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webrtc/modules/audio_coding/acm2/audio_coding_module.cc
Oskar Sundbom 12ab00b4d8 Optional: Use nullopt and implicit construction in /modules/audio_coding 
Changes places where we explicitly construct an Optional to instead use
nullopt or the requisite value type only.

This CL was uploaded by git cl split.

R=kwiberg@webrtc.org

Bug: None
Change-Id: I055411a3e521964c81100869a197dd92f5608f1b
Reviewed-on: https://webrtc-review.googlesource.com/23619
Commit-Queue: Oskar Sundbom <ossu@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Elad Alon <eladalon@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20728}
2017-11-17 11:58:37 +00:00

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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 "modules/audio_coding/include/audio_coding_module.h"
#include <algorithm>
#include "api/audio_codecs/builtin_audio_decoder_factory.h"
#include "modules/audio_coding/acm2/acm_receiver.h"
#include "modules/audio_coding/acm2/acm_resampler.h"
#include "modules/audio_coding/acm2/codec_manager.h"
#include "modules/audio_coding/acm2/rent_a_codec.h"
#include "rtc_base/checks.h"
#include "rtc_base/logging.h"
#include "rtc_base/safe_conversions.h"
#include "system_wrappers/include/metrics.h"
namespace webrtc {
namespace {
struct EncoderFactory {
AudioEncoder* external_speech_encoder = nullptr;
acm2::CodecManager codec_manager;
acm2::RentACodec rent_a_codec;
};
class AudioCodingModuleImpl final : public AudioCodingModule {
public:
explicit AudioCodingModuleImpl(const AudioCodingModule::Config& config);
~AudioCodingModuleImpl() override;
/////////////////////////////////////////
// Sender
//
// Can be called multiple times for Codec, CNG, RED.
int RegisterSendCodec(const CodecInst& send_codec) override;
void RegisterExternalSendCodec(
AudioEncoder* external_speech_encoder) override;
void ModifyEncoder(rtc::FunctionView<void(std::unique_ptr<AudioEncoder>*)>
modifier) override;
void QueryEncoder(
rtc::FunctionView<void(const AudioEncoder*)> query) override;
// Get current send codec.
rtc::Optional<CodecInst> SendCodec() const override;
// Get current send frequency.
int SendFrequency() const override;
// Sets the bitrate to the specified value in bits/sec. In case the codec does
// not support the requested value it will choose an appropriate value
// instead.
void SetBitRate(int bitrate_bps) override;
// Register a transport callback which will be
// called to deliver the encoded buffers.
int RegisterTransportCallback(AudioPacketizationCallback* transport) override;
// Add 10 ms of raw (PCM) audio data to the encoder.
int Add10MsData(const AudioFrame& audio_frame) override;
/////////////////////////////////////////
// (RED) Redundant Coding
//
// Configure RED status i.e. on/off.
int SetREDStatus(bool enable_red) override;
// Get RED status.
bool REDStatus() const override;
/////////////////////////////////////////
// (FEC) Forward Error Correction (codec internal)
//
// Configure FEC status i.e. on/off.
int SetCodecFEC(bool enabled_codec_fec) override;
// Get FEC status.
bool CodecFEC() const override;
// Set target packet loss rate
int SetPacketLossRate(int loss_rate) override;
/////////////////////////////////////////
// (VAD) Voice Activity Detection
// and
// (CNG) Comfort Noise Generation
//
int SetVAD(bool enable_dtx = true,
bool enable_vad = false,
ACMVADMode mode = VADNormal) override;
int VAD(bool* dtx_enabled,
bool* vad_enabled,
ACMVADMode* mode) const override;
int RegisterVADCallback(ACMVADCallback* vad_callback) override;
/////////////////////////////////////////
// Receiver
//
// Initialize receiver, resets codec database etc.
int InitializeReceiver() override;
// Get current receive frequency.
int ReceiveFrequency() const override;
// Get current playout frequency.
int PlayoutFrequency() const override;
void SetReceiveCodecs(const std::map<int, SdpAudioFormat>& codecs) override;
bool RegisterReceiveCodec(int rtp_payload_type,
const SdpAudioFormat& audio_format) override;
int RegisterReceiveCodec(const CodecInst& receive_codec) override;
int RegisterReceiveCodec(
const CodecInst& receive_codec,
rtc::FunctionView<std::unique_ptr<AudioDecoder>()> isac_factory) override;
int RegisterExternalReceiveCodec(int rtp_payload_type,
AudioDecoder* external_decoder,
int sample_rate_hz,
int num_channels,
const std::string& name) override;
// Get current received codec.
int ReceiveCodec(CodecInst* current_codec) const override;
rtc::Optional<SdpAudioFormat> ReceiveFormat() const override;
// Incoming packet from network parsed and ready for decode.
int IncomingPacket(const uint8_t* incoming_payload,
const size_t payload_length,
const WebRtcRTPHeader& rtp_info) override;
// Minimum playout delay.
int SetMinimumPlayoutDelay(int time_ms) override;
// Maximum playout delay.
int SetMaximumPlayoutDelay(int time_ms) override;
// Smallest latency NetEq will maintain.
int LeastRequiredDelayMs() const override;
RTC_DEPRECATED int32_t PlayoutTimestamp(uint32_t* timestamp) override;
rtc::Optional<uint32_t> PlayoutTimestamp() override;
int FilteredCurrentDelayMs() const override;
// Get 10 milliseconds of raw audio data to play out, and
// automatic resample to the requested frequency if > 0.
int PlayoutData10Ms(int desired_freq_hz,
AudioFrame* audio_frame,
bool* muted) override;
int PlayoutData10Ms(int desired_freq_hz, AudioFrame* audio_frame) override;
/////////////////////////////////////////
// Statistics
//
int GetNetworkStatistics(NetworkStatistics* statistics) override;
int SetOpusApplication(OpusApplicationMode application) override;
// If current send codec is Opus, informs it about the maximum playback rate
// the receiver will render.
int SetOpusMaxPlaybackRate(int frequency_hz) override;
int EnableOpusDtx() override;
int DisableOpusDtx() override;
int UnregisterReceiveCodec(uint8_t payload_type) override;
int EnableNack(size_t max_nack_list_size) override;
void DisableNack() override;
std::vector<uint16_t> GetNackList(int64_t round_trip_time_ms) const override;
void GetDecodingCallStatistics(AudioDecodingCallStats* stats) const override;
ANAStats GetANAStats() const override;
private:
struct InputData {
uint32_t input_timestamp;
const int16_t* audio;
size_t length_per_channel;
size_t audio_channel;
// If a re-mix is required (up or down), this buffer will store a re-mixed
// version of the input.
int16_t buffer[WEBRTC_10MS_PCM_AUDIO];
};
// This member class writes values to the named UMA histogram, but only if
// the value has changed since the last time (and always for the first call).
class ChangeLogger {
public:
explicit ChangeLogger(const std::string& histogram_name)
: histogram_name_(histogram_name) {}
// Logs the new value if it is different from the last logged value, or if
// this is the first call.
void MaybeLog(int value);
private:
int last_value_ = 0;
int first_time_ = true;
const std::string histogram_name_;
};
int RegisterReceiveCodecUnlocked(
const CodecInst& codec,
rtc::FunctionView<std::unique_ptr<AudioDecoder>()> isac_factory)
RTC_EXCLUSIVE_LOCKS_REQUIRED(acm_crit_sect_);
int Add10MsDataInternal(const AudioFrame& audio_frame, InputData* input_data)
RTC_EXCLUSIVE_LOCKS_REQUIRED(acm_crit_sect_);
int Encode(const InputData& input_data)
RTC_EXCLUSIVE_LOCKS_REQUIRED(acm_crit_sect_);
int InitializeReceiverSafe() RTC_EXCLUSIVE_LOCKS_REQUIRED(acm_crit_sect_);
bool HaveValidEncoder(const char* caller_name) const
RTC_EXCLUSIVE_LOCKS_REQUIRED(acm_crit_sect_);
// Preprocessing of input audio, including resampling and down-mixing if
// required, before pushing audio into encoder's buffer.
//
// in_frame: input audio-frame
// ptr_out: pointer to output audio_frame. If no preprocessing is required
// |ptr_out| will be pointing to |in_frame|, otherwise pointing to
// |preprocess_frame_|.
//
// Return value:
// -1: if encountering an error.
// 0: otherwise.
int PreprocessToAddData(const AudioFrame& in_frame,
const AudioFrame** ptr_out)
RTC_EXCLUSIVE_LOCKS_REQUIRED(acm_crit_sect_);
// Change required states after starting to receive the codec corresponding
// to |index|.
int UpdateUponReceivingCodec(int index);
rtc::CriticalSection acm_crit_sect_;
rtc::Buffer encode_buffer_ RTC_GUARDED_BY(acm_crit_sect_);
uint32_t expected_codec_ts_ RTC_GUARDED_BY(acm_crit_sect_);
uint32_t expected_in_ts_ RTC_GUARDED_BY(acm_crit_sect_);
acm2::ACMResampler resampler_ RTC_GUARDED_BY(acm_crit_sect_);
acm2::AcmReceiver receiver_; // AcmReceiver has it's own internal lock.
ChangeLogger bitrate_logger_ RTC_GUARDED_BY(acm_crit_sect_);
std::unique_ptr<EncoderFactory> encoder_factory_
RTC_GUARDED_BY(acm_crit_sect_);
// Current encoder stack, either obtained from
// encoder_factory_->rent_a_codec.RentEncoderStack or provided by a call to
// RegisterEncoder.
std::unique_ptr<AudioEncoder> encoder_stack_ RTC_GUARDED_BY(acm_crit_sect_);
std::unique_ptr<AudioDecoder> isac_decoder_16k_
RTC_GUARDED_BY(acm_crit_sect_);
std::unique_ptr<AudioDecoder> isac_decoder_32k_
RTC_GUARDED_BY(acm_crit_sect_);
// This is to keep track of CN instances where we can send DTMFs.
uint8_t previous_pltype_ RTC_GUARDED_BY(acm_crit_sect_);
bool receiver_initialized_ RTC_GUARDED_BY(acm_crit_sect_);
AudioFrame preprocess_frame_ RTC_GUARDED_BY(acm_crit_sect_);
bool first_10ms_data_ RTC_GUARDED_BY(acm_crit_sect_);
bool first_frame_ RTC_GUARDED_BY(acm_crit_sect_);
uint32_t last_timestamp_ RTC_GUARDED_BY(acm_crit_sect_);
uint32_t last_rtp_timestamp_ RTC_GUARDED_BY(acm_crit_sect_);
rtc::CriticalSection callback_crit_sect_;
AudioPacketizationCallback* packetization_callback_
RTC_GUARDED_BY(callback_crit_sect_);
ACMVADCallback* vad_callback_ RTC_GUARDED_BY(callback_crit_sect_);
int codec_histogram_bins_log_[static_cast<size_t>(
AudioEncoder::CodecType::kMaxLoggedAudioCodecTypes)];
int number_of_consecutive_empty_packets_;
};
// Adds a codec usage sample to the histogram.
void UpdateCodecTypeHistogram(size_t codec_type) {
RTC_HISTOGRAM_ENUMERATION(
"WebRTC.Audio.Encoder.CodecType", static_cast<int>(codec_type),
static_cast<int>(
webrtc::AudioEncoder::CodecType::kMaxLoggedAudioCodecTypes));
}
// Stereo-to-mono can be used as in-place.
int DownMix(const AudioFrame& frame,
size_t length_out_buff,
int16_t* out_buff) {
RTC_DCHECK_EQ(frame.num_channels_, 2);
RTC_DCHECK_GE(length_out_buff, frame.samples_per_channel_);
if (!frame.muted()) {
const int16_t* frame_data = frame.data();
for (size_t n = 0; n < frame.samples_per_channel_; ++n) {
out_buff[n] = static_cast<int16_t>(
(static_cast<int32_t>(frame_data[2 * n]) +
static_cast<int32_t>(frame_data[2 * n + 1])) >> 1);
}
} else {
std::fill(out_buff, out_buff + frame.samples_per_channel_, 0);
}
return 0;
}
// Mono-to-stereo can be used as in-place.
int UpMix(const AudioFrame& frame, size_t length_out_buff, int16_t* out_buff) {
RTC_DCHECK_EQ(frame.num_channels_, 1);
RTC_DCHECK_GE(length_out_buff, 2 * frame.samples_per_channel_);
if (!frame.muted()) {
const int16_t* frame_data = frame.data();
for (size_t n = frame.samples_per_channel_; n != 0; --n) {
size_t i = n - 1;
int16_t sample = frame_data[i];
out_buff[2 * i + 1] = sample;
out_buff[2 * i] = sample;
}
} else {
std::fill(out_buff, out_buff + frame.samples_per_channel_ * 2, 0);
}
return 0;
}
void ConvertEncodedInfoToFragmentationHeader(
const AudioEncoder::EncodedInfo& info,
RTPFragmentationHeader* frag) {
if (info.redundant.empty()) {
frag->fragmentationVectorSize = 0;
return;
}
frag->VerifyAndAllocateFragmentationHeader(
static_cast<uint16_t>(info.redundant.size()));
frag->fragmentationVectorSize = static_cast<uint16_t>(info.redundant.size());
size_t offset = 0;
for (size_t i = 0; i < info.redundant.size(); ++i) {
frag->fragmentationOffset[i] = offset;
offset += info.redundant[i].encoded_bytes;
frag->fragmentationLength[i] = info.redundant[i].encoded_bytes;
frag->fragmentationTimeDiff[i] = rtc::dchecked_cast<uint16_t>(
info.encoded_timestamp - info.redundant[i].encoded_timestamp);
frag->fragmentationPlType[i] = info.redundant[i].payload_type;
}
}
// Wraps a raw AudioEncoder pointer. The idea is that you can put one of these
// in a unique_ptr, to protect the contained raw pointer from being deleted
// when the unique_ptr expires. (This is of course a bad idea in general, but
// backwards compatibility.)
class RawAudioEncoderWrapper final : public AudioEncoder {
public:
RawAudioEncoderWrapper(AudioEncoder* enc) : enc_(enc) {}
int SampleRateHz() const override { return enc_->SampleRateHz(); }
size_t NumChannels() const override { return enc_->NumChannels(); }
int RtpTimestampRateHz() const override { return enc_->RtpTimestampRateHz(); }
size_t Num10MsFramesInNextPacket() const override {
return enc_->Num10MsFramesInNextPacket();
}
size_t Max10MsFramesInAPacket() const override {
return enc_->Max10MsFramesInAPacket();
}
int GetTargetBitrate() const override { return enc_->GetTargetBitrate(); }
EncodedInfo EncodeImpl(uint32_t rtp_timestamp,
rtc::ArrayView<const int16_t> audio,
rtc::Buffer* encoded) override {
return enc_->Encode(rtp_timestamp, audio, encoded);
}
void Reset() override { return enc_->Reset(); }
bool SetFec(bool enable) override { return enc_->SetFec(enable); }
bool SetDtx(bool enable) override { return enc_->SetDtx(enable); }
bool SetApplication(Application application) override {
return enc_->SetApplication(application);
}
void SetMaxPlaybackRate(int frequency_hz) override {
return enc_->SetMaxPlaybackRate(frequency_hz);
}
private:
AudioEncoder* enc_;
};
// Return false on error.
bool CreateSpeechEncoderIfNecessary(EncoderFactory* ef) {
auto* sp = ef->codec_manager.GetStackParams();
if (sp->speech_encoder) {
// Do nothing; we already have a speech encoder.
} else if (ef->codec_manager.GetCodecInst()) {
RTC_DCHECK(!ef->external_speech_encoder);
// We have no speech encoder, but we have a specification for making one.
std::unique_ptr<AudioEncoder> enc =
ef->rent_a_codec.RentEncoder(*ef->codec_manager.GetCodecInst());
if (!enc)
return false; // Encoder spec was bad.
sp->speech_encoder = std::move(enc);
} else if (ef->external_speech_encoder) {
RTC_DCHECK(!ef->codec_manager.GetCodecInst());
// We have an external speech encoder.
sp->speech_encoder = std::unique_ptr<AudioEncoder>(
new RawAudioEncoderWrapper(ef->external_speech_encoder));
}
return true;
}
void AudioCodingModuleImpl::ChangeLogger::MaybeLog(int value) {
if (value != last_value_ || first_time_) {
first_time_ = false;
last_value_ = value;
RTC_HISTOGRAM_COUNTS_SPARSE_100(histogram_name_, value);
}
}
AudioCodingModuleImpl::AudioCodingModuleImpl(
const AudioCodingModule::Config& config)
: expected_codec_ts_(0xD87F3F9F),
expected_in_ts_(0xD87F3F9F),
receiver_(config),
bitrate_logger_("WebRTC.Audio.TargetBitrateInKbps"),
encoder_factory_(new EncoderFactory),
encoder_stack_(nullptr),
previous_pltype_(255),
receiver_initialized_(false),
first_10ms_data_(false),
first_frame_(true),
packetization_callback_(NULL),
vad_callback_(NULL),
codec_histogram_bins_log_(),
number_of_consecutive_empty_packets_(0) {
if (InitializeReceiverSafe() < 0) {
RTC_LOG(LS_ERROR) << "Cannot initialize receiver";
}
RTC_LOG(LS_INFO) << "Created";
}
AudioCodingModuleImpl::~AudioCodingModuleImpl() = default;
int32_t AudioCodingModuleImpl::Encode(const InputData& input_data) {
AudioEncoder::EncodedInfo encoded_info;
uint8_t previous_pltype;
// Check if there is an encoder before.
if (!HaveValidEncoder("Process"))
return -1;
if(!first_frame_) {
RTC_DCHECK(IsNewerTimestamp(input_data.input_timestamp, last_timestamp_))
<< "Time should not move backwards";
}
// Scale the timestamp to the codec's RTP timestamp rate.
uint32_t rtp_timestamp =
first_frame_ ? input_data.input_timestamp
: last_rtp_timestamp_ +
rtc::CheckedDivExact(
input_data.input_timestamp - last_timestamp_,
static_cast<uint32_t>(rtc::CheckedDivExact(
encoder_stack_->SampleRateHz(),
encoder_stack_->RtpTimestampRateHz())));
last_timestamp_ = input_data.input_timestamp;
last_rtp_timestamp_ = rtp_timestamp;
first_frame_ = false;
// Clear the buffer before reuse - encoded data will get appended.
encode_buffer_.Clear();
encoded_info = encoder_stack_->Encode(
rtp_timestamp, rtc::ArrayView<const int16_t>(
input_data.audio, input_data.audio_channel *
input_data.length_per_channel),
&encode_buffer_);
bitrate_logger_.MaybeLog(encoder_stack_->GetTargetBitrate() / 1000);
if (encode_buffer_.size() == 0 && !encoded_info.send_even_if_empty) {
// Not enough data.
return 0;
}
previous_pltype = previous_pltype_; // Read it while we have the critsect.
// Log codec type to histogram once every 500 packets.
if (encoded_info.encoded_bytes == 0) {
++number_of_consecutive_empty_packets_;
} else {
size_t codec_type = static_cast<size_t>(encoded_info.encoder_type);
codec_histogram_bins_log_[codec_type] +=
number_of_consecutive_empty_packets_ + 1;
number_of_consecutive_empty_packets_ = 0;
if (codec_histogram_bins_log_[codec_type] >= 500) {
codec_histogram_bins_log_[codec_type] -= 500;
UpdateCodecTypeHistogram(codec_type);
}
}
RTPFragmentationHeader my_fragmentation;
ConvertEncodedInfoToFragmentationHeader(encoded_info, &my_fragmentation);
FrameType frame_type;
if (encode_buffer_.size() == 0 && encoded_info.send_even_if_empty) {
frame_type = kEmptyFrame;
encoded_info.payload_type = previous_pltype;
} else {
RTC_DCHECK_GT(encode_buffer_.size(), 0);
frame_type = encoded_info.speech ? kAudioFrameSpeech : kAudioFrameCN;
}
{
rtc::CritScope lock(&callback_crit_sect_);
if (packetization_callback_) {
packetization_callback_->SendData(
frame_type, encoded_info.payload_type, encoded_info.encoded_timestamp,
encode_buffer_.data(), encode_buffer_.size(),
my_fragmentation.fragmentationVectorSize > 0 ? &my_fragmentation
: nullptr);
}
if (vad_callback_) {
// Callback with VAD decision.
vad_callback_->InFrameType(frame_type);
}
}
previous_pltype_ = encoded_info.payload_type;
return static_cast<int32_t>(encode_buffer_.size());
}
/////////////////////////////////////////
// Sender
//
// Can be called multiple times for Codec, CNG, RED.
int AudioCodingModuleImpl::RegisterSendCodec(const CodecInst& send_codec) {
rtc::CritScope lock(&acm_crit_sect_);
if (!encoder_factory_->codec_manager.RegisterEncoder(send_codec)) {
return -1;
}
if (encoder_factory_->codec_manager.GetCodecInst()) {
encoder_factory_->external_speech_encoder = nullptr;
}
if (!CreateSpeechEncoderIfNecessary(encoder_factory_.get())) {
return -1;
}
auto* sp = encoder_factory_->codec_manager.GetStackParams();
if (sp->speech_encoder)
encoder_stack_ = encoder_factory_->rent_a_codec.RentEncoderStack(sp);
return 0;
}
void AudioCodingModuleImpl::RegisterExternalSendCodec(
AudioEncoder* external_speech_encoder) {
rtc::CritScope lock(&acm_crit_sect_);
encoder_factory_->codec_manager.UnsetCodecInst();
encoder_factory_->external_speech_encoder = external_speech_encoder;
RTC_CHECK(CreateSpeechEncoderIfNecessary(encoder_factory_.get()));
auto* sp = encoder_factory_->codec_manager.GetStackParams();
RTC_CHECK(sp->speech_encoder);
encoder_stack_ = encoder_factory_->rent_a_codec.RentEncoderStack(sp);
}
void AudioCodingModuleImpl::ModifyEncoder(
rtc::FunctionView<void(std::unique_ptr<AudioEncoder>*)> modifier) {
rtc::CritScope lock(&acm_crit_sect_);
// Wipe the encoder factory, so that everything that relies on it will fail.
// We don't want the complexity of supporting swapping back and forth.
if (encoder_factory_) {
encoder_factory_.reset();
RTC_CHECK(!encoder_stack_); // Ensure we hadn't started using the factory.
}
modifier(&encoder_stack_);
}
void AudioCodingModuleImpl::QueryEncoder(
rtc::FunctionView<void(const AudioEncoder*)> query) {
rtc::CritScope lock(&acm_crit_sect_);
query(encoder_stack_.get());
}
// Get current send codec.
rtc::Optional<CodecInst> AudioCodingModuleImpl::SendCodec() const {
rtc::CritScope lock(&acm_crit_sect_);
if (encoder_factory_) {
auto* ci = encoder_factory_->codec_manager.GetCodecInst();
if (ci) {
return *ci;
}
CreateSpeechEncoderIfNecessary(encoder_factory_.get());
const std::unique_ptr<AudioEncoder>& enc =
encoder_factory_->codec_manager.GetStackParams()->speech_encoder;
if (enc) {
return acm2::CodecManager::ForgeCodecInst(enc.get());
}
return rtc::nullopt;
} else {
return encoder_stack_
? rtc::Optional<CodecInst>(
acm2::CodecManager::ForgeCodecInst(encoder_stack_.get()))
: rtc::nullopt;
}
}
// Get current send frequency.
int AudioCodingModuleImpl::SendFrequency() const {
rtc::CritScope lock(&acm_crit_sect_);
if (!encoder_stack_) {
RTC_LOG(LS_ERROR) << "SendFrequency Failed, no codec is registered";
return -1;
}
return encoder_stack_->SampleRateHz();
}
void AudioCodingModuleImpl::SetBitRate(int bitrate_bps) {
rtc::CritScope lock(&acm_crit_sect_);
if (encoder_stack_) {
encoder_stack_->OnReceivedUplinkBandwidth(bitrate_bps, rtc::nullopt);
}
}
// Register a transport callback which will be called to deliver
// the encoded buffers.
int AudioCodingModuleImpl::RegisterTransportCallback(
AudioPacketizationCallback* transport) {
rtc::CritScope lock(&callback_crit_sect_);
packetization_callback_ = transport;
return 0;
}
// Add 10MS of raw (PCM) audio data to the encoder.
int AudioCodingModuleImpl::Add10MsData(const AudioFrame& audio_frame) {
InputData input_data;
rtc::CritScope lock(&acm_crit_sect_);
int r = Add10MsDataInternal(audio_frame, &input_data);
return r < 0 ? r : Encode(input_data);
}
int AudioCodingModuleImpl::Add10MsDataInternal(const AudioFrame& audio_frame,
InputData* input_data) {
if (audio_frame.samples_per_channel_ == 0) {
assert(false);
RTC_LOG(LS_ERROR) << "Cannot Add 10 ms audio, payload length is zero";
return -1;
}
if (audio_frame.sample_rate_hz_ > 48000) {
assert(false);
RTC_LOG(LS_ERROR) << "Cannot Add 10 ms audio, input frequency not valid";
return -1;
}
// If the length and frequency matches. We currently just support raw PCM.
if (static_cast<size_t>(audio_frame.sample_rate_hz_ / 100) !=
audio_frame.samples_per_channel_) {
RTC_LOG(LS_ERROR)
<< "Cannot Add 10 ms audio, input frequency and length doesn't match";
return -1;
}
if (audio_frame.num_channels_ != 1 && audio_frame.num_channels_ != 2) {
RTC_LOG(LS_ERROR) << "Cannot Add 10 ms audio, invalid number of channels.";
return -1;
}
// Do we have a codec registered?
if (!HaveValidEncoder("Add10MsData")) {
return -1;
}
const AudioFrame* ptr_frame;
// Perform a resampling, also down-mix if it is required and can be
// performed before resampling (a down mix prior to resampling will take
// place if both primary and secondary encoders are mono and input is in
// stereo).
if (PreprocessToAddData(audio_frame, &ptr_frame) < 0) {
return -1;
}
// Check whether we need an up-mix or down-mix?
const size_t current_num_channels = encoder_stack_->NumChannels();
const bool same_num_channels =
ptr_frame->num_channels_ == current_num_channels;
if (!same_num_channels) {
if (ptr_frame->num_channels_ == 1) {
if (UpMix(*ptr_frame, WEBRTC_10MS_PCM_AUDIO, input_data->buffer) < 0)
return -1;
} else {
if (DownMix(*ptr_frame, WEBRTC_10MS_PCM_AUDIO, input_data->buffer) < 0)
return -1;
}
}
// When adding data to encoders this pointer is pointing to an audio buffer
// with correct number of channels.
const int16_t* ptr_audio = ptr_frame->data();
// For pushing data to primary, point the |ptr_audio| to correct buffer.
if (!same_num_channels)
ptr_audio = input_data->buffer;
// TODO(yujo): Skip encode of muted frames.
input_data->input_timestamp = ptr_frame->timestamp_;
input_data->audio = ptr_audio;
input_data->length_per_channel = ptr_frame->samples_per_channel_;
input_data->audio_channel = current_num_channels;
return 0;
}
// Perform a resampling and down-mix if required. We down-mix only if
// encoder is mono and input is stereo. In case of dual-streaming, both
// encoders has to be mono for down-mix to take place.
// |*ptr_out| will point to the pre-processed audio-frame. If no pre-processing
// is required, |*ptr_out| points to |in_frame|.
// TODO(yujo): Make this more efficient for muted frames.
int AudioCodingModuleImpl::PreprocessToAddData(const AudioFrame& in_frame,
const AudioFrame** ptr_out) {
const bool resample =
in_frame.sample_rate_hz_ != encoder_stack_->SampleRateHz();
// This variable is true if primary codec and secondary codec (if exists)
// are both mono and input is stereo.
// TODO(henrik.lundin): This condition should probably be
// in_frame.num_channels_ > encoder_stack_->NumChannels()
const bool down_mix =
in_frame.num_channels_ == 2 && encoder_stack_->NumChannels() == 1;
if (!first_10ms_data_) {
expected_in_ts_ = in_frame.timestamp_;
expected_codec_ts_ = in_frame.timestamp_;
first_10ms_data_ = true;
} else if (in_frame.timestamp_ != expected_in_ts_) {
RTC_LOG(LS_WARNING) << "Unexpected input timestamp: " << in_frame.timestamp_
<< ", expected: " << expected_in_ts_;
expected_codec_ts_ +=
(in_frame.timestamp_ - expected_in_ts_) *
static_cast<uint32_t>(
static_cast<double>(encoder_stack_->SampleRateHz()) /
static_cast<double>(in_frame.sample_rate_hz_));
expected_in_ts_ = in_frame.timestamp_;
}
if (!down_mix && !resample) {
// No pre-processing is required.
if (expected_in_ts_ == expected_codec_ts_) {
// If we've never resampled, we can use the input frame as-is
*ptr_out = &in_frame;
} else {
// Otherwise we'll need to alter the timestamp. Since in_frame is const,
// we'll have to make a copy of it.
preprocess_frame_.CopyFrom(in_frame);
preprocess_frame_.timestamp_ = expected_codec_ts_;
*ptr_out = &preprocess_frame_;
}
expected_in_ts_ += static_cast<uint32_t>(in_frame.samples_per_channel_);
expected_codec_ts_ += static_cast<uint32_t>(in_frame.samples_per_channel_);
return 0;
}
*ptr_out = &preprocess_frame_;
preprocess_frame_.num_channels_ = in_frame.num_channels_;
int16_t audio[WEBRTC_10MS_PCM_AUDIO];
const int16_t* src_ptr_audio = in_frame.data();
if (down_mix) {
// If a resampling is required the output of a down-mix is written into a
// local buffer, otherwise, it will be written to the output frame.
int16_t* dest_ptr_audio = resample ?
audio : preprocess_frame_.mutable_data();
if (DownMix(in_frame, WEBRTC_10MS_PCM_AUDIO, dest_ptr_audio) < 0)
return -1;
preprocess_frame_.num_channels_ = 1;
// Set the input of the resampler is the down-mixed signal.
src_ptr_audio = audio;
}
preprocess_frame_.timestamp_ = expected_codec_ts_;
preprocess_frame_.samples_per_channel_ = in_frame.samples_per_channel_;
preprocess_frame_.sample_rate_hz_ = in_frame.sample_rate_hz_;
// If it is required, we have to do a resampling.
if (resample) {
// The result of the resampler is written to output frame.
int16_t* dest_ptr_audio = preprocess_frame_.mutable_data();
int samples_per_channel = resampler_.Resample10Msec(
src_ptr_audio, in_frame.sample_rate_hz_, encoder_stack_->SampleRateHz(),
preprocess_frame_.num_channels_, AudioFrame::kMaxDataSizeSamples,
dest_ptr_audio);
if (samples_per_channel < 0) {
RTC_LOG(LS_ERROR) << "Cannot add 10 ms audio, resampling failed";
return -1;
}
preprocess_frame_.samples_per_channel_ =
static_cast<size_t>(samples_per_channel);
preprocess_frame_.sample_rate_hz_ = encoder_stack_->SampleRateHz();
}
expected_codec_ts_ +=
static_cast<uint32_t>(preprocess_frame_.samples_per_channel_);
expected_in_ts_ += static_cast<uint32_t>(in_frame.samples_per_channel_);
return 0;
}
/////////////////////////////////////////
// (RED) Redundant Coding
//
bool AudioCodingModuleImpl::REDStatus() const {
rtc::CritScope lock(&acm_crit_sect_);
return encoder_factory_->codec_manager.GetStackParams()->use_red;
}
// Configure RED status i.e on/off.
int AudioCodingModuleImpl::SetREDStatus(bool enable_red) {
#ifdef WEBRTC_CODEC_RED
rtc::CritScope lock(&acm_crit_sect_);
CreateSpeechEncoderIfNecessary(encoder_factory_.get());
if (!encoder_factory_->codec_manager.SetCopyRed(enable_red)) {
return -1;
}
auto* sp = encoder_factory_->codec_manager.GetStackParams();
if (sp->speech_encoder)
encoder_stack_ = encoder_factory_->rent_a_codec.RentEncoderStack(sp);
return 0;
#else
RTC_LOG(LS_WARNING) << " WEBRTC_CODEC_RED is undefined";
return -1;
#endif
}
/////////////////////////////////////////
// (FEC) Forward Error Correction (codec internal)
//
bool AudioCodingModuleImpl::CodecFEC() const {
rtc::CritScope lock(&acm_crit_sect_);
return encoder_factory_->codec_manager.GetStackParams()->use_codec_fec;
}
int AudioCodingModuleImpl::SetCodecFEC(bool enable_codec_fec) {
rtc::CritScope lock(&acm_crit_sect_);
CreateSpeechEncoderIfNecessary(encoder_factory_.get());
if (!encoder_factory_->codec_manager.SetCodecFEC(enable_codec_fec)) {
return -1;
}
auto* sp = encoder_factory_->codec_manager.GetStackParams();
if (sp->speech_encoder)
encoder_stack_ = encoder_factory_->rent_a_codec.RentEncoderStack(sp);
if (enable_codec_fec) {
return sp->use_codec_fec ? 0 : -1;
} else {
RTC_DCHECK(!sp->use_codec_fec);
return 0;
}
}
int AudioCodingModuleImpl::SetPacketLossRate(int loss_rate) {
rtc::CritScope lock(&acm_crit_sect_);
if (HaveValidEncoder("SetPacketLossRate")) {
encoder_stack_->OnReceivedUplinkPacketLossFraction(loss_rate / 100.0);
}
return 0;
}
/////////////////////////////////////////
// (VAD) Voice Activity Detection
//
int AudioCodingModuleImpl::SetVAD(bool enable_dtx,
bool enable_vad,
ACMVADMode mode) {
// Note: |enable_vad| is not used; VAD is enabled based on the DTX setting.
RTC_DCHECK_EQ(enable_dtx, enable_vad);
rtc::CritScope lock(&acm_crit_sect_);
CreateSpeechEncoderIfNecessary(encoder_factory_.get());
if (!encoder_factory_->codec_manager.SetVAD(enable_dtx, mode)) {
return -1;
}
auto* sp = encoder_factory_->codec_manager.GetStackParams();
if (sp->speech_encoder)
encoder_stack_ = encoder_factory_->rent_a_codec.RentEncoderStack(sp);
return 0;
}
// Get VAD/DTX settings.
int AudioCodingModuleImpl::VAD(bool* dtx_enabled, bool* vad_enabled,
ACMVADMode* mode) const {
rtc::CritScope lock(&acm_crit_sect_);
const auto* sp = encoder_factory_->codec_manager.GetStackParams();
*dtx_enabled = *vad_enabled = sp->use_cng;
*mode = sp->vad_mode;
return 0;
}
/////////////////////////////////////////
// Receiver
//
int AudioCodingModuleImpl::InitializeReceiver() {
rtc::CritScope lock(&acm_crit_sect_);
return InitializeReceiverSafe();
}
// Initialize receiver, resets codec database etc.
int AudioCodingModuleImpl::InitializeReceiverSafe() {
// If the receiver is already initialized then we want to destroy any
// existing decoders. After a call to this function, we should have a clean
// start-up.
if (receiver_initialized_)
receiver_.RemoveAllCodecs();
receiver_.ResetInitialDelay();
receiver_.SetMinimumDelay(0);
receiver_.SetMaximumDelay(0);
receiver_.FlushBuffers();
receiver_initialized_ = true;
return 0;
}
// Get current receive frequency.
int AudioCodingModuleImpl::ReceiveFrequency() const {
const auto last_packet_sample_rate = receiver_.last_packet_sample_rate_hz();
return last_packet_sample_rate ? *last_packet_sample_rate
: receiver_.last_output_sample_rate_hz();
}
// Get current playout frequency.
int AudioCodingModuleImpl::PlayoutFrequency() const {
return receiver_.last_output_sample_rate_hz();
}
void AudioCodingModuleImpl::SetReceiveCodecs(
const std::map<int, SdpAudioFormat>& codecs) {
rtc::CritScope lock(&acm_crit_sect_);
receiver_.SetCodecs(codecs);
}
bool AudioCodingModuleImpl::RegisterReceiveCodec(
int rtp_payload_type,
const SdpAudioFormat& audio_format) {
rtc::CritScope lock(&acm_crit_sect_);
RTC_DCHECK(receiver_initialized_);
if (!acm2::RentACodec::IsPayloadTypeValid(rtp_payload_type)) {
RTC_LOG_F(LS_ERROR) << "Invalid payload-type " << rtp_payload_type
<< " for decoder.";
return false;
}
return receiver_.AddCodec(rtp_payload_type, audio_format);
}
int AudioCodingModuleImpl::RegisterReceiveCodec(const CodecInst& codec) {
rtc::CritScope lock(&acm_crit_sect_);
auto* ef = encoder_factory_.get();
return RegisterReceiveCodecUnlocked(
codec, [&] { return ef->rent_a_codec.RentIsacDecoder(codec.plfreq); });
}
int AudioCodingModuleImpl::RegisterReceiveCodec(
const CodecInst& codec,
rtc::FunctionView<std::unique_ptr<AudioDecoder>()> isac_factory) {
rtc::CritScope lock(&acm_crit_sect_);
return RegisterReceiveCodecUnlocked(codec, isac_factory);
}
int AudioCodingModuleImpl::RegisterReceiveCodecUnlocked(
const CodecInst& codec,
rtc::FunctionView<std::unique_ptr<AudioDecoder>()> isac_factory) {
RTC_DCHECK(receiver_initialized_);
if (codec.channels > 2) {
RTC_LOG_F(LS_ERROR) << "Unsupported number of channels: " << codec.channels;
return -1;
}
auto codec_id = acm2::RentACodec::CodecIdByParams(codec.plname, codec.plfreq,
codec.channels);
if (!codec_id) {
RTC_LOG_F(LS_ERROR)
<< "Wrong codec params to be registered as receive codec";
return -1;
}
auto codec_index = acm2::RentACodec::CodecIndexFromId(*codec_id);
RTC_CHECK(codec_index) << "Invalid codec ID: " << static_cast<int>(*codec_id);
// Check if the payload-type is valid.
if (!acm2::RentACodec::IsPayloadTypeValid(codec.pltype)) {
RTC_LOG_F(LS_ERROR) << "Invalid payload type " << codec.pltype << " for "
<< codec.plname;
return -1;
}
AudioDecoder* isac_decoder = nullptr;
if (STR_CASE_CMP(codec.plname, "isac") == 0) {
std::unique_ptr<AudioDecoder>& saved_isac_decoder =
codec.plfreq == 16000 ? isac_decoder_16k_ : isac_decoder_32k_;
if (!saved_isac_decoder) {
saved_isac_decoder = isac_factory();
}
isac_decoder = saved_isac_decoder.get();
}
return receiver_.AddCodec(*codec_index, codec.pltype, codec.channels,
codec.plfreq, isac_decoder, codec.plname);
}
int AudioCodingModuleImpl::RegisterExternalReceiveCodec(
int rtp_payload_type,
AudioDecoder* external_decoder,
int sample_rate_hz,
int num_channels,
const std::string& name) {
rtc::CritScope lock(&acm_crit_sect_);
RTC_DCHECK(receiver_initialized_);
if (num_channels > 2 || num_channels < 0) {
RTC_LOG_F(LS_ERROR) << "Unsupported number of channels: " << num_channels;
return -1;
}
// Check if the payload-type is valid.
if (!acm2::RentACodec::IsPayloadTypeValid(rtp_payload_type)) {
RTC_LOG_F(LS_ERROR) << "Invalid payload-type " << rtp_payload_type
<< " for external decoder.";
return -1;
}
return receiver_.AddCodec(-1 /* external */, rtp_payload_type, num_channels,
sample_rate_hz, external_decoder, name);
}
// Get current received codec.
int AudioCodingModuleImpl::ReceiveCodec(CodecInst* current_codec) const {
rtc::CritScope lock(&acm_crit_sect_);
return receiver_.LastAudioCodec(current_codec);
}
rtc::Optional<SdpAudioFormat> AudioCodingModuleImpl::ReceiveFormat() const {
rtc::CritScope lock(&acm_crit_sect_);
return receiver_.LastAudioFormat();
}
// Incoming packet from network parsed and ready for decode.
int AudioCodingModuleImpl::IncomingPacket(const uint8_t* incoming_payload,
const size_t payload_length,
const WebRtcRTPHeader& rtp_header) {
RTC_DCHECK_EQ(payload_length == 0, incoming_payload == nullptr);
return receiver_.InsertPacket(
rtp_header,
rtc::ArrayView<const uint8_t>(incoming_payload, payload_length));
}
// Minimum playout delay (Used for lip-sync).
int AudioCodingModuleImpl::SetMinimumPlayoutDelay(int time_ms) {
if ((time_ms < 0) || (time_ms > 10000)) {
RTC_LOG(LS_ERROR) << "Delay must be in the range of 0-10000 milliseconds.";
return -1;
}
return receiver_.SetMinimumDelay(time_ms);
}
int AudioCodingModuleImpl::SetMaximumPlayoutDelay(int time_ms) {
if ((time_ms < 0) || (time_ms > 10000)) {
RTC_LOG(LS_ERROR) << "Delay must be in the range of 0-10000 milliseconds.";
return -1;
}
return receiver_.SetMaximumDelay(time_ms);
}
// Get 10 milliseconds of raw audio data to play out.
// Automatic resample to the requested frequency.
int AudioCodingModuleImpl::PlayoutData10Ms(int desired_freq_hz,
AudioFrame* audio_frame,
bool* muted) {
// GetAudio always returns 10 ms, at the requested sample rate.
if (receiver_.GetAudio(desired_freq_hz, audio_frame, muted) != 0) {
RTC_LOG(LS_ERROR) << "PlayoutData failed, RecOut Failed";
return -1;
}
return 0;
}
int AudioCodingModuleImpl::PlayoutData10Ms(int desired_freq_hz,
AudioFrame* audio_frame) {
bool muted;
int ret = PlayoutData10Ms(desired_freq_hz, audio_frame, &muted);
RTC_DCHECK(!muted);
return ret;
}
/////////////////////////////////////////
// Statistics
//
// TODO(turajs) change the return value to void. Also change the corresponding
// NetEq function.
int AudioCodingModuleImpl::GetNetworkStatistics(NetworkStatistics* statistics) {
receiver_.GetNetworkStatistics(statistics);
return 0;
}
int AudioCodingModuleImpl::RegisterVADCallback(ACMVADCallback* vad_callback) {
RTC_LOG(LS_VERBOSE) << "RegisterVADCallback()";
rtc::CritScope lock(&callback_crit_sect_);
vad_callback_ = vad_callback;
return 0;
}
int AudioCodingModuleImpl::SetOpusApplication(OpusApplicationMode application) {
rtc::CritScope lock(&acm_crit_sect_);
if (!HaveValidEncoder("SetOpusApplication")) {
return -1;
}
AudioEncoder::Application app;
switch (application) {
case kVoip:
app = AudioEncoder::Application::kSpeech;
break;
case kAudio:
app = AudioEncoder::Application::kAudio;
break;
default:
FATAL();
return 0;
}
return encoder_stack_->SetApplication(app) ? 0 : -1;
}
// Informs Opus encoder of the maximum playback rate the receiver will render.
int AudioCodingModuleImpl::SetOpusMaxPlaybackRate(int frequency_hz) {
rtc::CritScope lock(&acm_crit_sect_);
if (!HaveValidEncoder("SetOpusMaxPlaybackRate")) {
return -1;
}
encoder_stack_->SetMaxPlaybackRate(frequency_hz);
return 0;
}
int AudioCodingModuleImpl::EnableOpusDtx() {
rtc::CritScope lock(&acm_crit_sect_);
if (!HaveValidEncoder("EnableOpusDtx")) {
return -1;
}
return encoder_stack_->SetDtx(true) ? 0 : -1;
}
int AudioCodingModuleImpl::DisableOpusDtx() {
rtc::CritScope lock(&acm_crit_sect_);
if (!HaveValidEncoder("DisableOpusDtx")) {
return -1;
}
return encoder_stack_->SetDtx(false) ? 0 : -1;
}
int32_t AudioCodingModuleImpl::PlayoutTimestamp(uint32_t* timestamp) {
rtc::Optional<uint32_t> ts = PlayoutTimestamp();
if (!ts)
return -1;
*timestamp = *ts;
return 0;
}
rtc::Optional<uint32_t> AudioCodingModuleImpl::PlayoutTimestamp() {
return receiver_.GetPlayoutTimestamp();
}
int AudioCodingModuleImpl::FilteredCurrentDelayMs() const {
return receiver_.FilteredCurrentDelayMs();
}
bool AudioCodingModuleImpl::HaveValidEncoder(const char* caller_name) const {
if (!encoder_stack_) {
RTC_LOG(LS_ERROR) << caller_name << " failed: No send codec is registered.";
return false;
}
return true;
}
int AudioCodingModuleImpl::UnregisterReceiveCodec(uint8_t payload_type) {
return receiver_.RemoveCodec(payload_type);
}
int AudioCodingModuleImpl::EnableNack(size_t max_nack_list_size) {
return receiver_.EnableNack(max_nack_list_size);
}
void AudioCodingModuleImpl::DisableNack() {
receiver_.DisableNack();
}
std::vector<uint16_t> AudioCodingModuleImpl::GetNackList(
int64_t round_trip_time_ms) const {
return receiver_.GetNackList(round_trip_time_ms);
}
int AudioCodingModuleImpl::LeastRequiredDelayMs() const {
return receiver_.LeastRequiredDelayMs();
}
void AudioCodingModuleImpl::GetDecodingCallStatistics(
AudioDecodingCallStats* call_stats) const {
receiver_.GetDecodingCallStatistics(call_stats);
}
ANAStats AudioCodingModuleImpl::GetANAStats() const {
rtc::CritScope lock(&acm_crit_sect_);
if (encoder_stack_)
return encoder_stack_->GetANAStats();
// If no encoder is set, return default stats.
return ANAStats();
}
} // namespace
AudioCodingModule::Config::Config()
: neteq_config(), clock(Clock::GetRealTimeClock()) {
// Post-decode VAD is disabled by default in NetEq, however, Audio
// Conference Mixer relies on VAD decisions and fails without them.
neteq_config.enable_post_decode_vad = true;
}
AudioCodingModule::Config::Config(const Config&) = default;
AudioCodingModule::Config::~Config() = default;
AudioCodingModule* AudioCodingModule::Create(int id) {
RTC_UNUSED(id);
return Create();
}
// Create module
AudioCodingModule* AudioCodingModule::Create() {
Config config;
config.clock = Clock::GetRealTimeClock();
config.decoder_factory = CreateBuiltinAudioDecoderFactory();
return Create(config);
}
AudioCodingModule* AudioCodingModule::Create(Clock* clock) {
Config config;
config.clock = clock;
config.decoder_factory = CreateBuiltinAudioDecoderFactory();
return Create(config);
}
AudioCodingModule* AudioCodingModule::Create(const Config& config) {
if (!config.decoder_factory) {
// TODO(ossu): Backwards compatibility. Will be removed after a deprecation
// cycle.
Config config_copy = config;
config_copy.decoder_factory = CreateBuiltinAudioDecoderFactory();
return new AudioCodingModuleImpl(config_copy);
}
return new AudioCodingModuleImpl(config);
}
int AudioCodingModule::NumberOfCodecs() {
return static_cast<int>(acm2::RentACodec::NumberOfCodecs());
}
int AudioCodingModule::Codec(int list_id, CodecInst* codec) {
auto codec_id = acm2::RentACodec::CodecIdFromIndex(list_id);
if (!codec_id)
return -1;
auto ci = acm2::RentACodec::CodecInstById(*codec_id);
if (!ci)
return -1;
*codec = *ci;
return 0;
}
int AudioCodingModule::Codec(const char* payload_name,
CodecInst* codec,
int sampling_freq_hz,
size_t channels) {
rtc::Optional<CodecInst> ci = acm2::RentACodec::CodecInstByParams(
payload_name, sampling_freq_hz, channels);
if (ci) {
*codec = *ci;
return 0;
} else {
// We couldn't find a matching codec, so set the parameters to unacceptable
// values and return.
codec->plname[0] = '\0';
codec->pltype = -1;
codec->pacsize = 0;
codec->rate = 0;
codec->plfreq = 0;
return -1;
}
}
int AudioCodingModule::Codec(const char* payload_name,
int sampling_freq_hz,
size_t channels) {
rtc::Optional<acm2::RentACodec::CodecId> ci =
acm2::RentACodec::CodecIdByParams(payload_name, sampling_freq_hz,
channels);
if (!ci)
return -1;
rtc::Optional<int> i = acm2::RentACodec::CodecIndexFromId(*ci);
return i ? *i : -1;
}
// Checks the validity of the parameters of the given codec
bool AudioCodingModule::IsCodecValid(const CodecInst& codec) {
bool valid = acm2::RentACodec::IsCodecValid(codec);
if (!valid)
RTC_LOG(LS_ERROR) << "Invalid codec setting";
return valid;
}
} // namespace webrtc
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