blob: 44ecb29d56ece7657e51c95ed4b051c20bfce6ee [file] [log] [blame]
// Copyright (c) 2012 The Chromium Authors. All rights reserved.
// Use of this source code is governed by a BSD-style license that can be
// found in the LICENSE file.
#include "media/audio/audio_input_controller.h"
#include "base/bind.h"
#include "base/strings/string_number_conversions.h"
#include "base/strings/stringprintf.h"
#include "base/threading/thread_restrictions.h"
#include "base/time/time.h"
#include "media/audio/audio_parameters.h"
#include "media/base/scoped_histogram_timer.h"
#include "media/base/user_input_monitor.h"
using base::TimeDelta;
namespace {
const int kMaxInputChannels = 3;
// TODO(henrika): remove usage of timers and add support for proper
// notification of when the input device is removed. This was originally added
// to resolve http://crbug.com/79936 for Windows platforms. This then caused
// breakage (very hard to repro bugs!) on other platforms: See
// http://crbug.com/226327 and http://crbug.com/230972.
// See also that the timer has been disabled on Mac now due to
// crbug.com/357501.
const int kTimerResetIntervalSeconds = 1;
// We have received reports that the timer can be too trigger happy on some
// Mac devices and the initial timer interval has therefore been increased
// from 1 second to 5 seconds.
const int kTimerInitialIntervalSeconds = 5;
#if defined(AUDIO_POWER_MONITORING)
// Time in seconds between two successive measurements of audio power levels.
const int kPowerMonitorLogIntervalSeconds = 15;
// A warning will be logged when the microphone audio volume is below this
// threshold.
const int kLowLevelMicrophoneLevelPercent = 10;
// Logs if the user has enabled the microphone mute or not. This is normally
// done by marking a checkbox in an audio-settings UI which is unique for each
// platform. Elements in this enum should not be added, deleted or rearranged.
enum MicrophoneMuteResult {
MICROPHONE_IS_MUTED = 0,
MICROPHONE_IS_NOT_MUTED = 1,
MICROPHONE_MUTE_MAX = MICROPHONE_IS_NOT_MUTED
};
void LogMicrophoneMuteResult(MicrophoneMuteResult result) {
UMA_HISTOGRAM_ENUMERATION("Media.MicrophoneMuted",
result,
MICROPHONE_MUTE_MAX + 1);
}
// Helper method which calculates the average power of an audio bus. Unit is in
// dBFS, where 0 dBFS corresponds to all channels and samples equal to 1.0.
float AveragePower(const media::AudioBus& buffer) {
const int frames = buffer.frames();
const int channels = buffer.channels();
if (frames <= 0 || channels <= 0)
return 0.0f;
// Scan all channels and accumulate the sum of squares for all samples.
float sum_power = 0.0f;
for (int ch = 0; ch < channels; ++ch) {
const float* channel_data = buffer.channel(ch);
for (int i = 0; i < frames; i++) {
const float sample = channel_data[i];
sum_power += sample * sample;
}
}
// Update accumulated average results, with clamping for sanity.
const float average_power =
std::max(0.0f, std::min(1.0f, sum_power / (frames * channels)));
// Convert average power level to dBFS units, and pin it down to zero if it
// is insignificantly small.
const float kInsignificantPower = 1.0e-10f; // -100 dBFS
const float power_dbfs = average_power < kInsignificantPower ?
-std::numeric_limits<float>::infinity() : 10.0f * log10f(average_power);
return power_dbfs;
}
#endif // AUDIO_POWER_MONITORING
}
// Used to log the result of capture startup.
// This was previously logged as a boolean with only the no callback and OK
// options. The enum order is kept to ensure backwards compatibility.
// Elements in this enum should not be deleted or rearranged; the only
// permitted operation is to add new elements before CAPTURE_STARTUP_RESULT_MAX
// and update CAPTURE_STARTUP_RESULT_MAX.
enum CaptureStartupResult {
CAPTURE_STARTUP_NO_DATA_CALLBACK = 0,
CAPTURE_STARTUP_OK = 1,
CAPTURE_STARTUP_CREATE_STREAM_FAILED = 2,
CAPTURE_STARTUP_OPEN_STREAM_FAILED = 3,
CAPTURE_STARTUP_RESULT_MAX = CAPTURE_STARTUP_OPEN_STREAM_FAILED
};
void LogCaptureStartupResult(CaptureStartupResult result) {
UMA_HISTOGRAM_ENUMERATION("Media.AudioInputControllerCaptureStartupSuccess",
result,
CAPTURE_STARTUP_RESULT_MAX + 1);
}
namespace media {
// static
AudioInputController::Factory* AudioInputController::factory_ = NULL;
AudioInputController::AudioInputController(EventHandler* handler,
SyncWriter* sync_writer,
UserInputMonitor* user_input_monitor,
const bool agc_is_enabled)
: creator_task_runner_(base::MessageLoopProxy::current()),
handler_(handler),
stream_(NULL),
data_is_active_(false),
state_(CLOSED),
sync_writer_(sync_writer),
max_volume_(0.0),
user_input_monitor_(user_input_monitor),
agc_is_enabled_(agc_is_enabled),
#if defined(AUDIO_POWER_MONITORING)
power_measurement_is_enabled_(false),
log_silence_state_(false),
silence_state_(SILENCE_STATE_NO_MEASUREMENT),
#endif
prev_key_down_count_(0) {
DCHECK(creator_task_runner_.get());
}
AudioInputController::~AudioInputController() {
DCHECK_EQ(state_, CLOSED);
}
// static
scoped_refptr<AudioInputController> AudioInputController::Create(
AudioManager* audio_manager,
EventHandler* event_handler,
const AudioParameters& params,
const std::string& device_id,
UserInputMonitor* user_input_monitor) {
DCHECK(audio_manager);
if (!params.IsValid() || (params.channels() > kMaxInputChannels))
return NULL;
if (factory_) {
return factory_->Create(
audio_manager, event_handler, params, user_input_monitor);
}
scoped_refptr<AudioInputController> controller(
new AudioInputController(event_handler, NULL, user_input_monitor, false));
controller->task_runner_ = audio_manager->GetTaskRunner();
// Create and open a new audio input stream from the existing
// audio-device thread.
if (!controller->task_runner_->PostTask(
FROM_HERE,
base::Bind(&AudioInputController::DoCreate,
controller,
base::Unretained(audio_manager),
params,
device_id))) {
controller = NULL;
}
return controller;
}
// static
scoped_refptr<AudioInputController> AudioInputController::CreateLowLatency(
AudioManager* audio_manager,
EventHandler* event_handler,
const AudioParameters& params,
const std::string& device_id,
SyncWriter* sync_writer,
UserInputMonitor* user_input_monitor,
const bool agc_is_enabled) {
DCHECK(audio_manager);
DCHECK(sync_writer);
if (!params.IsValid() || (params.channels() > kMaxInputChannels))
return NULL;
// Create the AudioInputController object and ensure that it runs on
// the audio-manager thread.
scoped_refptr<AudioInputController> controller(new AudioInputController(
event_handler, sync_writer, user_input_monitor, agc_is_enabled));
controller->task_runner_ = audio_manager->GetTaskRunner();
// Create and open a new audio input stream from the existing
// audio-device thread. Use the provided audio-input device.
if (!controller->task_runner_->PostTask(
FROM_HERE,
base::Bind(&AudioInputController::DoCreateForLowLatency,
controller,
base::Unretained(audio_manager),
params,
device_id))) {
controller = NULL;
}
return controller;
}
// static
scoped_refptr<AudioInputController> AudioInputController::CreateForStream(
const scoped_refptr<base::SingleThreadTaskRunner>& task_runner,
EventHandler* event_handler,
AudioInputStream* stream,
SyncWriter* sync_writer,
UserInputMonitor* user_input_monitor) {
DCHECK(sync_writer);
DCHECK(stream);
// Create the AudioInputController object and ensure that it runs on
// the audio-manager thread.
scoped_refptr<AudioInputController> controller(new AudioInputController(
event_handler, sync_writer, user_input_monitor, false));
controller->task_runner_ = task_runner;
// TODO(miu): See TODO at top of file. Until that's resolved, we need to
// disable the error auto-detection here (since the audio mirroring
// implementation will reliably report error and close events). Note, of
// course, that we're assuming CreateForStream() has been called for the audio
// mirroring use case only.
if (!controller->task_runner_->PostTask(
FROM_HERE,
base::Bind(&AudioInputController::DoCreateForStream,
controller,
stream))) {
controller = NULL;
}
return controller;
}
void AudioInputController::Record() {
task_runner_->PostTask(FROM_HERE, base::Bind(
&AudioInputController::DoRecord, this));
}
void AudioInputController::Close(const base::Closure& closed_task) {
DCHECK(!closed_task.is_null());
DCHECK(creator_task_runner_->BelongsToCurrentThread());
task_runner_->PostTaskAndReply(
FROM_HERE, base::Bind(&AudioInputController::DoClose, this), closed_task);
}
void AudioInputController::SetVolume(double volume) {
task_runner_->PostTask(FROM_HERE, base::Bind(
&AudioInputController::DoSetVolume, this, volume));
}
void AudioInputController::DoCreate(AudioManager* audio_manager,
const AudioParameters& params,
const std::string& device_id) {
DCHECK(task_runner_->BelongsToCurrentThread());
SCOPED_UMA_HISTOGRAM_TIMER("Media.AudioInputController.CreateTime");
if (handler_)
handler_->OnLog(this, "AIC::DoCreate");
#if defined(AUDIO_POWER_MONITORING)
// Disable power monitoring for streams that run without AGC enabled to
// avoid adding logs and UMA for non-WebRTC clients.
power_measurement_is_enabled_ = agc_is_enabled_;
last_audio_level_log_time_ = base::TimeTicks::Now();
silence_state_ = SILENCE_STATE_NO_MEASUREMENT;
#endif
// TODO(miu): See TODO at top of file. Until that's resolved, assume all
// platform audio input requires the |no_data_timer_| be used to auto-detect
// errors. In reality, probably only Windows needs to be treated as
// unreliable here.
DoCreateForStream(audio_manager->MakeAudioInputStream(params, device_id));
}
void AudioInputController::DoCreateForLowLatency(AudioManager* audio_manager,
const AudioParameters& params,
const std::string& device_id) {
DCHECK(task_runner_->BelongsToCurrentThread());
#if defined(AUDIO_POWER_MONITORING)
// We only log silence state UMA stats for low latency mode and if we use a
// real device.
if (params.format() != AudioParameters::AUDIO_FAKE)
log_silence_state_ = true;
#endif
low_latency_create_time_ = base::TimeTicks::Now();
DoCreate(audio_manager, params, device_id);
}
void AudioInputController::DoCreateForStream(
AudioInputStream* stream_to_control) {
DCHECK(task_runner_->BelongsToCurrentThread());
DCHECK(!stream_);
stream_ = stream_to_control;
if (!stream_) {
if (handler_)
handler_->OnError(this, STREAM_CREATE_ERROR);
LogCaptureStartupResult(CAPTURE_STARTUP_CREATE_STREAM_FAILED);
return;
}
if (stream_ && !stream_->Open()) {
stream_->Close();
stream_ = NULL;
if (handler_)
handler_->OnError(this, STREAM_OPEN_ERROR);
LogCaptureStartupResult(CAPTURE_STARTUP_OPEN_STREAM_FAILED);
return;
}
DCHECK(!no_data_timer_.get());
// Set AGC state using mode in |agc_is_enabled_| which can only be enabled in
// CreateLowLatency().
stream_->SetAutomaticGainControl(agc_is_enabled_);
// Create the data timer which will call FirstCheckForNoData(). The timer
// is started in DoRecord() and restarted in each DoCheckForNoData()
// callback.
// The timer is enabled for logging purposes. The NO_DATA_ERROR triggered
// from the timer must be ignored by the EventHandler.
// TODO(henrika): remove usage of timer when it has been verified on Canary
// that we are safe doing so. Goal is to get rid of |no_data_timer_| and
// everything that is tied to it. crbug.com/357569.
no_data_timer_.reset(new base::Timer(
FROM_HERE, base::TimeDelta::FromSeconds(kTimerInitialIntervalSeconds),
base::Bind(&AudioInputController::FirstCheckForNoData,
base::Unretained(this)), false));
state_ = CREATED;
if (handler_)
handler_->OnCreated(this);
if (user_input_monitor_) {
user_input_monitor_->EnableKeyPressMonitoring();
prev_key_down_count_ = user_input_monitor_->GetKeyPressCount();
}
}
void AudioInputController::DoRecord() {
DCHECK(task_runner_->BelongsToCurrentThread());
SCOPED_UMA_HISTOGRAM_TIMER("Media.AudioInputController.RecordTime");
if (state_ != CREATED)
return;
{
base::AutoLock auto_lock(lock_);
state_ = RECORDING;
}
if (handler_)
handler_->OnLog(this, "AIC::DoRecord");
if (no_data_timer_) {
// Start the data timer. Once |kTimerResetIntervalSeconds| have passed,
// a callback to FirstCheckForNoData() is made.
no_data_timer_->Reset();
}
stream_->Start(this);
if (handler_)
handler_->OnRecording(this);
}
void AudioInputController::DoClose() {
DCHECK(task_runner_->BelongsToCurrentThread());
SCOPED_UMA_HISTOGRAM_TIMER("Media.AudioInputController.CloseTime");
if (state_ == CLOSED)
return;
// If this is a low-latency stream, log the total duration (since DoCreate)
// and add it to a UMA histogram.
if (!low_latency_create_time_.is_null()) {
base::TimeDelta duration =
base::TimeTicks::Now() - low_latency_create_time_;
UMA_HISTOGRAM_LONG_TIMES("Media.InputStreamDuration", duration);
if (handler_) {
std::string log_string =
base::StringPrintf("AIC::DoClose: stream duration=");
log_string += base::Int64ToString(duration.InSeconds());
log_string += " seconds";
handler_->OnLog(this, log_string);
}
}
// Delete the timer on the same thread that created it.
no_data_timer_.reset();
DoStopCloseAndClearStream();
SetDataIsActive(false);
if (SharedMemoryAndSyncSocketMode())
sync_writer_->Close();
if (user_input_monitor_)
user_input_monitor_->DisableKeyPressMonitoring();
#if defined(AUDIO_POWER_MONITORING)
// Send UMA stats if enabled.
if (log_silence_state_)
LogSilenceState(silence_state_);
log_silence_state_ = false;
#endif
state_ = CLOSED;
}
void AudioInputController::DoReportError() {
DCHECK(task_runner_->BelongsToCurrentThread());
if (handler_)
handler_->OnError(this, STREAM_ERROR);
}
void AudioInputController::DoSetVolume(double volume) {
DCHECK(task_runner_->BelongsToCurrentThread());
DCHECK_GE(volume, 0);
DCHECK_LE(volume, 1.0);
if (state_ != CREATED && state_ != RECORDING)
return;
// Only ask for the maximum volume at first call and use cached value
// for remaining function calls.
if (!max_volume_) {
max_volume_ = stream_->GetMaxVolume();
}
if (max_volume_ == 0.0) {
DLOG(WARNING) << "Failed to access input volume control";
return;
}
// Set the stream volume and scale to a range matched to the platform.
stream_->SetVolume(max_volume_ * volume);
}
void AudioInputController::FirstCheckForNoData() {
DCHECK(task_runner_->BelongsToCurrentThread());
LogCaptureStartupResult(GetDataIsActive() ?
CAPTURE_STARTUP_OK :
CAPTURE_STARTUP_NO_DATA_CALLBACK);
if (handler_) {
handler_->OnLog(this, GetDataIsActive() ?
"AIC::FirstCheckForNoData => data is active" :
"AIC::FirstCheckForNoData => data is NOT active");
}
DoCheckForNoData();
}
void AudioInputController::DoCheckForNoData() {
DCHECK(task_runner_->BelongsToCurrentThread());
if (!GetDataIsActive()) {
// The data-is-active marker will be false only if it has been more than
// one second since a data packet was recorded. This can happen if a
// capture device has been removed or disabled.
if (handler_)
handler_->OnError(this, NO_DATA_ERROR);
}
// Mark data as non-active. The flag will be re-enabled in OnData() each
// time a data packet is received. Hence, under normal conditions, the
// flag will only be disabled during a very short period.
SetDataIsActive(false);
// Restart the timer to ensure that we check the flag again in
// |kTimerResetIntervalSeconds|.
no_data_timer_->Start(
FROM_HERE, base::TimeDelta::FromSeconds(kTimerResetIntervalSeconds),
base::Bind(&AudioInputController::DoCheckForNoData,
base::Unretained(this)));
}
void AudioInputController::OnData(AudioInputStream* stream,
const AudioBus* source,
uint32 hardware_delay_bytes,
double volume) {
// Mark data as active to ensure that the periodic calls to
// DoCheckForNoData() does not report an error to the event handler.
SetDataIsActive(true);
{
base::AutoLock auto_lock(lock_);
if (state_ != RECORDING)
return;
}
bool key_pressed = false;
if (user_input_monitor_) {
size_t current_count = user_input_monitor_->GetKeyPressCount();
key_pressed = current_count != prev_key_down_count_;
prev_key_down_count_ = current_count;
DVLOG_IF(6, key_pressed) << "Detected keypress.";
}
// Use SharedMemory and SyncSocket if the client has created a SyncWriter.
// Used by all low-latency clients except WebSpeech.
if (SharedMemoryAndSyncSocketMode()) {
sync_writer_->Write(source, volume, key_pressed);
sync_writer_->UpdateRecordedBytes(hardware_delay_bytes);
#if defined(AUDIO_POWER_MONITORING)
// Only do power-level measurements if DoCreate() has been called. It will
// ensure that logging will mainly be done for WebRTC and WebSpeech
// clients.
if (!power_measurement_is_enabled_)
return;
// Perform periodic audio (power) level measurements.
if ((base::TimeTicks::Now() - last_audio_level_log_time_).InSeconds() >
kPowerMonitorLogIntervalSeconds) {
// Calculate the average power of the signal, or the energy per sample.
const float average_power_dbfs = AveragePower(*source);
// Add current microphone volume to log and UMA histogram.
const int mic_volume_percent = static_cast<int>(100.0 * volume);
// Use event handler on the audio thread to relay a message to the ARIH
// in content which does the actual logging on the IO thread.
task_runner_->PostTask(FROM_HERE,
base::Bind(&AudioInputController::DoLogAudioLevels,
this,
average_power_dbfs,
mic_volume_percent));
last_audio_level_log_time_ = base::TimeTicks::Now();
}
#endif
return;
}
// TODO(henrika): Investigate if we can avoid the extra copy here.
// (see http://crbug.com/249316 for details). AFAIK, this scope is only
// active for WebSpeech clients.
scoped_ptr<AudioBus> audio_data =
AudioBus::Create(source->channels(), source->frames());
source->CopyTo(audio_data.get());
// Ownership of the audio buffer will be with the callback until it is run,
// when ownership is passed to the callback function.
task_runner_->PostTask(
FROM_HERE,
base::Bind(
&AudioInputController::DoOnData, this, base::Passed(&audio_data)));
}
void AudioInputController::DoOnData(scoped_ptr<AudioBus> data) {
DCHECK(task_runner_->BelongsToCurrentThread());
if (handler_)
handler_->OnData(this, data.get());
}
void AudioInputController::DoLogAudioLevels(float level_dbfs,
int microphone_volume_percent) {
#if defined(AUDIO_POWER_MONITORING)
DCHECK(task_runner_->BelongsToCurrentThread());
if (!handler_)
return;
// Detect if the user has enabled hardware mute by pressing the mute
// button in audio settings for the selected microphone.
const bool microphone_is_muted = stream_->IsMuted();
if (microphone_is_muted) {
LogMicrophoneMuteResult(MICROPHONE_IS_MUTED);
handler_->OnLog(this, "AIC::OnData: microphone is muted!");
// Return early if microphone is muted. No need to adding logs and UMA stats
// of audio levels if we know that the micropone is muted.
return;
}
LogMicrophoneMuteResult(MICROPHONE_IS_NOT_MUTED);
std::string log_string = base::StringPrintf(
"AIC::OnData: average audio level=%.2f dBFS", level_dbfs);
static const float kSilenceThresholdDBFS = -72.24719896f;
if (level_dbfs < kSilenceThresholdDBFS)
log_string += " <=> low audio input level!";
handler_->OnLog(this, log_string);
UpdateSilenceState(level_dbfs < kSilenceThresholdDBFS);
UMA_HISTOGRAM_PERCENTAGE("Media.MicrophoneVolume", microphone_volume_percent);
log_string = base::StringPrintf(
"AIC::OnData: microphone volume=%d%%", microphone_volume_percent);
if (microphone_volume_percent < kLowLevelMicrophoneLevelPercent)
log_string += " <=> low microphone level!";
handler_->OnLog(this, log_string);
#endif
}
void AudioInputController::OnError(AudioInputStream* stream) {
// Handle error on the audio-manager thread.
task_runner_->PostTask(FROM_HERE, base::Bind(
&AudioInputController::DoReportError, this));
}
void AudioInputController::DoStopCloseAndClearStream() {
DCHECK(task_runner_->BelongsToCurrentThread());
// Allow calling unconditionally and bail if we don't have a stream to close.
if (stream_ != NULL) {
stream_->Stop();
stream_->Close();
stream_ = NULL;
}
// The event handler should not be touched after the stream has been closed.
handler_ = NULL;
}
void AudioInputController::SetDataIsActive(bool enabled) {
base::subtle::Release_Store(&data_is_active_, enabled);
}
bool AudioInputController::GetDataIsActive() {
return (base::subtle::Acquire_Load(&data_is_active_) != false);
}
#if defined(AUDIO_POWER_MONITORING)
void AudioInputController::UpdateSilenceState(bool silence) {
if (silence) {
if (silence_state_ == SILENCE_STATE_NO_MEASUREMENT) {
silence_state_ = SILENCE_STATE_ONLY_SILENCE;
} else if (silence_state_ == SILENCE_STATE_ONLY_AUDIO) {
silence_state_ = SILENCE_STATE_AUDIO_AND_SILENCE;
} else {
DCHECK(silence_state_ == SILENCE_STATE_ONLY_SILENCE ||
silence_state_ == SILENCE_STATE_AUDIO_AND_SILENCE);
}
} else {
if (silence_state_ == SILENCE_STATE_NO_MEASUREMENT) {
silence_state_ = SILENCE_STATE_ONLY_AUDIO;
} else if (silence_state_ == SILENCE_STATE_ONLY_SILENCE) {
silence_state_ = SILENCE_STATE_AUDIO_AND_SILENCE;
} else {
DCHECK(silence_state_ == SILENCE_STATE_ONLY_AUDIO ||
silence_state_ == SILENCE_STATE_AUDIO_AND_SILENCE);
}
}
}
void AudioInputController::LogSilenceState(SilenceState value) {
UMA_HISTOGRAM_ENUMERATION("Media.AudioInputControllerSessionSilenceReport",
value,
SILENCE_STATE_MAX + 1);
}
#endif
} // namespace media