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webrtc / src / 48ed930975ef7e84023044ed584c4eff914e6c9a / . / webrtc / modules / audio_coding / neteq / neteq_impl.cc
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/*
* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#include "webrtc/modules/audio_coding/neteq/neteq_impl.h"
#include <assert.h>
#include <memory.h> // memset
#include <algorithm>
#include "webrtc/base/checks.h"
#include "webrtc/base/logging.h"
#include "webrtc/base/safe_conversions.h"
#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
#include "webrtc/modules/audio_coding/codecs/audio_decoder.h"
#include "webrtc/modules/audio_coding/neteq/accelerate.h"
#include "webrtc/modules/audio_coding/neteq/background_noise.h"
#include "webrtc/modules/audio_coding/neteq/buffer_level_filter.h"
#include "webrtc/modules/audio_coding/neteq/comfort_noise.h"
#include "webrtc/modules/audio_coding/neteq/decision_logic.h"
#include "webrtc/modules/audio_coding/neteq/decoder_database.h"
#include "webrtc/modules/audio_coding/neteq/defines.h"
#include "webrtc/modules/audio_coding/neteq/delay_manager.h"
#include "webrtc/modules/audio_coding/neteq/delay_peak_detector.h"
#include "webrtc/modules/audio_coding/neteq/dtmf_buffer.h"
#include "webrtc/modules/audio_coding/neteq/dtmf_tone_generator.h"
#include "webrtc/modules/audio_coding/neteq/expand.h"
#include "webrtc/modules/audio_coding/neteq/merge.h"
#include "webrtc/modules/audio_coding/neteq/nack.h"
#include "webrtc/modules/audio_coding/neteq/normal.h"
#include "webrtc/modules/audio_coding/neteq/packet_buffer.h"
#include "webrtc/modules/audio_coding/neteq/packet.h"
#include "webrtc/modules/audio_coding/neteq/payload_splitter.h"
#include "webrtc/modules/audio_coding/neteq/post_decode_vad.h"
#include "webrtc/modules/audio_coding/neteq/preemptive_expand.h"
#include "webrtc/modules/audio_coding/neteq/sync_buffer.h"
#include "webrtc/modules/audio_coding/neteq/timestamp_scaler.h"
#include "webrtc/modules/interface/module_common_types.h"
#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
// Modify the code to obtain backwards bit-exactness. Once bit-exactness is no
// longer required, this #define should be removed (and the code that it
// enables).
#define LEGACY_BITEXACT
namespace webrtc {
NetEqImpl::NetEqImpl(const NetEq::Config& config,
BufferLevelFilter* buffer_level_filter,
DecoderDatabase* decoder_database,
DelayManager* delay_manager,
DelayPeakDetector* delay_peak_detector,
DtmfBuffer* dtmf_buffer,
DtmfToneGenerator* dtmf_tone_generator,
PacketBuffer* packet_buffer,
PayloadSplitter* payload_splitter,
TimestampScaler* timestamp_scaler,
AccelerateFactory* accelerate_factory,
ExpandFactory* expand_factory,
PreemptiveExpandFactory* preemptive_expand_factory,
bool create_components)
: crit_sect_(CriticalSectionWrapper::CreateCriticalSection()),
buffer_level_filter_(buffer_level_filter),
decoder_database_(decoder_database),
delay_manager_(delay_manager),
delay_peak_detector_(delay_peak_detector),
dtmf_buffer_(dtmf_buffer),
dtmf_tone_generator_(dtmf_tone_generator),
packet_buffer_(packet_buffer),
payload_splitter_(payload_splitter),
timestamp_scaler_(timestamp_scaler),
vad_(new PostDecodeVad()),
expand_factory_(expand_factory),
accelerate_factory_(accelerate_factory),
preemptive_expand_factory_(preemptive_expand_factory),
last_mode_(kModeNormal),
decoded_buffer_length_(kMaxFrameSize),
decoded_buffer_(new int16_t[decoded_buffer_length_]),
playout_timestamp_(0),
new_codec_(false),
timestamp_(0),
reset_decoder_(false),
current_rtp_payload_type_(0xFF), // Invalid RTP payload type.
current_cng_rtp_payload_type_(0xFF), // Invalid RTP payload type.
ssrc_(0),
first_packet_(true),
error_code_(0),
decoder_error_code_(0),
background_noise_mode_(config.background_noise_mode),
playout_mode_(config.playout_mode),
enable_fast_accelerate_(config.enable_fast_accelerate),
nack_enabled_(false) {
LOG(LS_INFO) << "NetEq config: " << config.ToString();
int fs = config.sample_rate_hz;
if (fs != 8000 && fs != 16000 && fs != 32000 && fs != 48000) {
LOG(LS_ERROR) << "Sample rate " << fs << " Hz not supported. " <<
"Changing to 8000 Hz.";
fs = 8000;
}
fs_hz_ = fs;
fs_mult_ = fs / 8000;
output_size_samples_ = static_cast<size_t>(kOutputSizeMs * 8 * fs_mult_);
decoder_frame_length_ = 3 * output_size_samples_;
WebRtcSpl_Init();
if (create_components) {
SetSampleRateAndChannels(fs, 1); // Default is 1 channel.
}
}
NetEqImpl::~NetEqImpl() = default;
int NetEqImpl::InsertPacket(const WebRtcRTPHeader& rtp_header,
const uint8_t* payload,
size_t length_bytes,
uint32_t receive_timestamp) {
CriticalSectionScoped lock(crit_sect_.get());
LOG(LS_VERBOSE) << "InsertPacket: ts=" << rtp_header.header.timestamp <<
", sn=" << rtp_header.header.sequenceNumber <<
", pt=" << static_cast<int>(rtp_header.header.payloadType) <<
", ssrc=" << rtp_header.header.ssrc <<
", len=" << length_bytes;
int error = InsertPacketInternal(rtp_header, payload, length_bytes,
receive_timestamp, false);
if (error != 0) {
error_code_ = error;
return kFail;
}
return kOK;
}
int NetEqImpl::InsertSyncPacket(const WebRtcRTPHeader& rtp_header,
uint32_t receive_timestamp) {
CriticalSectionScoped lock(crit_sect_.get());
LOG(LS_VERBOSE) << "InsertPacket-Sync: ts="
<< rtp_header.header.timestamp <<
", sn=" << rtp_header.header.sequenceNumber <<
", pt=" << static_cast<int>(rtp_header.header.payloadType) <<
", ssrc=" << rtp_header.header.ssrc;
const uint8_t kSyncPayload[] = { 's', 'y', 'n', 'c' };
int error = InsertPacketInternal(
rtp_header, kSyncPayload, sizeof(kSyncPayload), receive_timestamp, true);
if (error != 0) {
error_code_ = error;
return kFail;
}
return kOK;
}
int NetEqImpl::GetAudio(size_t max_length, int16_t* output_audio,
size_t* samples_per_channel, int* num_channels,
NetEqOutputType* type) {
CriticalSectionScoped lock(crit_sect_.get());
LOG(LS_VERBOSE) << "GetAudio";
int error = GetAudioInternal(max_length, output_audio, samples_per_channel,
num_channels);
LOG(LS_VERBOSE) << "Produced " << *samples_per_channel <<
" samples/channel for " << *num_channels << " channel(s)";
if (error != 0) {
error_code_ = error;
return kFail;
}
if (type) {
*type = LastOutputType();
}
return kOK;
}
int NetEqImpl::RegisterPayloadType(enum NetEqDecoder codec,
uint8_t rtp_payload_type) {
CriticalSectionScoped lock(crit_sect_.get());
LOG(LS_VERBOSE) << "RegisterPayloadType "
<< static_cast<int>(rtp_payload_type) << " " << codec;
int ret = decoder_database_->RegisterPayload(rtp_payload_type, codec);
if (ret != DecoderDatabase::kOK) {
switch (ret) {
case DecoderDatabase::kInvalidRtpPayloadType:
error_code_ = kInvalidRtpPayloadType;
break;
case DecoderDatabase::kCodecNotSupported:
error_code_ = kCodecNotSupported;
break;
case DecoderDatabase::kDecoderExists:
error_code_ = kDecoderExists;
break;
default:
error_code_ = kOtherError;
}
return kFail;
}
return kOK;
}
int NetEqImpl::RegisterExternalDecoder(AudioDecoder* decoder,
enum NetEqDecoder codec,
uint8_t rtp_payload_type,
int sample_rate_hz) {
CriticalSectionScoped lock(crit_sect_.get());
LOG(LS_VERBOSE) << "RegisterExternalDecoder "
<< static_cast<int>(rtp_payload_type) << " " << codec;
if (!decoder) {
LOG(LS_ERROR) << "Cannot register external decoder with NULL pointer";
assert(false);
return kFail;
}
int ret = decoder_database_->InsertExternal(rtp_payload_type, codec,
sample_rate_hz, decoder);
if (ret != DecoderDatabase::kOK) {
switch (ret) {
case DecoderDatabase::kInvalidRtpPayloadType:
error_code_ = kInvalidRtpPayloadType;
break;
case DecoderDatabase::kCodecNotSupported:
error_code_ = kCodecNotSupported;
break;
case DecoderDatabase::kDecoderExists:
error_code_ = kDecoderExists;
break;
case DecoderDatabase::kInvalidSampleRate:
error_code_ = kInvalidSampleRate;
break;
case DecoderDatabase::kInvalidPointer:
error_code_ = kInvalidPointer;
break;
default:
error_code_ = kOtherError;
}
return kFail;
}
return kOK;
}
int NetEqImpl::RemovePayloadType(uint8_t rtp_payload_type) {
CriticalSectionScoped lock(crit_sect_.get());
int ret = decoder_database_->Remove(rtp_payload_type);
if (ret == DecoderDatabase::kOK) {
return kOK;
} else if (ret == DecoderDatabase::kDecoderNotFound) {
error_code_ = kDecoderNotFound;
} else {
error_code_ = kOtherError;
}
return kFail;
}
bool NetEqImpl::SetMinimumDelay(int delay_ms) {
CriticalSectionScoped lock(crit_sect_.get());
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) {
CriticalSectionScoped lock(crit_sect_.get());
if (delay_ms >= 0 && delay_ms < 10000) {
assert(delay_manager_.get());
return delay_manager_->SetMaximumDelay(delay_ms);
}
return false;
}
int NetEqImpl::LeastRequiredDelayMs() const {
CriticalSectionScoped lock(crit_sect_.get());
assert(delay_manager_.get());
return delay_manager_->least_required_delay_ms();
}
int NetEqImpl::SetTargetDelay() {
return kNotImplemented;
}
int NetEqImpl::TargetDelay() {
return kNotImplemented;
}
int NetEqImpl::CurrentDelayMs() const {
CriticalSectionScoped lock(crit_sect_.get());
if (fs_hz_ == 0)
return 0;
// Sum up the samples in the packet buffer with the future length of the sync
// buffer, and divide the sum by the sample rate.
const size_t delay_samples =
packet_buffer_->NumSamplesInBuffer(decoder_database_.get(),
decoder_frame_length_) +
sync_buffer_->FutureLength();
// The division below will truncate.
const int delay_ms =
static_cast<int>(delay_samples) / rtc::CheckedDivExact(fs_hz_, 1000);
return delay_ms;
}
// Deprecated.
// TODO(henrik.lundin) Delete.
void NetEqImpl::SetPlayoutMode(NetEqPlayoutMode mode) {
CriticalSectionScoped lock(crit_sect_.get());
if (mode != playout_mode_) {
playout_mode_ = mode;
CreateDecisionLogic();
}
}
// Deprecated.
// TODO(henrik.lundin) Delete.
NetEqPlayoutMode NetEqImpl::PlayoutMode() const {
CriticalSectionScoped lock(crit_sect_.get());
return playout_mode_;
}
int NetEqImpl::NetworkStatistics(NetEqNetworkStatistics* stats) {
CriticalSectionScoped lock(crit_sect_.get());
assert(decoder_database_.get());
const size_t total_samples_in_buffers =
packet_buffer_->NumSamplesInBuffer(decoder_database_.get(),
decoder_frame_length_) +
sync_buffer_->FutureLength();
assert(delay_manager_.get());
assert(decision_logic_.get());
stats_.GetNetworkStatistics(fs_hz_, total_samples_in_buffers,
decoder_frame_length_, *delay_manager_.get(),
*decision_logic_.get(), stats);
return 0;
}
void NetEqImpl::GetRtcpStatistics(RtcpStatistics* stats) {
CriticalSectionScoped lock(crit_sect_.get());
if (stats) {
rtcp_.GetStatistics(false, stats);
}
}
void NetEqImpl::GetRtcpStatisticsNoReset(RtcpStatistics* stats) {
CriticalSectionScoped lock(crit_sect_.get());
if (stats) {
rtcp_.GetStatistics(true, stats);
}
}
void NetEqImpl::EnableVad() {
CriticalSectionScoped lock(crit_sect_.get());
assert(vad_.get());
vad_->Enable();
}
void NetEqImpl::DisableVad() {
CriticalSectionScoped lock(crit_sect_.get());
assert(vad_.get());
vad_->Disable();
}
bool NetEqImpl::GetPlayoutTimestamp(uint32_t* timestamp) {
CriticalSectionScoped lock(crit_sect_.get());
if (first_packet_) {
// We don't have a valid RTP timestamp until we have decoded our first
// RTP packet.
return false;
}
*timestamp = timestamp_scaler_->ToExternal(playout_timestamp_);
return true;
}
int NetEqImpl::SetTargetNumberOfChannels() {
return kNotImplemented;
}
int NetEqImpl::SetTargetSampleRate() {
return kNotImplemented;
}
int NetEqImpl::LastError() const {
CriticalSectionScoped lock(crit_sect_.get());
return error_code_;
}
int NetEqImpl::LastDecoderError() {
CriticalSectionScoped lock(crit_sect_.get());
return decoder_error_code_;
}
void NetEqImpl::FlushBuffers() {
CriticalSectionScoped lock(crit_sect_.get());
LOG(LS_VERBOSE) << "FlushBuffers";
packet_buffer_->Flush();
assert(sync_buffer_.get());
assert(expand_.get());
sync_buffer_->Flush();
sync_buffer_->set_next_index(sync_buffer_->next_index() -
expand_->overlap_length());
// Set to wait for new codec.
first_packet_ = true;
}
void NetEqImpl::PacketBufferStatistics(int* current_num_packets,
int* max_num_packets) const {
CriticalSectionScoped lock(crit_sect_.get());
packet_buffer_->BufferStat(current_num_packets, max_num_packets);
}
void NetEqImpl::EnableNack(size_t max_nack_list_size) {
CriticalSectionScoped lock(crit_sect_.get());
if (!nack_enabled_) {
const int kNackThresholdPackets = 2;
nack_.reset(Nack::Create(kNackThresholdPackets));
nack_enabled_ = true;
nack_->UpdateSampleRate(fs_hz_);
}
nack_->SetMaxNackListSize(max_nack_list_size);
}
void NetEqImpl::DisableNack() {
CriticalSectionScoped lock(crit_sect_.get());
nack_.reset();
nack_enabled_ = false;
}
std::vector<uint16_t> NetEqImpl::GetNackList(int64_t round_trip_time_ms) const {
CriticalSectionScoped lock(crit_sect_.get());
if (!nack_enabled_) {
return std::vector<uint16_t>();
}
RTC_DCHECK(nack_.get());
return nack_->GetNackList(round_trip_time_ms);
}
const SyncBuffer* NetEqImpl::sync_buffer_for_test() const {
CriticalSectionScoped lock(crit_sect_.get());
return sync_buffer_.get();
}
// Methods below this line are private.
int NetEqImpl::InsertPacketInternal(const WebRtcRTPHeader& rtp_header,
const uint8_t* payload,
size_t length_bytes,
uint32_t receive_timestamp,
bool is_sync_packet) {
if (!payload) {
LOG_F(LS_ERROR) << "payload == NULL";
return kInvalidPointer;
}
// Sanity checks for sync-packets.
if (is_sync_packet) {
if (decoder_database_->IsDtmf(rtp_header.header.payloadType) ||
decoder_database_->IsRed(rtp_header.header.payloadType) ||
decoder_database_->IsComfortNoise(rtp_header.header.payloadType)) {
LOG_F(LS_ERROR) << "Sync-packet with an unacceptable payload type "
<< static_cast<int>(rtp_header.header.payloadType);
return kSyncPacketNotAccepted;
}
if (first_packet_ ||
rtp_header.header.payloadType != current_rtp_payload_type_ ||
rtp_header.header.ssrc != ssrc_) {
// Even if |current_rtp_payload_type_| is 0xFF, sync-packet isn't
// accepted.
LOG_F(LS_ERROR)
<< "Changing codec, SSRC or first packet with sync-packet.";
return kSyncPacketNotAccepted;
}
}
PacketList packet_list;
RTPHeader main_header;
{
// Convert to Packet.
// Create |packet| within this separate scope, since it should not be used
// directly once it's been inserted in the packet list. This way, |packet|
// is not defined outside of this block.
Packet* packet = new Packet;
packet->header.markerBit = false;
packet->header.payloadType = rtp_header.header.payloadType;
packet->header.sequenceNumber = rtp_header.header.sequenceNumber;
packet->header.timestamp = rtp_header.header.timestamp;
packet->header.ssrc = rtp_header.header.ssrc;
packet->header.numCSRCs = 0;
packet->payload_length = length_bytes;
packet->primary = true;
packet->waiting_time = 0;
packet->payload = new uint8_t[packet->payload_length];
packet->sync_packet = is_sync_packet;
if (!packet->payload) {
LOG_F(LS_ERROR) << "Payload pointer is NULL.";
}
assert(payload); // Already checked above.
memcpy(packet->payload, payload, packet->payload_length);
// Insert packet in a packet list.
packet_list.push_back(packet);
// Save main payloads header for later.
memcpy(&main_header, &packet->header, sizeof(main_header));
}
bool update_sample_rate_and_channels = false;
// Reinitialize NetEq if it's needed (changed SSRC or first call).
if ((main_header.ssrc != ssrc_) || first_packet_) {
// Note: |first_packet_| will be cleared further down in this method, once
// the packet has been successfully inserted into the packet buffer.
rtcp_.Init(main_header.sequenceNumber);
// Flush the packet buffer and DTMF buffer.
packet_buffer_->Flush();
dtmf_buffer_->Flush();
// Store new SSRC.
ssrc_ = main_header.ssrc;
// Update audio buffer timestamp.
sync_buffer_->IncreaseEndTimestamp(main_header.timestamp - timestamp_);
// Update codecs.
timestamp_ = main_header.timestamp;
current_rtp_payload_type_ = main_header.payloadType;
// Reset timestamp scaling.
timestamp_scaler_->Reset();
// Trigger an update of sampling rate and the number of channels.
update_sample_rate_and_channels = true;
}
// Update RTCP statistics, only for regular packets.
if (!is_sync_packet)
rtcp_.Update(main_header, receive_timestamp);
// Check for RED payload type, and separate payloads into several packets.
if (decoder_database_->IsRed(main_header.payloadType)) {
assert(!is_sync_packet); // We had a sanity check for this.
if (payload_splitter_->SplitRed(&packet_list) != PayloadSplitter::kOK) {
PacketBuffer::DeleteAllPackets(&packet_list);
return kRedundancySplitError;
}
// Only accept a few RED payloads of the same type as the main data,
// DTMF events and CNG.
payload_splitter_->CheckRedPayloads(&packet_list, *decoder_database_);
// Update the stored main payload header since the main payload has now
// changed.
memcpy(&main_header, &packet_list.front()->header, sizeof(main_header));
}
// Check payload types.
if (decoder_database_->CheckPayloadTypes(packet_list) ==
DecoderDatabase::kDecoderNotFound) {
PacketBuffer::DeleteAllPackets(&packet_list);
return kUnknownRtpPayloadType;
}
// Scale timestamp to internal domain (only for some codecs).
timestamp_scaler_->ToInternal(&packet_list);
// Process DTMF payloads. Cycle through the list of packets, and pick out any
// DTMF payloads found.
PacketList::iterator it = packet_list.begin();
while (it != packet_list.end()) {
Packet* current_packet = (*it);
assert(current_packet);
assert(current_packet->payload);
if (decoder_database_->IsDtmf(current_packet->header.payloadType)) {
assert(!current_packet->sync_packet); // We had a sanity check for this.
DtmfEvent event;
int ret = DtmfBuffer::ParseEvent(
current_packet->header.timestamp,
current_packet->payload,
current_packet->payload_length,
&event);
if (ret != DtmfBuffer::kOK) {
PacketBuffer::DeleteAllPackets(&packet_list);
return kDtmfParsingError;
}
if (dtmf_buffer_->InsertEvent(event) != DtmfBuffer::kOK) {
PacketBuffer::DeleteAllPackets(&packet_list);
return kDtmfInsertError;
}
// TODO(hlundin): Let the destructor of Packet handle the payload.
delete [] current_packet->payload;
delete current_packet;
it = packet_list.erase(it);
} else {
++it;
}
}
// Check for FEC in packets, and separate payloads into several packets.
int ret = payload_splitter_->SplitFec(&packet_list, decoder_database_.get());
if (ret != PayloadSplitter::kOK) {
PacketBuffer::DeleteAllPackets(&packet_list);
switch (ret) {
case PayloadSplitter::kUnknownPayloadType:
return kUnknownRtpPayloadType;
default:
return kOtherError;
}
}
// Split payloads into smaller chunks. This also verifies that all payloads
// are of a known payload type. SplitAudio() method is protected against
// sync-packets.
ret = payload_splitter_->SplitAudio(&packet_list, *decoder_database_);
if (ret != PayloadSplitter::kOK) {
PacketBuffer::DeleteAllPackets(&packet_list);
switch (ret) {
case PayloadSplitter::kUnknownPayloadType:
return kUnknownRtpPayloadType;
case PayloadSplitter::kFrameSplitError:
return kFrameSplitError;
default:
return kOtherError;
}
}
// Update bandwidth estimate, if the packet is not sync-packet.
if (!packet_list.empty() && !packet_list.front()->sync_packet) {
// The list can be empty here if we got nothing but DTMF payloads.
AudioDecoder* decoder =
decoder_database_->GetDecoder(main_header.payloadType);
assert(decoder); // Should always get a valid object, since we have
// already checked that the payload types are known.
decoder->IncomingPacket(packet_list.front()->payload,
packet_list.front()->payload_length,
packet_list.front()->header.sequenceNumber,
packet_list.front()->header.timestamp,
receive_timestamp);
}
if (nack_enabled_) {
RTC_DCHECK(nack_);
if (update_sample_rate_and_channels) {
nack_->Reset();
}
nack_->UpdateLastReceivedPacket(packet_list.front()->header.sequenceNumber,
packet_list.front()->header.timestamp);
}
// Insert packets in buffer.
const size_t buffer_length_before_insert =
packet_buffer_->NumPacketsInBuffer();
ret = packet_buffer_->InsertPacketList(
&packet_list,
*decoder_database_,
&current_rtp_payload_type_,
&current_cng_rtp_payload_type_);
if (ret == PacketBuffer::kFlushed) {
// Reset DSP timestamp etc. if packet buffer flushed.
new_codec_ = true;
update_sample_rate_and_channels = true;
} else if (ret != PacketBuffer::kOK) {
PacketBuffer::DeleteAllPackets(&packet_list);
return kOtherError;
}
if (first_packet_) {
first_packet_ = false;
// Update the codec on the next GetAudio call.
new_codec_ = true;
}
if (current_rtp_payload_type_ != 0xFF) {
const DecoderDatabase::DecoderInfo* dec_info =
decoder_database_->GetDecoderInfo(current_rtp_payload_type_);
if (!dec_info) {
assert(false); // Already checked that the payload type is known.
}
}
if (update_sample_rate_and_channels && !packet_buffer_->Empty()) {
// We do not use |current_rtp_payload_type_| to |set payload_type|, but
// get the next RTP header from |packet_buffer_| to obtain the payload type.
// The reason for it is the following corner case. If NetEq receives a
// CNG packet with a sample rate different than the current CNG then it
// flushes its buffer, assuming send codec must have been changed. However,
// payload type of the hypothetically new send codec is not known.
const RTPHeader* rtp_header = packet_buffer_->NextRtpHeader();
assert(rtp_header);
int payload_type = rtp_header->payloadType;
AudioDecoder* decoder = decoder_database_->GetDecoder(payload_type);
assert(decoder); // Payloads are already checked to be valid.
const DecoderDatabase::DecoderInfo* decoder_info =
decoder_database_->GetDecoderInfo(payload_type);
assert(decoder_info);
if (decoder_info->fs_hz != fs_hz_ ||
decoder->Channels() != algorithm_buffer_->Channels()) {
SetSampleRateAndChannels(decoder_info->fs_hz, decoder->Channels());
}
if (nack_enabled_) {
RTC_DCHECK(nack_);
// Update the sample rate even if the rate is not new, because of Reset().
nack_->UpdateSampleRate(fs_hz_);
}
}
// TODO(hlundin): Move this code to DelayManager class.
const DecoderDatabase::DecoderInfo* dec_info =
decoder_database_->GetDecoderInfo(main_header.payloadType);
assert(dec_info); // Already checked that the payload type is known.
delay_manager_->LastDecoderType(dec_info->codec_type);
if (delay_manager_->last_pack_cng_or_dtmf() == 0) {
// Calculate the total speech length carried in each packet.
const size_t buffer_length_after_insert =
packet_buffer_->NumPacketsInBuffer();
if (buffer_length_after_insert > buffer_length_before_insert) {
const size_t packet_length_samples =
(buffer_length_after_insert - buffer_length_before_insert) *
decoder_frame_length_;
if (packet_length_samples != decision_logic_->packet_length_samples()) {
decision_logic_->set_packet_length_samples(packet_length_samples);
delay_manager_->SetPacketAudioLength(
rtc::checked_cast<int>((1000 * packet_length_samples) / fs_hz_));
}
}
// Update statistics.
if ((int32_t) (main_header.timestamp - timestamp_) >= 0 &&
!new_codec_) {
// Only update statistics if incoming packet is not older than last played
// out packet, and if new codec flag is not set.
delay_manager_->Update(main_header.sequenceNumber, main_header.timestamp,
fs_hz_);
}
} else if (delay_manager_->last_pack_cng_or_dtmf() == -1) {
// This is first "normal" packet after CNG or DTMF.
// Reset packet time counter and measure time until next packet,
// but don't update statistics.
delay_manager_->set_last_pack_cng_or_dtmf(0);
delay_manager_->ResetPacketIatCount();
}
return 0;
}
int NetEqImpl::GetAudioInternal(size_t max_length,
int16_t* output,
size_t* samples_per_channel,
int* num_channels) {
PacketList packet_list;
DtmfEvent dtmf_event;
Operations operation;
bool play_dtmf;
int return_value = GetDecision(&operation, &packet_list, &dtmf_event,
&play_dtmf);
if (return_value != 0) {
last_mode_ = kModeError;
return return_value;
}
LOG(LS_VERBOSE) << "GetDecision returned operation=" << operation <<
" and " << packet_list.size() << " packet(s)";
AudioDecoder::SpeechType speech_type;
int length = 0;
int decode_return_value = Decode(&packet_list, &operation,
&length, &speech_type);
assert(vad_.get());
bool sid_frame_available =
(operation == kRfc3389Cng && !packet_list.empty());
vad_->Update(decoded_buffer_.get(), static_cast<size_t>(length), speech_type,
sid_frame_available, fs_hz_);
algorithm_buffer_->Clear();
switch (operation) {
case kNormal: {
DoNormal(decoded_buffer_.get(), length, speech_type, play_dtmf);
break;
}
case kMerge: {
DoMerge(decoded_buffer_.get(), length, speech_type, play_dtmf);
break;
}
case kExpand: {
return_value = DoExpand(play_dtmf);
break;
}
case kAccelerate:
case kFastAccelerate: {
const bool fast_accelerate =
enable_fast_accelerate_ && (operation == kFastAccelerate);
return_value = DoAccelerate(decoded_buffer_.get(), length, speech_type,
play_dtmf, fast_accelerate);
break;
}
case kPreemptiveExpand: {
return_value = DoPreemptiveExpand(decoded_buffer_.get(), length,
speech_type, play_dtmf);
break;
}
case kRfc3389Cng:
case kRfc3389CngNoPacket: {
return_value = DoRfc3389Cng(&packet_list, play_dtmf);
break;
}
case kCodecInternalCng: {
// This handles the case when there is no transmission and the decoder
// should produce internal comfort noise.
// TODO(hlundin): Write test for codec-internal CNG.
DoCodecInternalCng(decoded_buffer_.get(), length);
break;
}
case kDtmf: {
// TODO(hlundin): Write test for this.
return_value = DoDtmf(dtmf_event, &play_dtmf);
break;
}
case kAlternativePlc: {
// TODO(hlundin): Write test for this.
DoAlternativePlc(false);
break;
}
case kAlternativePlcIncreaseTimestamp: {
// TODO(hlundin): Write test for this.
DoAlternativePlc(true);
break;
}
case kAudioRepetitionIncreaseTimestamp: {
// TODO(hlundin): Write test for this.
sync_buffer_->IncreaseEndTimestamp(
static_cast<uint32_t>(output_size_samples_));
// Skipping break on purpose. Execution should move on into the
// next case.
FALLTHROUGH();
}
case kAudioRepetition: {
// TODO(hlundin): Write test for this.
// Copy last |output_size_samples_| from |sync_buffer_| to
// |algorithm_buffer|.
algorithm_buffer_->PushBackFromIndex(
*sync_buffer_, sync_buffer_->Size() - output_size_samples_);
expand_->Reset();
break;
}
case kUndefined: {
LOG(LS_ERROR) << "Invalid operation kUndefined.";
assert(false); // This should not happen.
last_mode_ = kModeError;
return kInvalidOperation;
}
} // End of switch.
if (return_value < 0) {
return return_value;
}
if (last_mode_ != kModeRfc3389Cng) {
comfort_noise_->Reset();
}
// Copy from |algorithm_buffer| to |sync_buffer_|.
sync_buffer_->PushBack(*algorithm_buffer_);
// Extract data from |sync_buffer_| to |output|.
size_t num_output_samples_per_channel = output_size_samples_;
size_t num_output_samples = output_size_samples_ * sync_buffer_->Channels();
if (num_output_samples > max_length) {
LOG(LS_WARNING) << "Output array is too short. " << max_length << " < " <<
output_size_samples_ << " * " << sync_buffer_->Channels();
num_output_samples = max_length;
num_output_samples_per_channel = max_length / sync_buffer_->Channels();
}
const size_t samples_from_sync =
sync_buffer_->GetNextAudioInterleaved(num_output_samples_per_channel,
output);
*num_channels = static_cast<int>(sync_buffer_->Channels());
LOG(LS_VERBOSE) << "Sync buffer (" << *num_channels << " channel(s)):" <<
" insert " << algorithm_buffer_->Size() << " samples, extract " <<
samples_from_sync << " samples";
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 (samples_from_sync != output_size_samples_) {
LOG(LS_ERROR) << "samples_from_sync (" << samples_from_sync
<< ") != output_size_samples_ (" << output_size_samples_
<< ")";
// TODO(minyue): treatment of under-run, filling zeros
memset(output, 0, num_output_samples * sizeof(int16_t));
*samples_per_channel = output_size_samples_;
return kSampleUnderrun;
}
*samples_per_channel = output_size_samples_;
// Should always have overlap samples left in the |sync_buffer_|.
RTC_DCHECK_GE(sync_buffer_->FutureLength(), expand_->overlap_length());
if (play_dtmf) {
return_value = DtmfOverdub(dtmf_event, sync_buffer_->Channels(), output);
}
// Update the background noise parameters if last operation wrote data
// straight from the decoder to the |sync_buffer_|. That is, none of the
// operations that modify the signal can be followed by a parameter update.
if ((last_mode_ == kModeNormal) ||
(last_mode_ == kModeAccelerateFail) ||
(last_mode_ == kModePreemptiveExpandFail) ||
(last_mode_ == kModeRfc3389Cng) ||
(last_mode_ == kModeCodecInternalCng)) {
background_noise_->Update(*sync_buffer_, *vad_.get());
}
if (operation == kDtmf) {
// DTMF data was written the end of |sync_buffer_|.
// Update index to end of DTMF data in |sync_buffer_|.
sync_buffer_->set_dtmf_index(sync_buffer_->Size());
}
if (last_mode_ != kModeExpand) {
// If last operation was not expand, calculate the |playout_timestamp_| from
// the |sync_buffer_|. However, do not update the |playout_timestamp_| if it
// would be moved "backwards".
uint32_t temp_timestamp = sync_buffer_->end_timestamp() -
static_cast<uint32_t>(sync_buffer_->FutureLength());
if (static_cast<int32_t>(temp_timestamp - playout_timestamp_) > 0) {
playout_timestamp_ = temp_timestamp;
}
} else {
// Use dead reckoning to estimate the |playout_timestamp_|.
playout_timestamp_ += static_cast<uint32_t>(output_size_samples_);
}
if (decode_return_value) return decode_return_value;
return return_value;
}
int NetEqImpl::GetDecision(Operations* operation,
PacketList* packet_list,
DtmfEvent* dtmf_event,
bool* play_dtmf) {
// Initialize output variables.
*play_dtmf = false;
*operation = kUndefined;
// Increment time counters.
packet_buffer_->IncrementWaitingTimes();
stats_.IncreaseCounter(output_size_samples_, fs_hz_);
assert(sync_buffer_.get());
uint32_t end_timestamp = sync_buffer_->end_timestamp();
if (!new_codec_) {
const uint32_t five_seconds_samples = 5 * fs_hz_;
packet_buffer_->DiscardOldPackets(end_timestamp, five_seconds_samples);
}
const RTPHeader* header = packet_buffer_->NextRtpHeader();
if (decision_logic_->CngRfc3389On() || last_mode_ == kModeRfc3389Cng) {
// Because of timestamp peculiarities, we have to "manually" disallow using
// a CNG packet with the same timestamp as the one that was last played.
// This can happen when using redundancy and will cause the timing to shift.
while (header && decoder_database_->IsComfortNoise(header->payloadType) &&
(end_timestamp >= header->timestamp ||
end_timestamp + decision_logic_->generated_noise_samples() >
header->timestamp)) {
// Don't use this packet, discard it.
if (packet_buffer_->DiscardNextPacket() != PacketBuffer::kOK) {
assert(false); // Must be ok by design.
}
// Check buffer again.
if (!new_codec_) {
packet_buffer_->DiscardOldPackets(end_timestamp, 5 * fs_hz_);
}
header = packet_buffer_->NextRtpHeader();
}
}
assert(expand_.get());
const int samples_left = static_cast<int>(sync_buffer_->FutureLength() -
expand_->overlap_length());
if (last_mode_ == kModeAccelerateSuccess ||
last_mode_ == kModeAccelerateLowEnergy ||
last_mode_ == kModePreemptiveExpandSuccess ||
last_mode_ == kModePreemptiveExpandLowEnergy) {
// Subtract (samples_left + output_size_samples_) from sampleMemory.
decision_logic_->AddSampleMemory(
-(samples_left + rtc::checked_cast<int>(output_size_samples_)));
}
// Check if it is time to play a DTMF event.
if (dtmf_buffer_->GetEvent(
static_cast<uint32_t>(
end_timestamp + decision_logic_->generated_noise_samples()),
dtmf_event)) {
*play_dtmf = true;
}
// Get instruction.
assert(sync_buffer_.get());
assert(expand_.get());
*operation = decision_logic_->GetDecision(*sync_buffer_,
*expand_,
decoder_frame_length_,
header,
last_mode_,
*play_dtmf,
&reset_decoder_);
// Check if we already have enough samples in the |sync_buffer_|. If so,
// change decision to normal, unless the decision was merge, accelerate, or
// preemptive expand.
if (samples_left >= rtc::checked_cast<int>(output_size_samples_) &&
*operation != kMerge &&
*operation != kAccelerate &&
*operation != kFastAccelerate &&
*operation != kPreemptiveExpand) {
*operation = kNormal;
return 0;
}
decision_logic_->ExpandDecision(*operation);
// Check conditions for reset.
if (new_codec_ || *operation == kUndefined) {
// The only valid reason to get kUndefined is that new_codec_ is set.
assert(new_codec_);
if (*play_dtmf && !header) {
timestamp_ = dtmf_event->timestamp;
} else {
if (!header) {
LOG(LS_ERROR) << "Packet missing where it shouldn't.";
return -1;
}
timestamp_ = header->timestamp;
if (*operation == kRfc3389CngNoPacket
#ifndef LEGACY_BITEXACT
// Without this check, it can happen that a non-CNG packet is sent to
// the CNG decoder as if it was a SID frame. This is clearly a bug,
// but is kept for now to maintain bit-exactness with the test
// vectors.
&& decoder_database_->IsComfortNoise(header->payloadType)
#endif
) {
// 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;
if (decision_logic_->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>(decision_logic_->generated_noise_samples());
sync_buffer_->IncreaseEndTimestamp(timestamp_jump);
timestamp_ += timestamp_jump;
}
decision_logic_->set_generated_noise_samples(0);
return 0;
}
case kAccelerate:
case kFastAccelerate: {
// In order to do an accelerate we need at least 30 ms of audio data.
if (samples_left >= static_cast<int>(samples_30_ms)) {
// Already have enough data, so we do not need to extract any more.
decision_logic_->set_sample_memory(samples_left);
decision_logic_->set_prev_time_scale(true);
return 0;
} else if (samples_left >= static_cast<int>(samples_10_ms) &&
decoder_frame_length_ >= samples_30_ms) {
// Avoid decoding more data as it might overflow the playout buffer.
*operation = kNormal;
return 0;
} else if (samples_left < static_cast<int>(samples_20_ms) &&
decoder_frame_length_ < samples_30_ms) {
// Build up decoded data by decoding at least 20 ms of audio data. Do
// not perform accelerate yet, but wait until we only need to do one
// decoding.
required_samples = 2 * output_size_samples_;
*operation = kNormal;
}
// If none of the above is true, we have one of two possible situations:
// (1) 20 ms <= samples_left < 30 ms and decoder_frame_length_ < 30 ms; or
// (2) samples_left < 10 ms and decoder_frame_length_ >= 30 ms.
// In either case, we move on with the accelerate decision, and decode one
// frame now.
break;
}
case kPreemptiveExpand: {
// In order to do a preemptive expand we need at least 30 ms of decoded
// audio data.
if ((samples_left >= static_cast<int>(samples_30_ms)) ||
(samples_left >= static_cast<int>(samples_10_ms) &&
decoder_frame_length_ >= samples_30_ms)) {
// Already have enough data, so we do not need to extract any more.
// Or, avoid decoding more data as it might overflow the playout buffer.
// Still try preemptive expand, though.
decision_logic_->set_sample_memory(samples_left);
decision_logic_->set_prev_time_scale(true);
return 0;
}
if (samples_left < static_cast<int>(samples_20_ms) &&
decoder_frame_length_ < samples_30_ms) {
// Build up decoded data by decoding at least 20 ms of audio data.
// Still try to perform preemptive expand.
required_samples = 2 * output_size_samples_;
}
// Move on with the preemptive expand decision.
break;
}
case kMerge: {
required_samples =
std::max(merge_->RequiredFutureSamples(), required_samples);
break;
}
default: {
// Do nothing.
}
}
// Get packets from buffer.
int extracted_samples = 0;
if (header &&
*operation != kAlternativePlc &&
*operation != kAlternativePlcIncreaseTimestamp &&
*operation != kAudioRepetition &&
*operation != kAudioRepetitionIncreaseTimestamp) {
sync_buffer_->IncreaseEndTimestamp(header->timestamp - end_timestamp);
if (decision_logic_->CngOff()) {
// Adjustment of timestamp only corresponds to an actual packet loss
// if comfort noise is not played. If comfort noise was just played,
// this adjustment of timestamp is only done to get back in sync with the
// stream timestamp; no loss to report.
stats_.LostSamples(header->timestamp - end_timestamp);
}
if (*operation != kRfc3389Cng) {
// We are about to decode and use a non-CNG packet.
decision_logic_->SetCngOff();
}
// Reset CNG timestamp as a new packet will be delivered.
// (Also if this is a CNG packet, since playedOutTS is updated.)
decision_logic_->set_generated_noise_samples(0);
extracted_samples = ExtractPackets(required_samples, packet_list);
if (extracted_samples < 0) {
return kPacketBufferCorruption;
}
}
if (*operation == kAccelerate || *operation == kFastAccelerate ||
*operation == kPreemptiveExpand) {
decision_logic_->set_sample_memory(samples_left + extracted_samples);
decision_logic_->set_prev_time_scale(true);
}
if (*operation == kAccelerate || *operation == kFastAccelerate) {
// Check that we have enough data (30ms) to do accelerate.
if (extracted_samples + samples_left < static_cast<int>(samples_30_ms)) {
// TODO(hlundin): Write test for this.
// Not enough, do normal operation instead.
*operation = kNormal;
}
}
timestamp_ = end_timestamp;
return 0;
}
int NetEqImpl::Decode(PacketList* packet_list, Operations* operation,
int* decoded_length,
AudioDecoder::SpeechType* speech_type) {
*speech_type = AudioDecoder::kSpeech;
// When packet_list is empty, we may be in kCodecInternalCng mode, and for
// that we use current active decoder.
AudioDecoder* decoder = decoder_database_->GetActiveDecoder();
if (!packet_list->empty()) {
const Packet* packet = packet_list->front();
uint8_t payload_type = packet->header.payloadType;
if (!decoder_database_->IsComfortNoise(payload_type)) {
decoder = decoder_database_->GetDecoder(payload_type);
assert(decoder);
if (!decoder) {
LOG(LS_WARNING) << "Unknown payload type "
<< static_cast<int>(payload_type);
PacketBuffer::DeleteAllPackets(packet_list);
return kDecoderNotFound;
}
bool decoder_changed;
decoder_database_->SetActiveDecoder(payload_type, &decoder_changed);
if (decoder_changed) {
// We have a new decoder. Re-init some values.
const DecoderDatabase::DecoderInfo* decoder_info = decoder_database_
->GetDecoderInfo(payload_type);
assert(decoder_info);
if (!decoder_info) {
LOG(LS_WARNING) << "Unknown payload type "
<< static_cast<int>(payload_type);
PacketBuffer::DeleteAllPackets(packet_list);
return kDecoderNotFound;
}
// If sampling rate or number of channels has changed, we need to make
// a reset.
if (decoder_info->fs_hz != fs_hz_ ||
decoder->Channels() != algorithm_buffer_->Channels()) {
// TODO(tlegrand): Add unittest to cover this event.
SetSampleRateAndChannels(decoder_info->fs_hz, 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.
AudioDecoder* cng_decoder = decoder_database_->GetActiveCngDecoder();
if (cng_decoder)
cng_decoder->Reset();
reset_decoder_ = false;
}
#ifdef LEGACY_BITEXACT
// Due to a bug in old SignalMCU, it could happen that CNG operation was
// decided, but a speech packet was provided. The speech packet will be used
// to update the comfort noise decoder, as if it was a SID frame, which is
// clearly wrong.
if (*operation == kRfc3389Cng) {
return 0;
}
#endif
*decoded_length = 0;
// Update codec-internal PLC state.
if ((*operation == kMerge) && decoder && decoder->HasDecodePlc()) {
decoder->DecodePlc(1, &decoded_buffer_[*decoded_length]);
}
int return_value;
if (*operation == kCodecInternalCng) {
RTC_DCHECK(packet_list->empty());
return_value = DecodeCng(decoder, decoded_length, speech_type);
} else {
return_value = DecodeLoop(packet_list, *operation, decoder,
decoded_length, speech_type);
}
if (*decoded_length < 0) {
// Error returned from the decoder.
*decoded_length = 0;
sync_buffer_->IncreaseEndTimestamp(
static_cast<uint32_t>(decoder_frame_length_));
int error_code = 0;
if (decoder)
error_code = decoder->ErrorCode();
if (error_code != 0) {
// Got some error code from the decoder.
decoder_error_code_ = error_code;
return_value = kDecoderErrorCode;
LOG(LS_WARNING) << "Decoder returned error code: " << error_code;
} else {
// Decoder does not implement error codes. Return generic error.
return_value = kOtherDecoderError;
LOG(LS_WARNING) << "Decoder error (no error code)";
}
*operation = kExpand; // Do expansion to get data instead.
}
if (*speech_type != AudioDecoder::kComfortNoise) {
// Don't increment timestamp if codec returned CNG speech type
// since in this case, the we will increment the CNGplayedTS counter.
// Increase with number of samples per channel.
assert(*decoded_length == 0 ||
(decoder && decoder->Channels() == sync_buffer_->Channels()));
sync_buffer_->IncreaseEndTimestamp(
*decoded_length / static_cast<int>(sync_buffer_->Channels()));
}
return return_value;
}
int NetEqImpl::DecodeCng(AudioDecoder* decoder, int* decoded_length,
AudioDecoder::SpeechType* speech_type) {
if (!decoder) {
// This happens when active decoder is not defined.
*decoded_length = -1;
return 0;
}
while (*decoded_length < rtc::checked_cast<int>(output_size_samples_)) {
const int length = decoder->Decode(
nullptr, 0, fs_hz_,
(decoded_buffer_length_ - *decoded_length) * sizeof(int16_t),
&decoded_buffer_[*decoded_length], speech_type);
if (length > 0) {
*decoded_length += length;
LOG(LS_VERBOSE) << "Decoded " << length << " CNG samples";
} else {
// Error.
LOG(LS_WARNING) << "Failed to decode CNG";
*decoded_length = -1;
break;
}
if (*decoded_length > static_cast<int>(decoded_buffer_length_)) {
// Guard against overflow.
LOG(LS_WARNING) << "Decoded too much CNG.";
return kDecodedTooMuch;
}
}
return 0;
}
int NetEqImpl::DecodeLoop(PacketList* packet_list, const Operations& operation,
AudioDecoder* decoder, int* decoded_length,
AudioDecoder::SpeechType* speech_type) {
Packet* packet = NULL;
if (!packet_list->empty()) {
packet = packet_list->front();
}
// Do decoding.
while (packet &&
!decoder_database_->IsComfortNoise(packet->header.payloadType)) {
assert(decoder); // At this point, we must have a decoder object.
// The number of channels in the |sync_buffer_| should be the same as the
// number decoder channels.
assert(sync_buffer_->Channels() == decoder->Channels());
assert(decoded_buffer_length_ >= kMaxFrameSize * decoder->Channels());
assert(operation == kNormal || operation == kAccelerate ||
operation == kFastAccelerate || operation == kMerge ||
operation == kPreemptiveExpand);
packet_list->pop_front();
size_t payload_length = packet->payload_length;
int decode_length;
if (packet->sync_packet) {
// Decode to silence with the same frame size as the last decode.
LOG(LS_VERBOSE) << "Decoding sync-packet: " <<
" ts=" << packet->header.timestamp <<
", sn=" << packet->header.sequenceNumber <<
", pt=" << static_cast<int>(packet->header.payloadType) <<
", ssrc=" << packet->header.ssrc <<
", len=" << packet->payload_length;
memset(&decoded_buffer_[*decoded_length], 0,
decoder_frame_length_ * decoder->Channels() *
sizeof(decoded_buffer_[0]));
decode_length = rtc::checked_cast<int>(decoder_frame_length_);
} else if (!packet->primary) {
// This is a redundant payload; call the special decoder method.
LOG(LS_VERBOSE) << "Decoding packet (redundant):" <<
" ts=" << packet->header.timestamp <<
", sn=" << packet->header.sequenceNumber <<
", pt=" << static_cast<int>(packet->header.payloadType) <<
", ssrc=" << packet->header.ssrc <<
", len=" << packet->payload_length;
decode_length = decoder->DecodeRedundant(
packet->payload, packet->payload_length, fs_hz_,
(decoded_buffer_length_ - *decoded_length) * sizeof(int16_t),
&decoded_buffer_[*decoded_length], speech_type);
} else {
LOG(LS_VERBOSE) << "Decoding packet: ts=" << packet->header.timestamp <<
", sn=" << packet->header.sequenceNumber <<
", pt=" << static_cast<int>(packet->header.payloadType) <<
", ssrc=" << packet->header.ssrc <<
", len=" << packet->payload_length;
decode_length =
decoder->Decode(
packet->payload, packet->payload_length, fs_hz_,
(decoded_buffer_length_ - *decoded_length) * sizeof(int16_t),
&decoded_buffer_[*decoded_length], speech_type);
}
delete[] packet->payload;
delete packet;
packet = NULL;
if (decode_length > 0) {
*decoded_length += decode_length;
// Update |decoder_frame_length_| with number of samples per channel.
decoder_frame_length_ =
static_cast<size_t>(decode_length) / decoder->Channels();
LOG(LS_VERBOSE) << "Decoded " << decode_length << " samples ("
<< decoder->Channels() << " channel(s) -> "
<< decoder_frame_length_ << " samples per channel)";
} else if (decode_length < 0) {
// Error.
LOG(LS_WARNING) << "Decode " << decode_length << " " << payload_length;
*decoded_length = -1;
PacketBuffer::DeleteAllPackets(packet_list);
break;
}
if (*decoded_length > static_cast<int>(decoded_buffer_length_)) {
// Guard against overflow.
LOG(LS_WARNING) << "Decoded too much.";
PacketBuffer::DeleteAllPackets(packet_list);
return kDecodedTooMuch;
}
if (!packet_list->empty()) {
packet = packet_list->front();
} else {
packet = NULL;
}
} // End of decode loop.
// If the list is not empty at this point, either a decoding error terminated
// the while-loop, or list must hold exactly one CNG packet.
assert(packet_list->empty() || *decoded_length < 0 ||
(packet_list->size() == 1 && packet &&
decoder_database_->IsComfortNoise(packet->header.payloadType)));
return 0;
}
void NetEqImpl::DoNormal(const int16_t* decoded_buffer, size_t decoded_length,
AudioDecoder::SpeechType speech_type, bool play_dtmf) {
assert(normal_.get());
assert(mute_factor_array_.get());
normal_->Process(decoded_buffer, decoded_length, last_mode_,
mute_factor_array_.get(), algorithm_buffer_.get());
if (decoded_length != 0) {
last_mode_ = kModeNormal;
}
// If last packet was decoded as an inband CNG, set mode to CNG instead.
if ((speech_type == AudioDecoder::kComfortNoise)
|| ((last_mode_ == kModeCodecInternalCng)
&& (decoded_length == 0))) {
// TODO(hlundin): Remove second part of || statement above.
last_mode_ = kModeCodecInternalCng;
}
if (!play_dtmf) {
dtmf_tone_generator_->Reset();
}
}
void NetEqImpl::DoMerge(int16_t* decoded_buffer, size_t decoded_length,
AudioDecoder::SpeechType speech_type, bool play_dtmf) {
assert(mute_factor_array_.get());
assert(merge_.get());
size_t new_length = merge_->Process(decoded_buffer, decoded_length,
mute_factor_array_.get(),
algorithm_buffer_.get());
size_t expand_length_correction = new_length -
decoded_length / algorithm_buffer_->Channels();
// Update in-call and post-call statistics.
if (expand_->MuteFactor(0) == 0) {
// Expand generates only noise.
stats_.ExpandedNoiseSamples(expand_length_correction);
} else {
// Expansion generates more than only noise.
stats_.ExpandedVoiceSamples(expand_length_correction);
}
last_mode_ = kModeMerge;
// If last packet was decoded as an inband CNG, set mode to CNG instead.
if (speech_type == AudioDecoder::kComfortNoise) {
last_mode_ = kModeCodecInternalCng;
}
expand_->Reset();
if (!play_dtmf) {
dtmf_tone_generator_->Reset();
}
}
int NetEqImpl::DoExpand(bool play_dtmf) {
while ((sync_buffer_->FutureLength() - expand_->overlap_length()) <
output_size_samples_) {
algorithm_buffer_->Clear();
int return_value = expand_->Process(algorithm_buffer_.get());
size_t length = algorithm_buffer_->Size();
// Update in-call and post-call statistics.
if (expand_->MuteFactor(0) == 0) {
// Expand operation generates only noise.
stats_.ExpandedNoiseSamples(length);
} else {
// Expand operation generates more than only noise.
stats_.ExpandedVoiceSamples(length);
}
last_mode_ = kModeExpand;
if (return_value < 0) {
return return_value;
}
sync_buffer_->PushBack(*algorithm_buffer_);
algorithm_buffer_->Clear();
}
if (!play_dtmf) {
dtmf_tone_generator_->Reset();
}
return 0;
}
int NetEqImpl::DoAccelerate(int16_t* decoded_buffer,
size_t decoded_length,
AudioDecoder::SpeechType speech_type,
bool play_dtmf,
bool fast_accelerate) {
const size_t required_samples =
static_cast<size_t>(240 * fs_mult_); // Must have 30 ms.
size_t borrowed_samples_per_channel = 0;
size_t num_channels = algorithm_buffer_->Channels();
size_t decoded_length_per_channel = decoded_length / num_channels;
if (decoded_length_per_channel < required_samples) {
// Must move data from the |sync_buffer_| in order to get 30 ms.
borrowed_samples_per_channel = static_cast<int>(required_samples -
decoded_length_per_channel);
memmove(&decoded_buffer[borrowed_samples_per_channel * num_channels],
decoded_buffer,
sizeof(int16_t) * decoded_length);
sync_buffer_->ReadInterleavedFromEnd(borrowed_samples_per_channel,
decoded_buffer);
decoded_length = required_samples * num_channels;
}
size_t samples_removed;
Accelerate::ReturnCodes return_code =
accelerate_->Process(decoded_buffer, decoded_length, fast_accelerate,
algorithm_buffer_.get(), &samples_removed);
stats_.AcceleratedSamples(samples_removed);
switch (return_code) {
case Accelerate::kSuccess:
last_mode_ = kModeAccelerateSuccess;
break;
case Accelerate::kSuccessLowEnergy:
last_mode_ = kModeAccelerateLowEnergy;
break;
case Accelerate::kNoStretch:
last_mode_ = kModeAccelerateFail;
break;
case Accelerate::kError:
// TODO(hlundin): Map to kModeError instead?
last_mode_ = kModeAccelerateFail;
return kAccelerateError;
}
if (borrowed_samples_per_channel > 0) {
// Copy borrowed samples back to the |sync_buffer_|.
size_t length = algorithm_buffer_->Size();
if (length < borrowed_samples_per_channel) {
// This destroys the beginning of the buffer, but will not cause any
// problems.
sync_buffer_->ReplaceAtIndex(*algorithm_buffer_,
sync_buffer_->Size() -
borrowed_samples_per_channel);
sync_buffer_->PushFrontZeros(borrowed_samples_per_channel - length);
algorithm_buffer_->PopFront(length);
assert(algorithm_buffer_->Empty());
} else {
sync_buffer_->ReplaceAtIndex(*algorithm_buffer_,
borrowed_samples_per_channel,
sync_buffer_->Size() -
borrowed_samples_per_channel);
algorithm_buffer_->PopFront(borrowed_samples_per_channel);
}
}
// If last packet was decoded as an inband CNG, set mode to CNG instead.
if (speech_type == AudioDecoder::kComfortNoise) {
last_mode_ = kModeCodecInternalCng;
}
if (!play_dtmf) {
dtmf_tone_generator_->Reset();
}
expand_->Reset();
return 0;
}
int NetEqImpl::DoPreemptiveExpand(int16_t* decoded_buffer,
size_t decoded_length,
AudioDecoder::SpeechType speech_type,
bool play_dtmf) {
const size_t required_samples =
static_cast<size_t>(240 * fs_mult_); // Must have 30 ms.
size_t num_channels = algorithm_buffer_->Channels();
size_t borrowed_samples_per_channel = 0;
size_t old_borrowed_samples_per_channel = 0;
size_t decoded_length_per_channel = decoded_length / num_channels;
if (decoded_length_per_channel < required_samples) {
// Must move data from the |sync_buffer_| in order to get 30 ms.
borrowed_samples_per_channel =
required_samples - decoded_length_per_channel;
// Calculate how many of these were already played out.
old_borrowed_samples_per_channel =
(borrowed_samples_per_channel > sync_buffer_->FutureLength()) ?
(borrowed_samples_per_channel - sync_buffer_->FutureLength()) : 0;
memmove(&decoded_buffer[borrowed_samples_per_channel * num_channels],
decoded_buffer,
sizeof(int16_t) * decoded_length);
sync_buffer_->ReadInterleavedFromEnd(borrowed_samples_per_channel,
decoded_buffer);
decoded_length = required_samples * num_channels;
}
size_t samples_added;
PreemptiveExpand::ReturnCodes return_code = preemptive_expand_->Process(
decoded_buffer, decoded_length,
old_borrowed_samples_per_channel,
algorithm_buffer_.get(), &samples_added);
stats_.PreemptiveExpandedSamples(samples_added);
switch (return_code) {
case PreemptiveExpand::kSuccess:
last_mode_ = kModePreemptiveExpandSuccess;
break;
case PreemptiveExpand::kSuccessLowEnergy:
last_mode_ = kModePreemptiveExpandLowEnergy;
break;
case PreemptiveExpand::kNoStretch:
last_mode_ = kModePreemptiveExpandFail;
break;
case PreemptiveExpand::kError:
// TODO(hlundin): Map to kModeError instead?
last_mode_ = kModePreemptiveExpandFail;
return kPreemptiveExpandError;
}
if (borrowed_samples_per_channel > 0) {
// Copy borrowed samples back to the |sync_buffer_|.
sync_buffer_->ReplaceAtIndex(
*algorithm_buffer_, borrowed_samples_per_channel,
sync_buffer_->Size() - borrowed_samples_per_channel);
algorithm_buffer_->PopFront(borrowed_samples_per_channel);
}
// If last packet was decoded as an inband CNG, set mode to CNG instead.
if (speech_type == AudioDecoder::kComfortNoise) {
last_mode_ = kModeCodecInternalCng;
}
if (!play_dtmf) {
dtmf_tone_generator_->Reset();
}
expand_->Reset();
return 0;
}
int NetEqImpl::DoRfc3389Cng(PacketList* packet_list, bool play_dtmf) {
if (!packet_list->empty()) {
// Must have exactly one SID frame at this point.
assert(packet_list->size() == 1);
Packet* packet = packet_list->front();
packet_list->pop_front();
if (!decoder_database_->IsComfortNoise(packet->header.payloadType)) {
#ifdef LEGACY_BITEXACT
// This can happen due to a bug in GetDecision. Change the payload type
// to a CNG type, and move on. Note that this means that we are in fact
// sending a non-CNG payload to the comfort noise decoder for decoding.
// Clearly wrong, but will maintain bit-exactness with legacy.
if (fs_hz_ == 8000) {
packet->header.payloadType =
decoder_database_->GetRtpPayloadType(kDecoderCNGnb);
} else if (fs_hz_ == 16000) {
packet->header.payloadType =
decoder_database_->GetRtpPayloadType(kDecoderCNGwb);
} else if (fs_hz_ == 32000) {
packet->header.payloadType =
decoder_database_->GetRtpPayloadType(kDecoderCNGswb32kHz);
} else if (fs_hz_ == 48000) {
packet->header.payloadType =
decoder_database_->GetRtpPayloadType(kDecoderCNGswb48kHz);
}
assert(decoder_database_->IsComfortNoise(packet->header.payloadType));
#else
LOG(LS_ERROR) << "Trying to decode non-CNG payload as CNG.";
return kOtherError;
#endif
}
// UpdateParameters() deletes |packet|.
if (comfort_noise_->UpdateParameters(packet) ==
ComfortNoise::kInternalError) {
algorithm_buffer_->Zeros(output_size_samples_);
return -comfort_noise_->internal_error_code();
}
}
int cn_return = comfort_noise_->Generate(output_size_samples_,
algorithm_buffer_.get());
expand_->Reset();
last_mode_ = kModeRfc3389Cng;
if (!play_dtmf) {
dtmf_tone_generator_->Reset();
}
if (cn_return == ComfortNoise::kInternalError) {
decoder_error_code_ = comfort_noise_->internal_error_code();
return kComfortNoiseErrorCode;
} else if (cn_return == ComfortNoise::kUnknownPayloadType) {
return kUnknownRtpPayloadType;
}
return 0;
}
void NetEqImpl::DoCodecInternalCng(const int16_t* decoded_buffer,
size_t decoded_length) {
RTC_DCHECK(normal_.get());
RTC_DCHECK(mute_factor_array_.get());
normal_->Process(decoded_buffer, decoded_length, last_mode_,
mute_factor_array_.get(), algorithm_buffer_.get());
last_mode_ = kModeCodecInternalCng;
expand_->Reset();
}
int NetEqImpl::DoDtmf(const DtmfEvent& dtmf_event, bool* play_dtmf) {
// This block of the code and the block further down, handling |dtmf_switch|
// are commented out. Otherwise playing out-of-band DTMF would fail in VoE
// test, DtmfTest.ManualSuccessfullySendsOutOfBandTelephoneEvents. This is
// equivalent to |dtmf_switch| always be false.
//
// See http://webrtc-codereview.appspot.com/1195004/ for discussion
// On this issue. This change might cause some glitches at the point of
// switch from audio to DTMF. Issue 1545 is filed to track this.
//
// bool dtmf_switch = false;
// if ((last_mode_ != kModeDtmf) && dtmf_tone_generator_->initialized()) {
// // Special case; see below.
// // We must catch this before calling Generate, since |initialized| is
// // modified in that call.
// dtmf_switch = true;
// }
int dtmf_return_value = 0;
if (!dtmf_tone_generator_->initialized()) {
// Initialize if not already done.
dtmf_return_value = dtmf_tone_generator_->Init(fs_hz_, dtmf_event.event_no,
dtmf_event.volume);
}
if (dtmf_return_value == 0) {
// Generate DTMF signal.
dtmf_return_value = dtmf_tone_generator_->Generate(output_size_samples_,
algorithm_buffer_.get());
}
if (dtmf_return_value < 0) {
algorithm_buffer_->Zeros(output_size_samples_);
return dtmf_return_value;
}
// if (dtmf_switch) {
// // This is the special case where the previous operation was DTMF
// // overdub, but the current instruction is "regular" DTMF. We must make
// // sure that the DTMF does not have any discontinuities. The first DTMF
// // sample that we generate now must be played out immediately, therefore
// // it must be copied to the speech buffer.
// // TODO(hlundin): This code seems incorrect. (Legacy.) Write test and
// // verify correct operation.
// assert(false);
// // Must generate enough data to replace all of the |sync_buffer_|
// // "future".
// int required_length = sync_buffer_->FutureLength();
// assert(dtmf_tone_generator_->initialized());
// dtmf_return_value = dtmf_tone_generator_->Generate(required_length,
// algorithm_buffer_);
// assert((size_t) required_length == algorithm_buffer_->Size());
// if (dtmf_return_value < 0) {
// algorithm_buffer_->Zeros(output_size_samples_);
// return dtmf_return_value;
// }
//
// // Overwrite the "future" part of the speech buffer with the new DTMF
// // data.
// // TODO(hlundin): It seems that this overwriting has gone lost.
// // Not adapted for multi-channel yet.
// assert(algorithm_buffer_->Channels() == 1);
// if (algorithm_buffer_->Channels() != 1) {
// LOG(LS_WARNING) << "DTMF not supported for more than one channel";
// return kStereoNotSupported;
// }
// // Shuffle the remaining data to the beginning of algorithm buffer.
// algorithm_buffer_->PopFront(sync_buffer_->FutureLength());
// }
sync_buffer_->IncreaseEndTimestamp(
static_cast<uint32_t>(output_size_samples_));
expand_->Reset();
last_mode_ = kModeDtmf;
// Set to false because the DTMF is already in the algorithm buffer.
*play_dtmf = false;
return 0;
}
void NetEqImpl::DoAlternativePlc(bool increase_timestamp) {
AudioDecoder* decoder = decoder_database_->GetActiveDecoder();
size_t length;
if (decoder && decoder->HasDecodePlc()) {
// Use the decoder's packet-loss concealment.
// TODO(hlundin): Will probably need a longer buffer for multi-channel.
int16_t decoded_buffer[kMaxFrameSize];
length = decoder->DecodePlc(1, decoded_buffer);
if (length > 0)
algorithm_buffer_->PushBackInterleaved(decoded_buffer, length);
} else {
// Do simple zero-stuffing.
length = output_size_samples_;
algorithm_buffer_->Zeros(length);
// By not advancing the timestamp, NetEq inserts samples.
stats_.AddZeros(length);
}
if (increase_timestamp) {
sync_buffer_->IncreaseEndTimestamp(static_cast<uint32_t>(length));
}
expand_->Reset();
}
int NetEqImpl::DtmfOverdub(const DtmfEvent& dtmf_event, size_t num_channels,
int16_t* output) const {
size_t out_index = 0;
size_t overdub_length = output_size_samples_; // Default value.
if (sync_buffer_->dtmf_index() > sync_buffer_->next_index()) {
// Special operation for transition from "DTMF only" to "DTMF overdub".
out_index = std::min(
sync_buffer_->dtmf_index() - sync_buffer_->next_index(),
output_size_samples_);
overdub_length = output_size_samples_ - out_index;
}
AudioMultiVector dtmf_output(num_channels);
int dtmf_return_value = 0;
if (!dtmf_tone_generator_->initialized()) {
dtmf_return_value = dtmf_tone_generator_->Init(fs_hz_, dtmf_event.event_no,
dtmf_event.volume);
}
if (dtmf_return_value == 0) {
dtmf_return_value = dtmf_tone_generator_->Generate(overdub_length,
&dtmf_output);
assert(overdub_length == dtmf_output.Size());
}
dtmf_output.ReadInterleaved(overdub_length, &output[out_index]);
return dtmf_return_value < 0 ? dtmf_return_value : 0;
}
int NetEqImpl::ExtractPackets(size_t required_samples,
PacketList* packet_list) {
bool first_packet = true;
uint8_t prev_payload_type = 0;
uint32_t prev_timestamp = 0;
uint16_t prev_sequence_number = 0;
bool next_packet_available = false;
const RTPHeader* header = packet_buffer_->NextRtpHeader();
assert(header);
if (!header) {
LOG(LS_ERROR) << "Packet buffer unexpectedly empty.";
return -1;
}
uint32_t first_timestamp = header->timestamp;
int extracted_samples = 0;
// Packet extraction loop.
do {
timestamp_ = header->timestamp;
size_t discard_count = 0;
Packet* packet = packet_buffer_->GetNextPacket(&discard_count);
// |header| may be invalid after the |packet_buffer_| operation.
header = NULL;
if (!packet) {
LOG(LS_ERROR) << "Should always be able to extract a packet here";
assert(false); // Should always be able to extract a packet here.
return -1;
}
stats_.PacketsDiscarded(discard_count);
// Store waiting time in ms; packets->waiting_time is in "output blocks".
stats_.StoreWaitingTime(packet->waiting_time * kOutputSizeMs);
assert(packet->payload_length > 0);
packet_list->push_back(packet); // Store packet in list.
if (first_packet) {
first_packet = false;
if (nack_enabled_) {
RTC_DCHECK(nack_);
// TODO(henrik.lundin): Should we update this for all decoded packets?
nack_->UpdateLastDecodedPacket(packet->header.sequenceNumber,
packet->header.timestamp);
}
prev_sequence_number = packet->header.sequenceNumber;
prev_timestamp = packet->header.timestamp;
prev_payload_type = packet->header.payloadType;
}
// Store number of extracted samples.
int packet_duration = 0;
AudioDecoder* decoder = decoder_database_->GetDecoder(
packet->header.payloadType);
if (decoder) {
if (packet->sync_packet) {
packet_duration = rtc::checked_cast<int>(decoder_frame_length_);
} else {
if (packet->primary) {
packet_duration = decoder->PacketDuration(packet->payload,
packet->payload_length);
} else {
packet_duration = decoder->
PacketDurationRedundant(packet->payload, packet->payload_length);
stats_.SecondaryDecodedSamples(packet_duration);
}
}
} else {
LOG(LS_WARNING) << "Unknown payload type "
<< static_cast<int>(packet->header.payloadType);
assert(false);
}
if (packet_duration <= 0) {
// Decoder did not return a packet duration. Assume that the packet
// contains the same number of samples as the previous one.
packet_duration = rtc::checked_cast<int>(decoder_frame_length_);
}
extracted_samples = packet->header.timestamp - first_timestamp +
packet_duration;
// Check what packet is available next.
header = packet_buffer_->NextRtpHeader();
next_packet_available = false;
if (header && prev_payload_type == header->payloadType) {
int16_t seq_no_diff = header->sequenceNumber - prev_sequence_number;
size_t ts_diff = header->timestamp - prev_timestamp;
if (seq_no_diff == 1 ||
(seq_no_diff == 0 && ts_diff == decoder_frame_length_)) {
// The next sequence number is available, or the next part of a packet
// that was split into pieces upon insertion.
next_packet_available = true;
}
prev_sequence_number = header->sequenceNumber;
}
} while (extracted_samples < rtc::checked_cast<int>(required_samples) &&
next_packet_available);
if (extracted_samples > 0) {
// Delete old packets only when we are going to decode something. Otherwise,
// we could end up in the situation where we never decode anything, since
// all incoming packets are considered too old but the buffer will also
// never be flooded and flushed.
packet_buffer_->DiscardAllOldPackets(timestamp_);
}
return extracted_samples;
}
void NetEqImpl::UpdatePlcComponents(int fs_hz, size_t channels) {
// Delete objects and create new ones.
expand_.reset(expand_factory_->Create(background_noise_.get(),
sync_buffer_.get(), &random_vector_,
&stats_, fs_hz, channels));
merge_.reset(new Merge(fs_hz, channels, expand_.get(), sync_buffer_.get()));
}
void NetEqImpl::SetSampleRateAndChannels(int fs_hz, size_t channels) {
LOG(LS_VERBOSE) << "SetSampleRateAndChannels " << fs_hz << " " << channels;
// TODO(hlundin): Change to an enumerator and skip assert.
assert(fs_hz == 8000 || fs_hz == 16000 || fs_hz == 32000 || fs_hz == 48000);
assert(channels > 0);
fs_hz_ = fs_hz;
fs_mult_ = fs_hz / 8000;
output_size_samples_ = static_cast<size_t>(kOutputSizeMs * 8 * fs_mult_);
decoder_frame_length_ = 3 * output_size_samples_; // Initialize to 30ms.
last_mode_ = kModeNormal;
// Create a new array of mute factors and set all to 1.
mute_factor_array_.reset(new int16_t[channels]);
for (size_t i = 0; i < channels; ++i) {
mute_factor_array_[i] = 16384; // 1.0 in Q14.
}
AudioDecoder* cng_decoder = decoder_database_->GetActiveCngDecoder();
if (cng_decoder)
cng_decoder->Reset();
// Reinit post-decode VAD with new sample rate.
assert(vad_.get()); // Cannot be NULL here.
vad_->Init();
// Delete algorithm buffer and create a new one.
algorithm_buffer_.reset(new AudioMultiVector(channels));
// Delete sync buffer and create a new one.
sync_buffer_.reset(new SyncBuffer(channels, kSyncBufferSize * fs_mult_));
// Delete BackgroundNoise object and create a new one.
background_noise_.reset(new BackgroundNoise(channels));
background_noise_->set_mode(background_noise_mode_);
// Reset random vector.
random_vector_.Reset();
UpdatePlcComponents(fs_hz, channels);
// Move index so that we create a small set of future samples (all 0).
sync_buffer_->set_next_index(sync_buffer_->next_index() -
expand_->overlap_length());
normal_.reset(new Normal(fs_hz, decoder_database_.get(), *background_noise_,
expand_.get()));
accelerate_.reset(
accelerate_factory_->Create(fs_hz, channels, *background_noise_));
preemptive_expand_.reset(preemptive_expand_factory_->Create(
fs_hz, channels, *background_noise_, expand_->overlap_length()));
// Delete ComfortNoise object and create a new one.
comfort_noise_.reset(new ComfortNoise(fs_hz, decoder_database_.get(),
sync_buffer_.get()));
// Verify that |decoded_buffer_| is long enough.
if (decoded_buffer_length_ < kMaxFrameSize * channels) {
// Reallocate to larger size.
decoded_buffer_length_ = kMaxFrameSize * channels;
decoded_buffer_.reset(new int16_t[decoded_buffer_length_]);
}
// Create DecisionLogic if it is not created yet, then communicate new sample
// rate and output size to DecisionLogic object.
if (!decision_logic_.get()) {
CreateDecisionLogic();
}
decision_logic_->SetSampleRate(fs_hz_, output_size_samples_);
}
NetEqOutputType NetEqImpl::LastOutputType() {
assert(vad_.get());
assert(expand_.get());
if (last_mode_ == kModeCodecInternalCng || last_mode_ == kModeRfc3389Cng) {
return kOutputCNG;
} else if (last_mode_ == kModeExpand && expand_->MuteFactor(0) == 0) {
// Expand mode has faded down to background noise only (very long expand).
return kOutputPLCtoCNG;
} else if (last_mode_ == kModeExpand) {
return kOutputPLC;
} else if (vad_->running() && !vad_->active_speech()) {
return kOutputVADPassive;
} else {
return kOutputNormal;
}
}
void NetEqImpl::CreateDecisionLogic() {
decision_logic_.reset(DecisionLogic::Create(fs_hz_, output_size_samples_,
playout_mode_,
decoder_database_.get(),
*packet_buffer_.get(),
delay_manager_.get(),
buffer_level_filter_.get()));
}
} // namespace webrtc