auto sequence_number = packet.sequence_number; |
| const auto payload_type = packet.payload_type; |
| const Packet::Priority original_priority = packet.priority; |
| const auto& packet_info = packet.packet_info; |
| auto packet_from_result = [&](AudioDecoder::ParseResult& result) { |
| Packet new_packet; |
| new_packet.sequence_number = sequence_number; |
| new_packet.payload_type = payload_type; |
| new_packet.timestamp = result.timestamp; |
| new_packet.priority.codec_level = result.priority; |
| new_packet.priority.red_level = original_priority.red_level; |
| new_packet.packet_info = packet_info; |
| new_packet.frame = std::move(result.frame); |
| return new_packet; |
| }; |
| |
| std::vector<AudioDecoder::ParseResult> results = |
| info->GetDecoder()->ParsePayload(std::move(packet.payload), |
| packet.timestamp); |
| if (results.empty()) { |
| packet_list.pop_front(); |
| } else { |
| bool first = true; |
| for (auto& result : results) { |
| RTC_DCHECK(result.frame); |
| RTC_DCHECK_GE(result.priority, 0); |
| if (first) { |
| // Re-use the node and move it to parsed_packet_list. |
| packet_list.front() = packet_from_result(result); |
| parsed_packet_list.splice(parsed_packet_list.end(), packet_list, |
| packet_list.begin()); |
| first = false; |
| } else { |
| parsed_packet_list.push_back(packet_from_result(result)); |
| } |
| } |
| } |
| } |
| } |
| |
| // Calculate the number of primary (non-FEC/RED) packets. |
| const size_t number_of_primary_packets = std::count_if( |
| parsed_packet_list.begin(), parsed_packet_list.end(), |
| [](const Packet& in) { return in.priority.codec_level == 0; }); |
| if (number_of_primary_packets < parsed_packet_list.size()) { |
| stats_->SecondaryPacketsReceived(parsed_packet_list.size() - |
| number_of_primary_packets); |
| } |
| |
| // Insert packets in buffer. |
| const int ret = packet_buffer_->InsertPacketList( |
| &parsed_packet_list, *decoder_database_, ¤t_rtp_payload_type_, |
| ¤t_cng_rtp_payload_type_, stats_.get()); |
| 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) { |
| 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 Packet* next_packet = packet_buffer_->PeekNextPacket(); |
| RTC_DCHECK(next_packet); |
| const int payload_type = next_packet->payload_type; |
| 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_payload_type); |
| assert(dec_info); // Already checked that the payload type is known. |
| delay_manager_->LastDecodedWasCngOrDtmf(dec_info->IsComfortNoise() || |
| dec_info->IsDtmf()); |
| if (delay_manager_->last_pack_cng_or_dtmf() == 0) { |
| // Calculate the total speech length carried in each packet. |
| if (number_of_primary_packets > 0) { |
| const size_t packet_length_samples = |
| number_of_primary_packets * 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::dchecked_cast<int>((1000 * packet_length_samples) / fs_hz_)); |
| } |
| } |
| |
| // Update statistics. |
| if ((enable_rtx_handling_ || (int32_t)(main_timestamp - timestamp_) >= 0) && |
| !new_codec_) { |
| // Only update statistics if incoming packet is not older than last played |
| // out packet or RTX handling is enabled, and if new codec flag is not |
| // set. |
| delay_manager_->Update(main_sequence_number, main_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, |
| absl::optional<Operations> action_override) { |
| PacketList packet_list; |
| DtmfEvent dtmf_event; |
| Operations operation; |
| bool play_dtmf; |
| *muted = false; |
| last_decoded_timestamps_.clear(); |
| tick_timer_->Increment(); |
| stats_->IncreaseCounter(output_size_samples_, fs_hz_); |
| const auto lifetime_stats = stats_->GetLifetimeStatistics(); |
| expand_uma_logger_.UpdateSampleCounter(lifetime_stats.concealed_samples, |
| fs_hz_); |
| speech_expand_uma_logger_.UpdateSampleCounter( |
| lifetime_stats.concealed_samples - |
| lifetime_stats.silent_concealed_samples, |
| fs_hz_); |
| |
| // Check for muted state. |
| if (enable_muted_state_ && expand_->Muted() && packet_buffer_->Empty()) { |
| RTC_DCHECK_EQ(last_mode_, kModeExpand); |
| audio_frame->Reset(); |
| RTC_DCHECK(audio_frame->muted()); // Reset() should mute the frame. |
| 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_, false); |
| *muted = true; |
| return 0; |
| } |
| int return_value = GetDecision(&operation, &packet_list, &dtmf_event, |
| &play_dtmf, action_override); |
| if (return_value != 0) { |
| last_mode_ = kModeError; |
| return return_value; |
| } |
| |
| AudioDecoder::SpeechType speech_type; |
| int length = 0; |
| const size_t start_num_packets = packet_list.size(); |
| 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_); |
| |
| // This is the criterion that we did decode some data through the speech |
| // decoder, and the operation resulted in comfort noise. |
| const bool codec_internal_sid_frame = |
| (speech_type == AudioDecoder::kComfortNoise && |
| start_num_packets > packet_list.size()); |
| |
| if (sid_frame_available || codec_internal_sid_frame) { |
| // 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); |
| if (length > 0) { |
| stats_->DecodedOutputPlayed(); |
| } |
| break; |
| } |
| case kMerge: { |
| DoMerge(decoded_buffer_.get(), length, speech_type, play_dtmf); |
| break; |
| } |
| case kExpand: { |
| RTC_DCHECK_EQ(return_value, 0); |
| if (!current_rtp_payload_type_ || !DoCodecPlc()) { |
| return_value = DoExpand(play_dtmf); |
| } |
| RTC_DCHECK_GE(sync_buffer_->FutureLength() - expand_->overlap_length(), |
| output_size_samples_); |
| 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 kUndefined: { |
| RTC_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(); |
| } |
| |
| // We treat it as if all packets referenced to by |last_decoded_packet_infos_| |
| // were mashed together when creating the samples in |algorithm_buffer_|. |
| RtpPacketInfos packet_infos(std::move(last_decoded_packet_infos_)); |
| last_decoded_packet_infos_.clear(); |
| |
| // Copy samples from |algorithm_buffer_| to |sync_buffer_|. |
| // |
| // TODO(bugs.webrtc.org/10757): |
| // We would in the future also like to pass |packet_infos| so that we can do |
| // sample-perfect tracking of that information across |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) { |
| RTC_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_; |
| // TODO(bugs.webrtc.org/10757): |
| // We don't have the ability to properly track individual packets once their |
| // audio samples have entered |sync_buffer_|. So for now, treat it as if |
| // |packet_infos| from packets decoded by the current |GetAudioInternal()| |
| // call were all consumed assembling the current audio frame and the current |
| // audio frame only. |
| audio_frame->packet_infos_ = std::move(packet_infos); |
| 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_) { |
| RTC_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 |
| audio_frame->Mute(); |
| return kSampleUnderrun; |
| } |
| |
| // Should always have overlap samples left in the |sync_buffer_|. |
| RTC_DCHECK_GE(sync_buffer_->FutureLength(), expand_->overlap_length()); |
| |
| // TODO(yujo): For muted frames, this can be a copy rather than an addition. |
| if (play_dtmf) { |
| return_value = DtmfOverdub(dtmf_event, sync_buffer_->Channels(), |
| audio_frame->mutable_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 && last_mode_ != kModeCodecPlc) { |
| // 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 || last_mode_ == kModeCodecPlc)) { |
| 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, |
| absl::optional<Operations> action_override) { |
| // 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, |
| stats_.get()); |
| } |
| const Packet* packet = packet_buffer_->PeekNextPacket(); |
| |
| 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 (packet && decoder_database_->IsComfortNoise(packet->payload_type) && |
| (end_timestamp >= packet->timestamp || |
| end_timestamp + generated_noise_samples > packet->timestamp)) { |
| // Don't use this packet, discard it. |
| if (packet_buffer_->DiscardNextPacket(stats_.get()) != |
| PacketBuffer::kOK) { |
| assert(false); // Must be ok by design. |
| } |
| // Check buffer again. |
| if (!new_codec_) { |
| packet_buffer_->DiscardOldPackets(end_timestamp, 5 * fs_hz_, |
| stats_.get()); |
| } |
| packet = packet_buffer_->PeekNextPacket(); |
| } |
| } |
| |
| 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::dchecked_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_, packet, last_mode_, |
| *play_dtmf, generated_noise_samples, &reset_decoder_); |
| |
| // Disallow time stretching if this packet is DTX, because such a decision may |
| // be based on earlier buffer level estimate, as we do not update buffer level |
| // during DTX. When we have a better way to update buffer level during DTX, |
| // this can be discarded. |
| if (packet && packet->frame && packet->frame->IsDtxPacket() && |
| (*operation == kMerge || *operation == kAccelerate || |
| *operation == kFastAccelerate || *operation == kPreemptiveExpand)) { |
| *operation = kNormal; |
| } |
| |
| if (action_override) { |
| // Use the provided action instead of the decision NetEq decided on. |
| *operation = *action_override; |
| } |
| // 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::dchecked_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 && !packet) { |
| timestamp_ = dtmf_event->timestamp; |
| } else { |
| if (!packet) { |
| RTC_LOG(LS_ERROR) << "Packet missing where it shouldn't."; |
| return -1; |
| } |
| timestamp_ = packet->timestamp; |
| if (*operation == kRfc3389CngNoPacket && |
| decoder_database_->IsComfortNoise(packet->payload_type)) { |
| // 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 (packet) { |
| sync_buffer_->IncreaseEndTimestamp(packet->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(packet->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.payload_type; |
| if (!decoder_database_->IsComfortNoise(payload_type)) { |
| decoder = decoder_database_->GetDecoder(payload_type); |
| assert(decoder); |
| if (!decoder) { |
| RTC_LOG(LS_WARNING) |
| << "Unknown payload type " << static_cast<int>(payload_type); |
| packet_list->clear(); |
| 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) { |
| RTC_LOG(LS_WARNING) |
| << "Unknown payload type " << static_cast<int>(payload_type); |
| packet_list->clear(); |
| 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. |
| return_value = kDecoderErrorCode; |
| RTC_LOG(LS_WARNING) << "Decoder returned error code: " << error_code; |
| } else { |
| // Decoder does not implement error codes. Return generic error. |
| return_value = kOtherDecoderError; |
| RTC_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::dchecked_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. |
| RTC_LOG(LS_WARNING) << "Failed to decode CNG"; |
| *decoded_length = -1; |
| break; |
| } |
| if (*decoded_length > static_cast<int>(decoded_buffer_length_)) { |
| // Guard against overflow. |
| RTC_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) { |
| RTC_DCHECK(last_decoded_timestamps_.empty()); |
| RTC_DCHECK(last_decoded_packet_infos_.empty()); |
| |
| // Do decoding. |
| while (!packet_list->empty() && !decoder_database_->IsComfortNoise( |
| packet_list->front().payload_type)) { |
| 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); |
| |
| auto opt_result = packet_list->front().frame->Decode( |
| rtc::ArrayView<int16_t>(&decoded_buffer_[*decoded_length], |
| decoded_buffer_length_ - *decoded_length)); |
| last_decoded_timestamps_.push_back(packet_list->front().timestamp); |
| last_decoded_packet_infos_.push_back( |
| std::move(packet_list->front().packet_info)); |
| packet_list->pop_front(); |
| if (opt_result) { |
| const auto& result = *opt_result; |
| *speech_type = result.speech_type; |
| if (result.num_decoded_samples > 0) { |
| *decoded_length += rtc::dchecked_cast<int>(result.num_decoded_samples); |
| // Update |decoder_frame_length_| with number of samples per channel. |
| decoder_frame_length_ = |
| result.num_decoded_samples / decoder->Channels(); |
| } |
| } else { |
| // Error. |
| // TODO(ossu): What to put here? |
| RTC_LOG(LS_WARNING) << "Decode error"; |
| *decoded_length = -1; |
| last_decoded_packet_infos_.clear(); |
| packet_list->clear(); |
| break; |
| } |
| if (*decoded_length > rtc::dchecked_cast<int>(decoded_buffer_length_)) { |
| // Guard against overflow. |
| RTC_LOG(LS_WARNING) << "Decoded too much."; |
| packet_list->clear(); |
| return kDecodedTooMuch; |
| } |
| } // 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 && decoder_database_->IsComfortNoise( |
| packet_list->front().payload_type))); |
| return 0; |
| } |
| |
| void NetEqImpl::DoNormal(const int16_t* decoded_buffer, |
| size_t decoded_length, |
| AudioDecoder::SpeechType speech_type, |
| bool play_dtmf) { |
| assert(normal_.get()); |
| normal_->Process(decoded_buffer, decoded_length, last_mode_, |
| 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(merge_.get()); |
| size_t new_length = |
| merge_->Process(decoded_buffer, decoded_length, algorithm_buffer_.get()); |
| // Correction can be negative. |
| int expand_length_correction = |
| rtc::dchecked_cast<int>(new_length) - |
| rtc::dchecked_cast<int>(decoded_length / algorithm_buffer_->Channels()); |
| |
| // Update in-call and post-call statistics. |
| if (expand_->MuteFactor(0) == 0) { |
| // Expand generates only noise. |
| stats_->ExpandedNoiseSamplesCorrection(expand_length_correction); |
| } else { |
| // Expansion generates more than only noise. |
| stats_->ExpandedVoiceSamplesCorrection(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(); |
| } |
| } |
| |
| bool NetEqImpl::DoCodecPlc() { |
| AudioDecoder* decoder = decoder_database_->GetActiveDecoder(); |
| if (!decoder) { |
| return false; |
| } |
| const size_t channels = algorithm_buffer_->Channels(); |
| const size_t requested_samples_per_channel = |
| output_size_samples_ - |
| (sync_buffer_->FutureLength() - expand_->overlap_length()); |
| concealment_audio_.Clear(); |
| decoder->GeneratePlc(requested_samples_per_channel, &concealment_audio_); |
| if (concealment_audio_.empty()) { |
| // Nothing produced. Resort to regular expand. |
| return false; |
| } |
| RTC_CHECK_GE(concealment_audio_.size(), |
| requested_samples_per_channel * channels); |
| sync_buffer_->PushBackInterleaved(concealment_audio_); |
| RTC_DCHECK_NE(algorithm_buffer_->Channels(), 0); |
| const size_t concealed_samples_per_channel = |
| concealment_audio_.size() / channels; |
| |
| // Update in-call and post-call statistics. |
| const bool is_new_concealment_event = (last_mode_ != kModeCodecPlc); |
| if (std::all_of(concealment_audio_.cbegin(), concealment_audio_.cend(), |
| [](int16_t i) { return i == 0; })) { |
| // Expand operation generates only noise. |
| stats_->ExpandedNoiseSamples(concealed_samples_per_channel, |
| is_new_concealment_event); |
| } else { |
| // Expand operation generates more than only noise. |
| stats_->ExpandedVoiceSamples(concealed_samples_per_channel, |
| is_new_concealment_event); |
| } |
| last_mode_ = kModeCodecPlc; |
| 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 true; |
| } |
| |
| 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(); |
| bool is_new_concealment_event = (last_mode_ != kModeExpand); |
| |
| // Update in-call and post-call statistics. |
| if (expand_->MuteFactor(0) == 0) { |
| // Expand operation generates only noise. |
| stats_->ExpandedNoiseSamples(length, is_new_concealment_event); |
| } else { |
| // Expand operation generates more than only noise. |
| stats_->ExpandedVoiceSamples(length, is_new_concealment_event); |
| } |
| |
| 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); |
| const Packet& packet = packet_list->front(); |
| if (!decoder_database_->IsComfortNoise(packet.payload_type)) { |
| RTC_LOG(LS_ERROR) << "Trying to decode non-CNG payload as CNG."; |
| return kOtherError; |
| } |
| 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) { |
| RTC_LOG(LS_WARNING) << "Comfort noise generator returned 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()); |
| normal_->Process(decoded_buffer, decoded_length, last_mode_, |
| 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) { |
| // RTC_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; |
| } |
| |
| 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 Packet* next_packet = packet_buffer_->PeekNextPacket(); |
| RTC_DCHECK(next_packet); |
| if (!next_packet) { |
| RTC_LOG(LS_ERROR) << "Packet buffer unexpectedly empty."; |
| return -1; |
| } |
| uint32_t first_timestamp = next_packet->timestamp; |
| size_t extracted_samples = 0; |
| |
| // Packet extraction loop. |
| do { |
| timestamp_ = next_packet->timestamp; |
| absl::optional<Packet> packet = packet_buffer_->GetNextPacket(); |
| // |next_packet| may be invalid after the |packet_buffer_| operation. |
| next_packet = nullptr; |
| if (!packet) { |
| RTC_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; |
| } |
| const uint64_t waiting_time_ms = packet->waiting_time->ElapsedMs(); |
| stats_->StoreWaitingTime(waiting_time_ms); |
| RTC_DCHECK(!packet->empty()); |
| |
| 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->sequence_number, |
| packet->timestamp); |
| } |
| prev_sequence_number = packet->sequence_number; |
| prev_timestamp = packet->timestamp; |
| prev_payload_type = packet->payload_type; |
| } |
| |
| const bool has_cng_packet = |
| decoder_database_->IsComfortNoise(packet->payload_type); |
| // Store number of extracted samples. |
| size_t packet_duration = 0; |
| if (packet->frame) { |
| packet_duration = packet->frame->Duration(); |
| // TODO(ossu): Is this the correct way to track Opus FEC packets? |
| if (packet->priority.codec_level > 0) { |
| stats_->SecondaryDecodedSamples( |
| rtc::dchecked_cast<int>(packet_duration)); |
| } |
| } else if (!has_cng_packet) { |
| RTC_LOG(LS_WARNING) << "Unknown payload type " |
| << static_cast<int>(packet->payload_type); |
| RTC_NOTREACHED(); |
| } |
| |
| 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 = decoder_frame_length_; |
| } |
| extracted_samples = packet->timestamp - first_timestamp + packet_duration; |
| |
| stats_->JitterBufferDelay(packet_duration, waiting_time_ms); |
| |
| packet_list->push_back(std::move(*packet)); // Store packet in list. |
| packet = absl::nullopt; // Ensure it's never used after the move. |
| |
| // Check what packet is available next. |
| next_packet = packet_buffer_->PeekNextPacket(); |
| next_packet_available = false; |
| if (next_packet && prev_payload_type == next_packet->payload_type && |
| !has_cng_packet) { |
| int16_t seq_no_diff = next_packet->sequence_number - prev_sequence_number; |
| size_t ts_diff = next_packet->timestamp - prev_timestamp; |
| if ((seq_no_diff == 1 || seq_no_diff == 0) && |
| ts_diff <= packet_duration) { |
| // 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 = next_packet->sequence_number; |
| prev_timestamp = next_packet->timestamp; |
| } |
| } while (extracted_samples < 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_, stats_.get()); |
| } |
| |
| return rtc::dchecked_cast<int>(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_.get(), fs_hz, channels)); |
| merge_.reset(new Merge(fs_hz, channels, expand_.get(), sync_buffer_.get())); |
| } |
| |
| void NetEqImpl::SetSampleRateAndChannels(int fs_hz, size_t channels) { |
| RTC_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; |
| |
| 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)); |
| |
| // 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_, no_time_stretching_, |
| decoder_database_.get(), *packet_buffer_.get(), delay_manager_.get(), |
| buffer_level_filter_.get(), tick_timer_.get())); |
| } |
| } // namespace webrtc |