blob: 30e45646d9697f16ed7b9bb62dc9f3ac5f0936de [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_device.h"
#include <stdint.h>
#include <utility>
#include <vector>
#include "base/bind.h"
#include "base/callback_forward.h"
#include "base/format_macros.h"
#include "base/macros.h"
#include "base/memory/ptr_util.h"
#include "base/metrics/histogram_macros.h"
#include "base/strings/stringprintf.h"
#include "base/threading/thread_restrictions.h"
#include "base/trace_event/trace_event.h"
#include "build/build_config.h"
#include "media/audio/audio_manager_base.h"
#include "media/base/audio_bus.h"
namespace media {
namespace {
// The number of shared memory buffer segments indicated to browser process
// in order to avoid data overwriting. This number can be any positive number,
// dependent how fast the renderer process can pick up captured data from
// shared memory.
const int kRequestedSharedMemoryCount = 10;
// The number of seconds with missing callbacks before we report a capture
// error. The value is based on that the Mac audio implementation can defer
// start for 5 seconds when resuming after standby, and has a startup success
// check 5 seconds after actually starting, where stats is logged. We must allow
// enough time for this. See AUAudioInputStream::CheckInputStartupSuccess().
const int kMissingCallbacksTimeBeforeErrorSeconds = 12;
// The interval for checking missing callbacks.
const int kCheckMissingCallbacksIntervalSeconds = 5;
// How often AudioInputDevice::AudioThreadCallback informs that it has gotten
// data from the source.
const int kGotDataCallbackIntervalSeconds = 1;
base::ThreadPriority ThreadPriorityFromPurpose(
AudioInputDevice::Purpose purpose) {
switch (purpose) {
case AudioInputDevice::Purpose::kUserInput:
return base::ThreadPriority::REALTIME_AUDIO;
case AudioInputDevice::Purpose::kLoopback:
return base::ThreadPriority::NORMAL;
}
}
} // namespace
// Takes care of invoking the capture callback on the audio thread.
// An instance of this class is created for each capture stream in
// OnLowLatencyCreated().
class AudioInputDevice::AudioThreadCallback
: public AudioDeviceThread::Callback {
public:
AudioThreadCallback(const AudioParameters& audio_parameters,
base::ReadOnlySharedMemoryRegion shared_memory_region,
uint32_t total_segments,
bool enable_uma,
CaptureCallback* capture_callback,
base::RepeatingClosure got_data_callback);
~AudioThreadCallback() override;
void MapSharedMemory() override;
// Called whenever we receive notifications about pending data.
void Process(uint32_t pending_data) override;
private:
const bool enable_uma_;
base::ReadOnlySharedMemoryRegion shared_memory_region_;
base::ReadOnlySharedMemoryMapping shared_memory_mapping_;
const base::TimeTicks start_time_;
bool no_callbacks_received_;
size_t current_segment_id_;
uint32_t last_buffer_id_;
std::vector<std::unique_ptr<const media::AudioBus>> audio_buses_;
CaptureCallback* capture_callback_;
// Used for informing AudioInputDevice that we have gotten data, i.e. the
// stream is alive. |got_data_callback_| is run every
// |got_data_callback_interval_in_frames_| frames, calculated from
// kGotDataCallbackIntervalSeconds.
const int got_data_callback_interval_in_frames_;
int frames_since_last_got_data_callback_;
base::RepeatingClosure got_data_callback_;
DISALLOW_COPY_AND_ASSIGN(AudioThreadCallback);
};
AudioInputDevice::AudioInputDevice(std::unique_ptr<AudioInputIPC> ipc,
Purpose purpose)
: thread_priority_(ThreadPriorityFromPurpose(purpose)),
enable_uma_(purpose == AudioInputDevice::Purpose::kUserInput),
callback_(nullptr),
ipc_(std::move(ipc)),
state_(IDLE),
agc_is_enabled_(false) {
CHECK(ipc_);
// The correctness of the code depends on the relative values assigned in the
// State enum.
static_assert(IPC_CLOSED < IDLE, "invalid enum value assignment 0");
static_assert(IDLE < CREATING_STREAM, "invalid enum value assignment 1");
static_assert(CREATING_STREAM < RECORDING, "invalid enum value assignment 2");
}
void AudioInputDevice::Initialize(const AudioParameters& params,
CaptureCallback* callback) {
DCHECK_CALLED_ON_VALID_SEQUENCE(sequence_checker_);
DCHECK(params.IsValid());
DCHECK(!callback_);
audio_parameters_ = params;
callback_ = callback;
}
void AudioInputDevice::Start() {
DCHECK_CALLED_ON_VALID_SEQUENCE(sequence_checker_);
DCHECK(callback_) << "Initialize hasn't been called";
TRACE_EVENT0("audio", "AudioInputDevice::Start");
// Make sure we don't call Start() more than once.
if (state_ != IDLE)
return;
state_ = CREATING_STREAM;
ipc_->CreateStream(this, audio_parameters_, agc_is_enabled_,
kRequestedSharedMemoryCount);
}
void AudioInputDevice::Stop() {
DCHECK_CALLED_ON_VALID_SEQUENCE(sequence_checker_);
TRACE_EVENT0("audio", "AudioInputDevice::Stop");
if (enable_uma_) {
UMA_HISTOGRAM_BOOLEAN(
"Media.Audio.Capture.DetectedMissingCallbacks",
alive_checker_ ? alive_checker_->DetectedDead() : false);
UMA_HISTOGRAM_ENUMERATION("Media.Audio.Capture.StreamCallbackError2",
had_error_);
}
had_error_ = kNoError;
// Close the stream, if we haven't already.
if (state_ >= CREATING_STREAM) {
ipc_->CloseStream();
state_ = IDLE;
agc_is_enabled_ = false;
}
// We can run into an issue where Stop is called right after
// OnStreamCreated is called in cases where Start/Stop are called before we
// get the OnStreamCreated callback. To handle that corner case, we call
// audio_thread_.reset(). In most cases, the thread will already be stopped.
//
// |alive_checker_| must outlive |audio_callback_|.
base::ScopedAllowBlocking allow_blocking;
audio_thread_.reset();
audio_callback_.reset();
alive_checker_.reset();
}
void AudioInputDevice::SetVolume(double volume) {
DCHECK_CALLED_ON_VALID_SEQUENCE(sequence_checker_);
TRACE_EVENT1("audio", "AudioInputDevice::SetVolume", "volume", volume);
if (volume < 0 || volume > 1.0) {
DLOG(ERROR) << "Invalid volume value specified";
return;
}
if (state_ >= CREATING_STREAM)
ipc_->SetVolume(volume);
}
void AudioInputDevice::SetAutomaticGainControl(bool enabled) {
DCHECK_CALLED_ON_VALID_SEQUENCE(sequence_checker_);
TRACE_EVENT1("audio", "AudioInputDevice::SetAutomaticGainControl", "enabled",
enabled);
if (state_ >= CREATING_STREAM) {
DLOG(WARNING) << "The AGC state can not be modified after starting.";
return;
}
// We simply store the new AGC setting here. This value will be used when
// a new stream is initialized and by GetAutomaticGainControl().
agc_is_enabled_ = enabled;
}
void AudioInputDevice::SetOutputDeviceForAec(
const std::string& output_device_id) {
DCHECK_CALLED_ON_VALID_SEQUENCE(sequence_checker_);
TRACE_EVENT1("audio", "AudioInputDevice::SetOutputDeviceForAec",
"output_device_id", output_device_id);
output_device_id_for_aec_ = output_device_id;
if (state_ > CREATING_STREAM)
ipc_->SetOutputDeviceForAec(output_device_id);
}
void AudioInputDevice::OnStreamCreated(
base::ReadOnlySharedMemoryRegion shared_memory_region,
base::SyncSocket::Handle socket_handle,
bool initially_muted) {
DCHECK_CALLED_ON_VALID_SEQUENCE(sequence_checker_);
TRACE_EVENT0("audio", "AudioInputDevice::OnStreamCreated");
DCHECK(shared_memory_region.IsValid());
#if defined(OS_WIN)
DCHECK(socket_handle);
#else
DCHECK_GE(socket_handle, 0);
#endif
DCHECK_GT(shared_memory_region.GetSize(), 0u);
if (state_ != CREATING_STREAM)
return;
DCHECK(!audio_callback_);
DCHECK(!audio_thread_);
if (initially_muted)
callback_->OnCaptureMuted(true);
if (auto* controls = ipc_->GetProcessorControls())
callback_->OnCaptureProcessorCreated(controls);
if (output_device_id_for_aec_)
ipc_->SetOutputDeviceForAec(*output_device_id_for_aec_);
// Set up checker for detecting missing audio data. We pass a callback which
// holds a reference to this. |alive_checker_| is deleted in
// Stop() which we expect to always be called (see comment in
// destructor). Suspend/resume notifications are not supported on Linux and
// there's a risk of false positives when suspending. So on Linux we only detect
// missing audio data until the first audio buffer arrives. Note that there's
// also a risk of false positives if we are suspending when starting the stream
// here. See comments in AliveChecker and PowerObserverHelper for details and
// todos.
#if defined(OS_LINUX)
const bool stop_at_first_alive_notification = true;
const bool pause_check_during_suspend = false;
#else
const bool stop_at_first_alive_notification = false;
const bool pause_check_during_suspend = true;
#endif
alive_checker_ = std::make_unique<AliveChecker>(
base::Bind(&AudioInputDevice::DetectedDeadInputStream, this),
base::TimeDelta::FromSeconds(kCheckMissingCallbacksIntervalSeconds),
base::TimeDelta::FromSeconds(kMissingCallbacksTimeBeforeErrorSeconds),
stop_at_first_alive_notification, pause_check_during_suspend);
// Unretained is safe since |alive_checker_| outlives |audio_callback_|.
audio_callback_ = std::make_unique<AudioInputDevice::AudioThreadCallback>(
audio_parameters_, std::move(shared_memory_region),
kRequestedSharedMemoryCount, enable_uma_, callback_,
base::BindRepeating(&AliveChecker::NotifyAlive,
base::Unretained(alive_checker_.get())));
audio_thread_ =
std::make_unique<AudioDeviceThread>(audio_callback_.get(), socket_handle,
"AudioInputDevice", thread_priority_);
state_ = RECORDING;
ipc_->RecordStream();
// Start detecting missing audio data.
alive_checker_->Start();
}
void AudioInputDevice::OnError() {
DCHECK_CALLED_ON_VALID_SEQUENCE(sequence_checker_);
TRACE_EVENT0("audio", "AudioInputDevice::OnError");
// Do nothing if the stream has been closed.
if (state_ < CREATING_STREAM)
return;
if (state_ == CREATING_STREAM) {
// At this point, we haven't attempted to start the audio thread.
// Accessing the hardware might have failed or we may have reached
// the limit of the number of allowed concurrent streams.
// We must report the error to the |callback_| so that a potential
// audio source object will enter the correct state (e.g. 'ended' for
// a local audio source).
had_error_ = kErrorDuringCreation;
callback_->OnCaptureError(
"Maximum allowed input device limit reached or OS failure.");
} else {
// Don't dereference the callback object if the audio thread
// is stopped or stopping. That could mean that the callback
// object has been deleted.
// TODO(tommi): Add an explicit contract for clearing the callback
// object. Possibly require calling Initialize again or provide
// a callback object via Start() and clear it in Stop().
had_error_ = kErrorDuringCapture;
if (audio_thread_)
callback_->OnCaptureError("IPC delegate state error.");
}
}
void AudioInputDevice::OnMuted(bool is_muted) {
DCHECK_CALLED_ON_VALID_SEQUENCE(sequence_checker_);
TRACE_EVENT0("audio", "AudioInputDevice::OnMuted");
// Do nothing if the stream has been closed.
if (state_ < CREATING_STREAM)
return;
callback_->OnCaptureMuted(is_muted);
}
void AudioInputDevice::OnIPCClosed() {
DCHECK_CALLED_ON_VALID_SEQUENCE(sequence_checker_);
TRACE_EVENT0("audio", "AudioInputDevice::OnIPCClosed");
state_ = IPC_CLOSED;
ipc_.reset();
}
AudioInputDevice::~AudioInputDevice() {
#if DCHECK_IS_ON()
// Make sure we've stopped the stream properly before destructing |this|.
DCHECK_LE(state_, IDLE);
DCHECK(!audio_thread_);
DCHECK(!audio_callback_);
DCHECK(!alive_checker_);
#endif // DCHECK_IS_ON()
}
void AudioInputDevice::DetectedDeadInputStream() {
callback_->OnCaptureError("No audio received from audio capture device.");
}
// AudioInputDevice::AudioThreadCallback
AudioInputDevice::AudioThreadCallback::AudioThreadCallback(
const AudioParameters& audio_parameters,
base::ReadOnlySharedMemoryRegion shared_memory_region,
uint32_t total_segments,
bool enable_uma,
CaptureCallback* capture_callback,
base::RepeatingClosure got_data_callback_)
: AudioDeviceThread::Callback(
audio_parameters,
ComputeAudioInputBufferSize(audio_parameters, 1u),
total_segments),
enable_uma_(enable_uma),
shared_memory_region_(std::move(shared_memory_region)),
start_time_(base::TimeTicks::Now()),
no_callbacks_received_(true),
current_segment_id_(0u),
last_buffer_id_(UINT32_MAX),
capture_callback_(capture_callback),
got_data_callback_interval_in_frames_(kGotDataCallbackIntervalSeconds *
audio_parameters.sample_rate()),
frames_since_last_got_data_callback_(0),
got_data_callback_(std::move(got_data_callback_)) {
// CHECK that the shared memory is large enough. The memory allocated must
// be at least as large as expected.
CHECK_LE(memory_length_, shared_memory_region_.GetSize());
}
AudioInputDevice::AudioThreadCallback::~AudioThreadCallback() {
if (enable_uma_) {
UMA_HISTOGRAM_LONG_TIMES("Media.Audio.Capture.InputStreamDuration",
base::TimeTicks::Now() - start_time_);
}
}
void AudioInputDevice::AudioThreadCallback::MapSharedMemory() {
shared_memory_mapping_ = shared_memory_region_.MapAt(0, memory_length_);
// Create vector of audio buses by wrapping existing blocks of memory.
const uint8_t* ptr =
static_cast<const uint8_t*>(shared_memory_mapping_.memory());
for (uint32_t i = 0; i < total_segments_; ++i) {
const media::AudioInputBuffer* buffer =
reinterpret_cast<const media::AudioInputBuffer*>(ptr);
audio_buses_.push_back(
media::AudioBus::WrapReadOnlyMemory(audio_parameters_, buffer->audio));
ptr += segment_length_;
}
// Indicate that browser side capture initialization has succeeded and IPC
// channel initialized. This effectively completes the
// AudioCapturerSource::Start()' phase as far as the caller of that function
// is concerned.
capture_callback_->OnCaptureStarted();
}
void AudioInputDevice::AudioThreadCallback::Process(uint32_t pending_data) {
TRACE_EVENT_BEGIN0("audio", "AudioInputDevice::AudioThreadCallback::Process");
if (no_callbacks_received_) {
if (enable_uma_) {
UMA_HISTOGRAM_TIMES("Media.Audio.Render.InputDeviceStartTime",
base::TimeTicks::Now() - start_time_);
}
no_callbacks_received_ = false;
}
// The shared memory represents parameters, size of the data buffer and the
// actual data buffer containing audio data. Map the memory into this
// structure and parse out parameters and the data area.
const uint8_t* ptr =
static_cast<const uint8_t*>(shared_memory_mapping_.memory());
ptr += current_segment_id_ * segment_length_;
const AudioInputBuffer* buffer =
reinterpret_cast<const AudioInputBuffer*>(ptr);
// Usually this will be equal but in the case of low sample rate (e.g. 8kHz,
// the buffer may be bigger (on mac at least)).
DCHECK_GE(buffer->params.size,
segment_length_ - sizeof(AudioInputBufferParameters));
// Verify correct sequence.
if (buffer->params.id != last_buffer_id_ + 1) {
std::string message = base::StringPrintf(
"Incorrect buffer sequence. Expected = %u. Actual = %u.",
last_buffer_id_ + 1, buffer->params.id);
LOG(ERROR) << message;
capture_callback_->OnCaptureError(message);
}
if (current_segment_id_ != pending_data) {
std::string message = base::StringPrintf(
"Segment id not matching. Remote = %u. Local = %" PRIuS ".",
pending_data, current_segment_id_);
LOG(ERROR) << message;
capture_callback_->OnCaptureError(message);
}
last_buffer_id_ = buffer->params.id;
// Use pre-allocated audio bus wrapping existing block of shared memory.
const media::AudioBus* audio_bus = audio_buses_[current_segment_id_].get();
// Regularly inform that we have gotten data.
frames_since_last_got_data_callback_ += audio_bus->frames();
if (frames_since_last_got_data_callback_ >=
got_data_callback_interval_in_frames_) {
got_data_callback_.Run();
frames_since_last_got_data_callback_ = 0;
}
// Deliver captured data to the client in floating point format and update
// the audio delay measurement.
// TODO(olka, tommi): Take advantage of |capture_time| in the renderer.
const base::TimeTicks capture_time =
base::TimeTicks() +
base::TimeDelta::FromMicroseconds(buffer->params.capture_time_us);
const base::TimeTicks now_time = base::TimeTicks::Now();
DCHECK_GE(now_time, capture_time);
capture_callback_->Capture(audio_bus,
(now_time - capture_time).InMilliseconds(),
buffer->params.volume, buffer->params.key_pressed);
if (++current_segment_id_ >= total_segments_)
current_segment_id_ = 0u;
TRACE_EVENT_END2(
"audio", "AudioInputDevice::AudioThreadCallback::Process",
"capture_time (ms)", (capture_time - base::TimeTicks()).InMillisecondsF(),
"now_time (ms)", (now_time - base::TimeTicks()).InMillisecondsF());
}
} // namespace media