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webrtc / src / 6e5433c4d4f37133ca56bc9c16e07f39d8e29897 / . / modules / audio_coding / neteq / neteq_impl.cc
blob: 37036e34c49247c0fc23845af374ca5a65969fd5 [file] [log] [blame]
/*
* 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) {
rtc::MsanCheckInitialized(payload);
TRACE_EVENT0("webrtc", "NetEqImpl::InsertPacket");
rtc::CritScope lock(&crit_sect_);
if (InsertPacketInternal(rtp_header, payload) != 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;
}
case NetEqImpl::OutputType::kCodecPLC: {
audio_frame->speech_type_ = AudioFrame::kCodecPLC;
audio_frame->vad_activity_ = last_vad_activity;
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<NetEq::DecoderFormat> NetEqImpl::GetDecoderFormat(
int payload_type) const {
rtc::CritScope lock(&crit_sect_);
const DecoderDatabase::DecoderInfo* const di =
decoder_database_->GetDecoderInfo(payload_type);
if (di) {
const AudioDecoder* const decoder = di->GetDecoder();
// TODO(kwiberg): Why the special case for RED?
return DecoderFormat{
/*sample_rate_hz=*/di->IsRed() ? 8000 : di->SampleRateHz(),
/*num_channels=*/
decoder ? rtc::dchecked_cast<int>(decoder->Channels()) : 1,
/*sdp_format=*/di->GetFormat()};
} else {
// Payload type not registered.
return absl::nullopt;
}
}
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) {
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;
}
}
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_, &current_rtp_payload_type_,
&current_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();
last_decoded_packet_infos_.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(last_decoded_packet_infos_);
// 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 if (last_mode_ == kModeCodecPlc) {
return OutputType::kCodecPLC;
} 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