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// 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 "content/browser/speech/speech_recognizer_impl.h"
#include "base/bind.h"
#include "base/time.h"
#include "content/browser/browser_main_loop.h"
#include "content/public/browser/speech_recognizer_delegate.h"
#include "content/public/browser/browser_thread.h"
#include "content/public/common/speech_recognition_result.h"
#include "net/url_request/url_request_context_getter.h"
using content::BrowserMainLoop;
using content::BrowserThread;
using content::SpeechRecognizer;
using content::SpeechRecognizerDelegate;
using media::AudioInputController;
using std::string;
namespace {
// The following constants are related to the volume level indicator shown in
// the UI for recorded audio.
// Multiplier used when new volume is greater than previous level.
const float kUpSmoothingFactor = 1.0f;
// Multiplier used when new volume is lesser than previous level.
const float kDownSmoothingFactor = 0.7f;
// RMS dB value of a maximum (unclipped) sine wave for int16 samples.
const float kAudioMeterMaxDb = 90.31f;
// This value corresponds to RMS dB for int16 with 6 most-significant-bits = 0.
// Values lower than this will display as empty level-meter.
const float kAudioMeterMinDb = 30.0f;
const float kAudioMeterDbRange = kAudioMeterMaxDb - kAudioMeterMinDb;
// Maximum level to draw to display unclipped meter. (1.0f displays clipping.)
const float kAudioMeterRangeMaxUnclipped = 47.0f / 48.0f;
// Returns true if more than 5% of the samples are at min or max value.
bool Clipping(const int16* samples, int num_samples) {
int clipping_samples = 0;
const int kThreshold = num_samples / 20;
for (int i = 0; i < num_samples; ++i) {
if (samples[i] <= -32767 || samples[i] >= 32767) {
if (++clipping_samples > kThreshold)
return true;
}
}
return false;
}
} // namespace
SpeechRecognizer* SpeechRecognizer::Create(
SpeechRecognizerDelegate* delegate,
int caller_id,
const std::string& language,
const std::string& grammar,
net::URLRequestContextGetter* context_getter,
bool filter_profanities,
const std::string& hardware_info,
const std::string& origin_url) {
return new speech::SpeechRecognizerImpl(
delegate, caller_id, language, grammar, context_getter,
filter_profanities, hardware_info, origin_url);
}
namespace speech {
const int SpeechRecognizerImpl::kAudioSampleRate = 16000;
const int SpeechRecognizerImpl::kAudioPacketIntervalMs = 100;
const ChannelLayout SpeechRecognizerImpl::kChannelLayout = CHANNEL_LAYOUT_MONO;
const int SpeechRecognizerImpl::kNumBitsPerAudioSample = 16;
const int SpeechRecognizerImpl::kNoSpeechTimeoutSec = 8;
const int SpeechRecognizerImpl::kEndpointerEstimationTimeMs = 300;
SpeechRecognizerImpl::SpeechRecognizerImpl(
SpeechRecognizerDelegate* delegate,
int caller_id,
const std::string& language,
const std::string& grammar,
net::URLRequestContextGetter* context_getter,
bool filter_profanities,
const std::string& hardware_info,
const std::string& origin_url)
: delegate_(delegate),
caller_id_(caller_id),
language_(language),
grammar_(grammar),
filter_profanities_(filter_profanities),
hardware_info_(hardware_info),
origin_url_(origin_url),
context_getter_(context_getter),
codec_(AudioEncoder::CODEC_FLAC),
encoder_(NULL),
endpointer_(kAudioSampleRate),
num_samples_recorded_(0),
audio_level_(0.0f),
audio_manager_(NULL) {
endpointer_.set_speech_input_complete_silence_length(
base::Time::kMicrosecondsPerSecond / 2);
endpointer_.set_long_speech_input_complete_silence_length(
base::Time::kMicrosecondsPerSecond);
endpointer_.set_long_speech_length(3 * base::Time::kMicrosecondsPerSecond);
endpointer_.StartSession();
}
SpeechRecognizerImpl::~SpeechRecognizerImpl() {
// Recording should have stopped earlier due to the endpointer or
// |StopRecording| being called.
DCHECK(!audio_controller_.get());
DCHECK(!request_.get() || !request_->HasPendingRequest());
DCHECK(!encoder_.get());
endpointer_.EndSession();
}
bool SpeechRecognizerImpl::StartRecording() {
DCHECK(BrowserThread::CurrentlyOn(BrowserThread::IO));
DCHECK(!audio_controller_.get());
DCHECK(!request_.get() || !request_->HasPendingRequest());
DCHECK(!encoder_.get());
// The endpointer needs to estimate the environment/background noise before
// starting to treat the audio as user input. In |HandleOnData| we wait until
// such time has passed before switching to user input mode.
endpointer_.SetEnvironmentEstimationMode();
encoder_.reset(AudioEncoder::Create(codec_, kAudioSampleRate,
kNumBitsPerAudioSample));
int samples_per_packet = (kAudioSampleRate * kAudioPacketIntervalMs) / 1000;
AudioParameters params(AudioParameters::AUDIO_PCM_LINEAR, kChannelLayout,
kAudioSampleRate, kNumBitsPerAudioSample,
samples_per_packet);
audio_controller_ = AudioInputController::Create(
audio_manager_ ? audio_manager_ : BrowserMainLoop::GetAudioManager(),
this, params);
DCHECK(audio_controller_.get());
VLOG(1) << "SpeechRecognizer starting record.";
num_samples_recorded_ = 0;
audio_controller_->Record();
return true;
}
void SpeechRecognizerImpl::CancelRecognition() {
DCHECK(BrowserThread::CurrentlyOn(BrowserThread::IO));
DCHECK(audio_controller_.get() || request_.get());
// Stop recording if required.
if (audio_controller_.get()) {
CloseAudioControllerSynchronously();
}
VLOG(1) << "SpeechRecognizer canceling recognition.";
encoder_.reset();
request_.reset();
}
void SpeechRecognizerImpl::StopRecording() {
DCHECK(BrowserThread::CurrentlyOn(BrowserThread::IO));
// If audio recording has already stopped and we are in recognition phase,
// silently ignore any more calls to stop recording.
if (!audio_controller_.get())
return;
CloseAudioControllerSynchronously();
delegate_->DidStopReceivingSpeech(caller_id_);
delegate_->DidCompleteRecording(caller_id_);
// UploadAudioChunk requires a non-empty final buffer. So we encode a packet
// of silence in case encoder had no data already.
std::vector<short> samples((kAudioSampleRate * kAudioPacketIntervalMs) /
1000);
encoder_->Encode(&samples[0], samples.size());
encoder_->Flush();
string encoded_data;
encoder_->GetEncodedDataAndClear(&encoded_data);
DCHECK(!encoded_data.empty());
encoder_.reset();
// If we haven't got any audio yet end the recognition sequence here.
if (request_ == NULL) {
// Guard against the delegate freeing us until we finish our job.
scoped_refptr<SpeechRecognizerImpl> me(this);
delegate_->DidCompleteRecognition(caller_id_);
} else {
request_->UploadAudioChunk(encoded_data, true /* is_last_chunk */);
}
}
// Invoked in the audio thread.
void SpeechRecognizerImpl::OnError(AudioInputController* controller,
int error_code) {
BrowserThread::PostTask(BrowserThread::IO, FROM_HERE,
base::Bind(&SpeechRecognizerImpl::HandleOnError,
this, error_code));
}
void SpeechRecognizerImpl::HandleOnError(int error_code) {
LOG(WARNING) << "SpeechRecognizer::HandleOnError, code=" << error_code;
// Check if we are still recording before canceling recognition, as
// recording might have been stopped after this error was posted to the queue
// by |OnError|.
if (!audio_controller_.get())
return;
InformErrorAndCancelRecognition(content::SPEECH_RECOGNITION_ERROR_AUDIO);
}
void SpeechRecognizerImpl::OnData(AudioInputController* controller,
const uint8* data, uint32 size) {
if (size == 0) // This could happen when recording stops and is normal.
return;
string* str_data = new string(reinterpret_cast<const char*>(data), size);
BrowserThread::PostTask(BrowserThread::IO, FROM_HERE,
base::Bind(&SpeechRecognizerImpl::HandleOnData,
this, str_data));
}
void SpeechRecognizerImpl::HandleOnData(string* data) {
// Check if we are still recording and if not discard this buffer, as
// recording might have been stopped after this buffer was posted to the queue
// by |OnData|.
if (!audio_controller_.get()) {
delete data;
return;
}
bool speech_was_heard_before_packet = endpointer_.DidStartReceivingSpeech();
const short* samples = reinterpret_cast<const short*>(data->data());
DCHECK_EQ((data->length() % sizeof(short)), 0U);
int num_samples = data->length() / sizeof(short);
encoder_->Encode(samples, num_samples);
float rms;
endpointer_.ProcessAudio(samples, num_samples, &rms);
bool did_clip = Clipping(samples, num_samples);
delete data;
num_samples_recorded_ += num_samples;
if (request_ == NULL) {
// This was the first audio packet recorded, so start a request to the
// server to send the data and inform the delegate.
delegate_->DidStartReceivingAudio(caller_id_);
request_.reset(new SpeechRecognitionRequest(context_getter_.get(), this));
request_->Start(language_, grammar_, filter_profanities_,
hardware_info_, origin_url_, encoder_->mime_type());
}
string encoded_data;
encoder_->GetEncodedDataAndClear(&encoded_data);
DCHECK(!encoded_data.empty());
request_->UploadAudioChunk(encoded_data, false /* is_last_chunk */);
if (endpointer_.IsEstimatingEnvironment()) {
// Check if we have gathered enough audio for the endpointer to do
// environment estimation and should move on to detect speech/end of speech.
if (num_samples_recorded_ >= (kEndpointerEstimationTimeMs *
kAudioSampleRate) / 1000) {
endpointer_.SetUserInputMode();
delegate_->DidCompleteEnvironmentEstimation(caller_id_);
}
return; // No more processing since we are still estimating environment.
}
// Check if we have waited too long without hearing any speech.
bool speech_was_heard_after_packet = endpointer_.DidStartReceivingSpeech();
if (!speech_was_heard_after_packet &&
num_samples_recorded_ >= kNoSpeechTimeoutSec * kAudioSampleRate) {
InformErrorAndCancelRecognition(
content::SPEECH_RECOGNITION_ERROR_NO_SPEECH);
return;
}
if (!speech_was_heard_before_packet && speech_was_heard_after_packet)
delegate_->DidStartReceivingSpeech(caller_id_);
// Calculate the input volume to display in the UI, smoothing towards the
// new level.
float level = (rms - kAudioMeterMinDb) /
(kAudioMeterDbRange / kAudioMeterRangeMaxUnclipped);
level = std::min(std::max(0.0f, level), kAudioMeterRangeMaxUnclipped);
if (level > audio_level_) {
audio_level_ += (level - audio_level_) * kUpSmoothingFactor;
} else {
audio_level_ += (level - audio_level_) * kDownSmoothingFactor;
}
float noise_level = (endpointer_.NoiseLevelDb() - kAudioMeterMinDb) /
(kAudioMeterDbRange / kAudioMeterRangeMaxUnclipped);
noise_level = std::min(std::max(0.0f, noise_level),
kAudioMeterRangeMaxUnclipped);
delegate_->SetInputVolume(caller_id_, did_clip ? 1.0f : audio_level_,
noise_level);
if (endpointer_.speech_input_complete())
StopRecording();
}
void SpeechRecognizerImpl::SetRecognitionResult(
const content::SpeechRecognitionResult& result) {
if (result.error != content::SPEECH_RECOGNITION_ERROR_NONE) {
InformErrorAndCancelRecognition(result.error);
return;
}
// Guard against the delegate freeing us until we finish our job.
scoped_refptr<SpeechRecognizerImpl> me(this);
delegate_->SetRecognitionResult(caller_id_, result);
delegate_->DidCompleteRecognition(caller_id_);
}
void SpeechRecognizerImpl::InformErrorAndCancelRecognition(
content::SpeechRecognitionErrorCode error) {
DCHECK_NE(error, content::SPEECH_RECOGNITION_ERROR_NONE);
CancelRecognition();
// Guard against the delegate freeing us until we finish our job.
scoped_refptr<SpeechRecognizerImpl> me(this);
delegate_->OnRecognizerError(caller_id_, error);
}
void SpeechRecognizerImpl::CloseAudioControllerSynchronously() {
VLOG(1) << "SpeechRecognizer stopping record.";
// TODO(satish): investigate the possibility to utilize the closure
// and switch to async. version of this method. Compare with how
// it's done in e.g. the AudioRendererHost.
base::WaitableEvent closed_event(true, false);
audio_controller_->Close(base::Bind(&base::WaitableEvent::Signal,
base::Unretained(&closed_event)));
closed_event.Wait();
audio_controller_ = NULL; // Releases the ref ptr.
}
void SpeechRecognizerImpl::SetAudioManagerForTesting(
AudioManager* audio_manager) {
audio_manager_ = audio_manager;
}
} // namespace speech
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