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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 <cstdint>
#include "absl/strings/match.h"
#include "absl/strings/string_view.h"
#include "api/array_view.h"
#include "modules/audio_coding/acm2/acm_remixing.h"
#include "modules/audio_coding/acm2/acm_resampler.h"
#include "modules/include/module_common_types.h"
#include "modules/include/module_common_types_public.h"
#include "rtc_base/buffer.h"
#include "rtc_base/checks.h"
#include "rtc_base/logging.h"
#include "rtc_base/numerics/safe_conversions.h"
#include "rtc_base/synchronization/mutex.h"
#include "rtc_base/thread_annotations.h"
#include "system_wrappers/include/metrics.h"
namespace webrtc {
namespace {
// Initial size for the buffer in InputBuffer. This matches 6 channels of 10 ms
// 48 kHz data.
constexpr size_t kInitialInputDataBufferSize = 6 * 480;
constexpr int32_t kMaxInputSampleRateHz = 192000;
class AudioCodingModuleImpl final : public AudioCodingModule {
public:
explicit AudioCodingModuleImpl();
~AudioCodingModuleImpl() override;
void Reset() override;
/////////////////////////////////////////
// Sender
//
void ModifyEncoder(rtc::FunctionView<void(std::unique_ptr<AudioEncoder>*)>
modifier) 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;
/////////////////////////////////////////
// (FEC) Forward Error Correction (codec internal)
//
// Set target packet loss rate
int SetPacketLossRate(int loss_rate) override;
/////////////////////////////////////////
// Statistics
//
ANAStats GetANAStats() const override;
int GetTargetBitrate() const override;
private:
struct InputData {
InputData() : buffer(kInitialInputDataBufferSize) {}
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.
std::vector<int16_t> buffer;
};
InputData input_data_ RTC_GUARDED_BY(acm_mutex_);
// 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(absl::string_view 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 Add10MsDataInternal(const AudioFrame& audio_frame, InputData* input_data)
RTC_EXCLUSIVE_LOCKS_REQUIRED(acm_mutex_);
// TODO(bugs.webrtc.org/10739): change `absolute_capture_timestamp_ms` to
// int64_t when it always receives a valid value.
int Encode(const InputData& input_data,
std::optional<int64_t> absolute_capture_timestamp_ms)
RTC_EXCLUSIVE_LOCKS_REQUIRED(acm_mutex_);
bool HaveValidEncoder(absl::string_view caller_name) const
RTC_EXCLUSIVE_LOCKS_REQUIRED(acm_mutex_);
// 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_mutex_);
// Change required states after starting to receive the codec corresponding
// to `index`.
int UpdateUponReceivingCodec(int index);
mutable Mutex acm_mutex_;
rtc::Buffer encode_buffer_ RTC_GUARDED_BY(acm_mutex_);
uint32_t expected_codec_ts_ RTC_GUARDED_BY(acm_mutex_);
uint32_t expected_in_ts_ RTC_GUARDED_BY(acm_mutex_);
acm2::ACMResampler resampler_ RTC_GUARDED_BY(acm_mutex_);
ChangeLogger bitrate_logger_ RTC_GUARDED_BY(acm_mutex_);
// Current encoder stack, provided by a call to RegisterEncoder.
std::unique_ptr<AudioEncoder> encoder_stack_ RTC_GUARDED_BY(acm_mutex_);
// This is to keep track of CN instances where we can send DTMFs.
uint8_t previous_pltype_ RTC_GUARDED_BY(acm_mutex_);
AudioFrame preprocess_frame_ RTC_GUARDED_BY(acm_mutex_);
bool first_10ms_data_ RTC_GUARDED_BY(acm_mutex_);
bool first_frame_ RTC_GUARDED_BY(acm_mutex_);
uint32_t last_timestamp_ RTC_GUARDED_BY(acm_mutex_);
uint32_t last_rtp_timestamp_ RTC_GUARDED_BY(acm_mutex_);
std::optional<int64_t> absolute_capture_timestamp_ms_
RTC_GUARDED_BY(acm_mutex_);
Mutex callback_mutex_;
AudioPacketizationCallback* packetization_callback_
RTC_GUARDED_BY(callback_mutex_);
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));
}
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()
: expected_codec_ts_(0xD87F3F9F),
expected_in_ts_(0xD87F3F9F),
bitrate_logger_("WebRTC.Audio.TargetBitrateInKbps"),
encoder_stack_(nullptr),
previous_pltype_(255),
first_10ms_data_(false),
first_frame_(true),
packetization_callback_(NULL),
codec_histogram_bins_log_(),
number_of_consecutive_empty_packets_(0) {
RTC_LOG(LS_INFO) << "Created";
}
AudioCodingModuleImpl::~AudioCodingModuleImpl() = default;
int32_t AudioCodingModuleImpl::Encode(
const InputData& input_data,
std::optional<int64_t> absolute_capture_timestamp_ms) {
// TODO(bugs.webrtc.org/10739): add dcheck that
// `audio_frame.absolute_capture_timestamp_ms()` always has a value.
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::dchecked_cast<uint32_t>(rtc::CheckedDivExact(
int64_t{input_data.input_timestamp - last_timestamp_} *
encoder_stack_->RtpTimestampRateHz(),
int64_t{encoder_stack_->SampleRateHz()}));
last_timestamp_ = input_data.input_timestamp;
last_rtp_timestamp_ = rtp_timestamp;
first_frame_ = false;
if (!absolute_capture_timestamp_ms_.has_value()) {
absolute_capture_timestamp_ms_ = absolute_capture_timestamp_ms;
}
// 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);
}
}
AudioFrameType frame_type;
if (encode_buffer_.size() == 0 && encoded_info.send_even_if_empty) {
frame_type = AudioFrameType::kEmptyFrame;
encoded_info.payload_type = previous_pltype;
} else {
RTC_DCHECK_GT(encode_buffer_.size(), 0);
frame_type = encoded_info.speech ? AudioFrameType::kAudioFrameSpeech
: AudioFrameType::kAudioFrameCN;
}
{
MutexLock lock(&callback_mutex_);
if (packetization_callback_) {
packetization_callback_->SendData(
frame_type, encoded_info.payload_type, encoded_info.encoded_timestamp,
encode_buffer_.data(), encode_buffer_.size(),
absolute_capture_timestamp_ms_.value_or(-1));
}
}
absolute_capture_timestamp_ms_.reset();
previous_pltype_ = encoded_info.payload_type;
return static_cast<int32_t>(encode_buffer_.size());
}
/////////////////////////////////////////
// Sender
//
void AudioCodingModuleImpl::Reset() {
MutexLock lock(&acm_mutex_);
absolute_capture_timestamp_ms_.reset();
if (HaveValidEncoder("Reset")) {
encoder_stack_->Reset();
}
}
void AudioCodingModuleImpl::ModifyEncoder(
rtc::FunctionView<void(std::unique_ptr<AudioEncoder>*)> modifier) {
MutexLock lock(&acm_mutex_);
modifier(&encoder_stack_);
}
// Register a transport callback which will be called to deliver
// the encoded buffers.
int AudioCodingModuleImpl::RegisterTransportCallback(
AudioPacketizationCallback* transport) {
MutexLock lock(&callback_mutex_);
packetization_callback_ = transport;
return 0;
}
// Add 10MS of raw (PCM) audio data to the encoder.
int AudioCodingModuleImpl::Add10MsData(const AudioFrame& audio_frame) {
MutexLock lock(&acm_mutex_);
int r = Add10MsDataInternal(audio_frame, &input_data_);
// TODO(bugs.webrtc.org/10739): add dcheck that
// `audio_frame.absolute_capture_timestamp_ms()` always has a value.
return r < 0
? r
: Encode(input_data_, audio_frame.absolute_capture_timestamp_ms());
}
int AudioCodingModuleImpl::Add10MsDataInternal(const AudioFrame& audio_frame,
InputData* input_data) {
if (audio_frame.samples_per_channel_ == 0) {
RTC_DCHECK_NOTREACHED();
RTC_LOG(LS_ERROR) << "Cannot Add 10 ms audio, payload length is zero";
return -1;
}
if (audio_frame.sample_rate_hz_ > kMaxInputSampleRateHz) {
RTC_DCHECK_NOTREACHED();
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 &&
audio_frame.num_channels_ != 4 && audio_frame.num_channels_ != 6 &&
audio_frame.num_channels_ != 8) {
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;
// TODO(yujo): Skip encode of muted frames.
input_data->input_timestamp = ptr_frame->timestamp_;
input_data->length_per_channel = ptr_frame->samples_per_channel_;
input_data->audio_channel = current_num_channels;
if (!same_num_channels) {
// Remixes the input frame to the output data and in the process resize the
// output data if needed.
ReMixFrame(*ptr_frame, current_num_channels, &input_data->buffer);
// For pushing data to primary, point the `ptr_audio` to correct buffer.
input_data->audio = input_data->buffer.data();
RTC_DCHECK_GE(input_data->buffer.size(),
input_data->length_per_channel * input_data->audio_channel);
} else {
// When adding data to encoders this pointer is pointing to an audio buffer
// with correct number of channels.
input_data->audio = ptr_frame->data();
}
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_;
preprocess_frame_.samples_per_channel_ = in_frame.samples_per_channel_;
std::array<int16_t, AudioFrame::kMaxDataSizeSamples> audio;
const int16_t* src_ptr_audio;
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.data() : preprocess_frame_.mutable_data();
RTC_DCHECK_GE(audio.size(), preprocess_frame_.samples_per_channel_);
RTC_DCHECK_GE(audio.size(), in_frame.samples_per_channel_);
DownMixFrame(in_frame,
rtc::ArrayView<int16_t>(
dest_ptr_audio, preprocess_frame_.samples_per_channel_));
preprocess_frame_.num_channels_ = 1;
// Set the input of the resampler to the down-mixed signal.
src_ptr_audio = audio.data();
} else {
// Set the input of the resampler to the original data.
src_ptr_audio = in_frame.data();
}
preprocess_frame_.timestamp_ = expected_codec_ts_;
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;
}
/////////////////////////////////////////
// (FEC) Forward Error Correction (codec internal)
//
int AudioCodingModuleImpl::SetPacketLossRate(int loss_rate) {
MutexLock lock(&acm_mutex_);
if (HaveValidEncoder("SetPacketLossRate")) {
encoder_stack_->OnReceivedUplinkPacketLossFraction(loss_rate / 100.0);
}
return 0;
}
/////////////////////////////////////////
// Statistics
//
bool AudioCodingModuleImpl::HaveValidEncoder(
absl::string_view caller_name) const {
if (!encoder_stack_) {
RTC_LOG(LS_ERROR) << caller_name << " failed: No send codec is registered.";
return false;
}
return true;
}
ANAStats AudioCodingModuleImpl::GetANAStats() const {
MutexLock lock(&acm_mutex_);
if (encoder_stack_)
return encoder_stack_->GetANAStats();
// If no encoder is set, return default stats.
return ANAStats();
}
int AudioCodingModuleImpl::GetTargetBitrate() const {
MutexLock lock(&acm_mutex_);
if (!encoder_stack_) {
return -1;
}
return encoder_stack_->GetTargetBitrate();
}
} // namespace
std::unique_ptr<AudioCodingModule> AudioCodingModule::Create() {
return std::make_unique<AudioCodingModuleImpl>();
}
} // namespace webrtc