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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_device/audio_device_buffer.h"
#include <string.h>
#include <cmath>
#include <cstddef>
#include <cstdint>
#include "common_audio/signal_processing/include/signal_processing_library.h"
#include "rtc_base/bind.h"
#include "rtc_base/checks.h"
#include "rtc_base/logging.h"
#include "rtc_base/time_utils.h"
#include "system_wrappers/include/metrics.h"
namespace webrtc {
static const char kTimerQueueName[] = "AudioDeviceBufferTimer";
// Time between two sucessive calls to LogStats().
static const size_t kTimerIntervalInSeconds = 10;
static const size_t kTimerIntervalInMilliseconds =
kTimerIntervalInSeconds * rtc::kNumMillisecsPerSec;
// Min time required to qualify an audio session as a "call". If playout or
// recording has been active for less than this time we will not store any
// logs or UMA stats but instead consider the call as too short.
static const size_t kMinValidCallTimeTimeInSeconds = 10;
static const size_t kMinValidCallTimeTimeInMilliseconds =
kMinValidCallTimeTimeInSeconds * rtc::kNumMillisecsPerSec;
#ifdef AUDIO_DEVICE_PLAYS_SINUS_TONE
static const double k2Pi = 6.28318530717959;
#endif
AudioDeviceBuffer::AudioDeviceBuffer(TaskQueueFactory* task_queue_factory)
: task_queue_(task_queue_factory->CreateTaskQueue(
kTimerQueueName,
TaskQueueFactory::Priority::NORMAL)),
audio_transport_cb_(nullptr),
rec_sample_rate_(0),
play_sample_rate_(0),
rec_channels_(0),
play_channels_(0),
playing_(false),
recording_(false),
typing_status_(false),
play_delay_ms_(0),
rec_delay_ms_(0),
num_stat_reports_(0),
last_timer_task_time_(0),
rec_stat_count_(0),
play_stat_count_(0),
play_start_time_(0),
only_silence_recorded_(true),
log_stats_(false) {
RTC_LOG(INFO) << "AudioDeviceBuffer::ctor";
#ifdef AUDIO_DEVICE_PLAYS_SINUS_TONE
phase_ = 0.0;
RTC_LOG(WARNING) << "AUDIO_DEVICE_PLAYS_SINUS_TONE is defined!";
#endif
}
AudioDeviceBuffer::~AudioDeviceBuffer() {
RTC_DCHECK_RUN_ON(&main_thread_checker_);
RTC_DCHECK(!playing_);
RTC_DCHECK(!recording_);
RTC_LOG(INFO) << "AudioDeviceBuffer::~dtor";
}
int32_t AudioDeviceBuffer::RegisterAudioCallback(
AudioTransport* audio_callback) {
RTC_DCHECK_RUN_ON(&main_thread_checker_);
RTC_LOG(INFO) << __FUNCTION__;
if (playing_ || recording_) {
RTC_LOG(LS_ERROR) << "Failed to set audio transport since media was active";
return -1;
}
audio_transport_cb_ = audio_callback;
return 0;
}
void AudioDeviceBuffer::StartPlayout() {
RTC_DCHECK_RUN_ON(&main_thread_checker_);
// TODO(henrika): allow for usage of DCHECK(!playing_) here instead. Today the
// ADM allows calling Start(), Start() by ignoring the second call but it
// makes more sense to only allow one call.
if (playing_) {
return;
}
RTC_LOG(INFO) << __FUNCTION__;
// Clear members tracking playout stats and do it on the task queue.
task_queue_.PostTask([this] { ResetPlayStats(); });
// Start a periodic timer based on task queue if not already done by the
// recording side.
if (!recording_) {
StartPeriodicLogging();
}
const int64_t now_time = rtc::TimeMillis();
// Clear members that are only touched on the main (creating) thread.
play_start_time_ = now_time;
playing_ = true;
}
void AudioDeviceBuffer::StartRecording() {
RTC_DCHECK_RUN_ON(&main_thread_checker_);
if (recording_) {
return;
}
RTC_LOG(INFO) << __FUNCTION__;
// Clear members tracking recording stats and do it on the task queue.
task_queue_.PostTask([this] { ResetRecStats(); });
// Start a periodic timer based on task queue if not already done by the
// playout side.
if (!playing_) {
StartPeriodicLogging();
}
// Clear members that will be touched on the main (creating) thread.
rec_start_time_ = rtc::TimeMillis();
recording_ = true;
// And finally a member which can be modified on the native audio thread.
// It is safe to do so since we know by design that the owning ADM has not
// yet started the native audio recording.
only_silence_recorded_ = true;
}
void AudioDeviceBuffer::StopPlayout() {
RTC_DCHECK_RUN_ON(&main_thread_checker_);
if (!playing_) {
return;
}
RTC_LOG(INFO) << __FUNCTION__;
playing_ = false;
// Stop periodic logging if no more media is active.
if (!recording_) {
StopPeriodicLogging();
}
RTC_LOG(INFO) << "total playout time: " << rtc::TimeSince(play_start_time_);
}
void AudioDeviceBuffer::StopRecording() {
RTC_DCHECK_RUN_ON(&main_thread_checker_);
if (!recording_) {
return;
}
RTC_LOG(INFO) << __FUNCTION__;
recording_ = false;
// Stop periodic logging if no more media is active.
if (!playing_) {
StopPeriodicLogging();
}
// Add UMA histogram to keep track of the case when only zeros have been
// recorded. Measurements (max of absolute level) are taken twice per second,
// which means that if e.g 10 seconds of audio has been recorded, a total of
// 20 level estimates must all be identical to zero to trigger the histogram.
// |only_silence_recorded_| can only be cleared on the native audio thread
// that drives audio capture but we know by design that the audio has stopped
// when this method is called, hence there should not be aby conflicts. Also,
// the fact that |only_silence_recorded_| can be affected during the complete
// call makes chances of conflicts with potentially one last callback very
// small.
const size_t time_since_start = rtc::TimeSince(rec_start_time_);
if (time_since_start > kMinValidCallTimeTimeInMilliseconds) {
const int only_zeros = static_cast<int>(only_silence_recorded_);
RTC_HISTOGRAM_BOOLEAN("WebRTC.Audio.RecordedOnlyZeros", only_zeros);
RTC_LOG(INFO) << "HISTOGRAM(WebRTC.Audio.RecordedOnlyZeros): "
<< only_zeros;
}
RTC_LOG(INFO) << "total recording time: " << time_since_start;
}
int32_t AudioDeviceBuffer::SetRecordingSampleRate(uint32_t fsHz) {
RTC_LOG(INFO) << "SetRecordingSampleRate(" << fsHz << ")";
rec_sample_rate_ = fsHz;
return 0;
}
int32_t AudioDeviceBuffer::SetPlayoutSampleRate(uint32_t fsHz) {
RTC_LOG(INFO) << "SetPlayoutSampleRate(" << fsHz << ")";
play_sample_rate_ = fsHz;
return 0;
}
uint32_t AudioDeviceBuffer::RecordingSampleRate() const {
return rec_sample_rate_;
}
uint32_t AudioDeviceBuffer::PlayoutSampleRate() const {
return play_sample_rate_;
}
int32_t AudioDeviceBuffer::SetRecordingChannels(size_t channels) {
RTC_LOG(INFO) << "SetRecordingChannels(" << channels << ")";
rec_channels_ = channels;
return 0;
}
int32_t AudioDeviceBuffer::SetPlayoutChannels(size_t channels) {
RTC_LOG(INFO) << "SetPlayoutChannels(" << channels << ")";
play_channels_ = channels;
return 0;
}
size_t AudioDeviceBuffer::RecordingChannels() const {
return rec_channels_;
}
size_t AudioDeviceBuffer::PlayoutChannels() const {
return play_channels_;
}
int32_t AudioDeviceBuffer::SetTypingStatus(bool typing_status) {
typing_status_ = typing_status;
return 0;
}
void AudioDeviceBuffer::SetVQEData(int play_delay_ms, int rec_delay_ms) {
play_delay_ms_ = play_delay_ms;
rec_delay_ms_ = rec_delay_ms;
}
int32_t AudioDeviceBuffer::SetRecordedBuffer(const void* audio_buffer,
size_t samples_per_channel) {
// Copy the complete input buffer to the local buffer.
const size_t old_size = rec_buffer_.size();
rec_buffer_.SetData(static_cast<const int16_t*>(audio_buffer),
rec_channels_ * samples_per_channel);
// Keep track of the size of the recording buffer. Only updated when the
// size changes, which is a rare event.
if (old_size != rec_buffer_.size()) {
RTC_LOG(LS_INFO) << "Size of recording buffer: " << rec_buffer_.size();
}
// Derive a new level value twice per second and check if it is non-zero.
int16_t max_abs = 0;
RTC_DCHECK_LT(rec_stat_count_, 50);
if (++rec_stat_count_ >= 50) {
// Returns the largest absolute value in a signed 16-bit vector.
max_abs = WebRtcSpl_MaxAbsValueW16(rec_buffer_.data(), rec_buffer_.size());
rec_stat_count_ = 0;
// Set |only_silence_recorded_| to false as soon as at least one detection
// of a non-zero audio packet is found. It can only be restored to true
// again by restarting the call.
if (max_abs > 0) {
only_silence_recorded_ = false;
}
}
// Update recording stats which is used as base for periodic logging of the
// audio input state.
UpdateRecStats(max_abs, samples_per_channel);
return 0;
}
int32_t AudioDeviceBuffer::DeliverRecordedData() {
if (!audio_transport_cb_) {
RTC_LOG(LS_WARNING) << "Invalid audio transport";
return 0;
}
const size_t frames = rec_buffer_.size() / rec_channels_;
const size_t bytes_per_frame = rec_channels_ * sizeof(int16_t);
uint32_t new_mic_level_dummy = 0;
uint32_t total_delay_ms = play_delay_ms_ + rec_delay_ms_;
int32_t res = audio_transport_cb_->RecordedDataIsAvailable(
rec_buffer_.data(), frames, bytes_per_frame, rec_channels_,
rec_sample_rate_, total_delay_ms, 0, 0, typing_status_,
new_mic_level_dummy);
if (res == -1) {
RTC_LOG(LS_ERROR) << "RecordedDataIsAvailable() failed";
}
return 0;
}
int32_t AudioDeviceBuffer::RequestPlayoutData(size_t samples_per_channel) {
// The consumer can change the requested size on the fly and we therefore
// resize the buffer accordingly. Also takes place at the first call to this
// method.
const size_t total_samples = play_channels_ * samples_per_channel;
if (play_buffer_.size() != total_samples) {
play_buffer_.SetSize(total_samples);
RTC_LOG(LS_INFO) << "Size of playout buffer: " << play_buffer_.size();
}
size_t num_samples_out(0);
// It is currently supported to start playout without a valid audio
// transport object. Leads to warning and silence.
if (!audio_transport_cb_) {
RTC_LOG(LS_WARNING) << "Invalid audio transport";
return 0;
}
// Retrieve new 16-bit PCM audio data using the audio transport instance.
int64_t elapsed_time_ms = -1;
int64_t ntp_time_ms = -1;
const size_t bytes_per_frame = play_channels_ * sizeof(int16_t);
uint32_t res = audio_transport_cb_->NeedMorePlayData(
samples_per_channel, bytes_per_frame, play_channels_, play_sample_rate_,
play_buffer_.data(), num_samples_out, &elapsed_time_ms, &ntp_time_ms);
if (res != 0) {
RTC_LOG(LS_ERROR) << "NeedMorePlayData() failed";
}
// Derive a new level value twice per second.
int16_t max_abs = 0;
RTC_DCHECK_LT(play_stat_count_, 50);
if (++play_stat_count_ >= 50) {
// Returns the largest absolute value in a signed 16-bit vector.
max_abs =
WebRtcSpl_MaxAbsValueW16(play_buffer_.data(), play_buffer_.size());
play_stat_count_ = 0;
}
// Update playout stats which is used as base for periodic logging of the
// audio output state.
UpdatePlayStats(max_abs, num_samples_out / play_channels_);
return static_cast<int32_t>(num_samples_out / play_channels_);
}
int32_t AudioDeviceBuffer::GetPlayoutData(void* audio_buffer) {
RTC_DCHECK_GT(play_buffer_.size(), 0);
#ifdef AUDIO_DEVICE_PLAYS_SINUS_TONE
const double phase_increment =
k2Pi * 440.0 / static_cast<double>(play_sample_rate_);
int16_t* destination_r = reinterpret_cast<int16_t*>(audio_buffer);
if (play_channels_ == 1) {
for (size_t i = 0; i < play_buffer_.size(); ++i) {
destination_r[i] = static_cast<int16_t>((sin(phase_) * (1 << 14)));
phase_ += phase_increment;
}
} else if (play_channels_ == 2) {
for (size_t i = 0; i < play_buffer_.size() / 2; ++i) {
destination_r[2 * i] = destination_r[2 * i + 1] =
static_cast<int16_t>((sin(phase_) * (1 << 14)));
phase_ += phase_increment;
}
}
#else
memcpy(audio_buffer, play_buffer_.data(),
play_buffer_.size() * sizeof(int16_t));
#endif
// Return samples per channel or number of frames.
return static_cast<int32_t>(play_buffer_.size() / play_channels_);
}
void AudioDeviceBuffer::StartPeriodicLogging() {
task_queue_.PostTask(rtc::Bind(&AudioDeviceBuffer::LogStats, this,
AudioDeviceBuffer::LOG_START));
}
void AudioDeviceBuffer::StopPeriodicLogging() {
task_queue_.PostTask(rtc::Bind(&AudioDeviceBuffer::LogStats, this,
AudioDeviceBuffer::LOG_STOP));
}
void AudioDeviceBuffer::LogStats(LogState state) {
RTC_DCHECK_RUN_ON(&task_queue_);
int64_t now_time = rtc::TimeMillis();
if (state == AudioDeviceBuffer::LOG_START) {
// Reset counters at start. We will not add any logging in this state but
// the timer will started by posting a new (delayed) task.
num_stat_reports_ = 0;
last_timer_task_time_ = now_time;
log_stats_ = true;
} else if (state == AudioDeviceBuffer::LOG_STOP) {
// Stop logging and posting new tasks.
log_stats_ = false;
} else if (state == AudioDeviceBuffer::LOG_ACTIVE) {
// Keep logging unless logging was disabled while task was posted.
}
// Avoid adding more logs since we are in STOP mode.
if (!log_stats_) {
return;
}
int64_t next_callback_time = now_time + kTimerIntervalInMilliseconds;
int64_t time_since_last = rtc::TimeDiff(now_time, last_timer_task_time_);
last_timer_task_time_ = now_time;
Stats stats;
{
rtc::CritScope cs(&lock_);
stats = stats_;
stats_.max_rec_level = 0;
stats_.max_play_level = 0;
}
// Cache current sample rate from atomic members.
const uint32_t rec_sample_rate = rec_sample_rate_;
const uint32_t play_sample_rate = play_sample_rate_;
// Log the latest statistics but skip the first two rounds just after state
// was set to LOG_START to ensure that we have at least one full stable
// 10-second interval for sample-rate estimation. Hence, first printed log
// will be after ~20 seconds.
if (++num_stat_reports_ > 2 &&
static_cast<size_t>(time_since_last) > kTimerIntervalInMilliseconds / 2) {
uint32_t diff_samples = stats.rec_samples - last_stats_.rec_samples;
float rate = diff_samples / (static_cast<float>(time_since_last) / 1000.0);
uint32_t abs_diff_rate_in_percent = 0;
if (rec_sample_rate > 0 && rate > 0) {
abs_diff_rate_in_percent = static_cast<uint32_t>(
0.5f +
((100.0f * std::abs(rate - rec_sample_rate)) / rec_sample_rate));
RTC_HISTOGRAM_PERCENTAGE("WebRTC.Audio.RecordSampleRateOffsetInPercent",
abs_diff_rate_in_percent);
RTC_LOG(INFO) << "[REC : " << time_since_last << "msec, "
<< rec_sample_rate / 1000 << "kHz] callbacks: "
<< stats.rec_callbacks - last_stats_.rec_callbacks << ", "
<< "samples: " << diff_samples << ", "
<< "rate: " << static_cast<int>(rate + 0.5) << ", "
<< "rate diff: " << abs_diff_rate_in_percent << "%, "
<< "level: " << stats.max_rec_level;
}
diff_samples = stats.play_samples - last_stats_.play_samples;
rate = diff_samples / (static_cast<float>(time_since_last) / 1000.0);
abs_diff_rate_in_percent = 0;
if (play_sample_rate > 0 && rate > 0) {
abs_diff_rate_in_percent = static_cast<uint32_t>(
0.5f +
((100.0f * std::abs(rate - play_sample_rate)) / play_sample_rate));
RTC_HISTOGRAM_PERCENTAGE("WebRTC.Audio.PlayoutSampleRateOffsetInPercent",
abs_diff_rate_in_percent);
RTC_LOG(INFO) << "[PLAY: " << time_since_last << "msec, "
<< play_sample_rate / 1000 << "kHz] callbacks: "
<< stats.play_callbacks - last_stats_.play_callbacks << ", "
<< "samples: " << diff_samples << ", "
<< "rate: " << static_cast<int>(rate + 0.5) << ", "
<< "rate diff: " << abs_diff_rate_in_percent << "%, "
<< "level: " << stats.max_play_level;
}
}
last_stats_ = stats;
int64_t time_to_wait_ms = next_callback_time - rtc::TimeMillis();
RTC_DCHECK_GT(time_to_wait_ms, 0) << "Invalid timer interval";
// Keep posting new (delayed) tasks until state is changed to kLogStop.
task_queue_.PostDelayedTask(rtc::Bind(&AudioDeviceBuffer::LogStats, this,
AudioDeviceBuffer::LOG_ACTIVE),
time_to_wait_ms);
}
void AudioDeviceBuffer::ResetRecStats() {
RTC_DCHECK_RUN_ON(&task_queue_);
last_stats_.ResetRecStats();
rtc::CritScope cs(&lock_);
stats_.ResetRecStats();
}
void AudioDeviceBuffer::ResetPlayStats() {
RTC_DCHECK_RUN_ON(&task_queue_);
last_stats_.ResetPlayStats();
rtc::CritScope cs(&lock_);
stats_.ResetPlayStats();
}
void AudioDeviceBuffer::UpdateRecStats(int16_t max_abs,
size_t samples_per_channel) {
rtc::CritScope cs(&lock_);
++stats_.rec_callbacks;
stats_.rec_samples += samples_per_channel;
if (max_abs > stats_.max_rec_level) {
stats_.max_rec_level = max_abs;
}
}
void AudioDeviceBuffer::UpdatePlayStats(int16_t max_abs,
size_t samples_per_channel) {
rtc::CritScope cs(&lock_);
++stats_.play_callbacks;
stats_.play_samples += samples_per_channel;
if (max_abs > stats_.max_play_level) {
stats_.max_play_level = max_abs;
}
}
} // namespace webrtc