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Merge pull request #2425 from mozilla/expose-sample-rate-value

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Expose sample rate value in API and use it in in-tree consumers
This commit is contained in:
Reuben Morais 2019-10-11 09:39:31 +02:00 committed by GitHub
commit fcbebbe71a
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GPG Key ID: 4AEE18F83AFDEB23
14 changed files with 113 additions and 48 deletions
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8
examples/net_framework/DeepSpeechWPF/MainWindow.xaml.cs
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@ -202,15 +202,13 @@ namespace DeepSpeechWPF
{
_audioCapture.Device = _audioCaptureDevices[cbxAudioInputs.SelectedIndex];
}
InitilizeAudioCapture();
InitializeAudioCapture(_sttClient.GetModelSampleRate());
}
/// <summary>
/// Initializes the recorder and setup the native stream.
/// </summary>
private void InitilizeAudioCapture()
private void InitializeAudioCapture(int desiredSampleRate)
{
_audioCapture.Initialize();
_audioCapture.DataAvailable += _capture_DataAvailable;
@ -218,7 +216,7 @@ namespace DeepSpeechWPF
//create a source, that converts the data provided by the
//soundInSource to required by the deepspeech model
_convertedSource = _soundInSource
.ChangeSampleRate(16000) // sample rate
.ChangeSampleRate(desiredSampleRate) // sample rate
.ToSampleSource()
.ToWaveSource(16); //bits per sample

11
examples/nodejs_wav/index.js
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@ -11,6 +11,8 @@ let alphabetPath = './models/alphabet.txt';
let model = new DeepSpeech.Model(modelPath, alphabetPath, BEAM_WIDTH);
let desiredSampleRate = model.sampleRate();
const LM_ALPHA = 0.75;
const LM_BETA = 1.85;
let lmPath = './models/lm.binary';
@ -28,8 +30,8 @@ if (!Fs.existsSync(audioFile)) {
const buffer = Fs.readFileSync(audioFile);
const result = Wav.decode(buffer);
if (result.sampleRate < 16000) {
console.error('Warning: original sample rate (' + result.sampleRate + ') is lower than 16kHz. Up-sampling might produce erratic speech recognition.');
if (result.sampleRate < desiredSampleRate) {
console.error('Warning: original sample rate (' + result.sampleRate + ') is lower than ' + desiredSampleRate + 'Hz. Up-sampling might produce erratic speech recognition.');
}
function bufferToStream(buffer) {
@ -47,7 +49,7 @@ pipe(Sox({
},
output: {
bits: 16,
rate: 16000,
rate: desiredSampleRate,
channels: 1,
encoding: 'signed-integer',
endian: 'little',
@ -58,10 +60,9 @@ pipe(Sox({
pipe(audioStream);
audioStream.on('finish', () => {
let audioBuffer = audioStream.toBuffer();
const audioLength = (audioBuffer.length / 2) * ( 1 / 16000);
const audioLength = (audioBuffer.length / 2) * (1 / desiredSampleRate);
console.log('audio length', audioLength);
let result = model.stt(audioBuffer.slice(0, audioBuffer.length / 2));

18
native_client/client.cc
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@ -111,7 +111,7 @@ typedef struct {
} ds_audio_buffer;
ds_audio_buffer
GetAudioBuffer(const char* path)
GetAudioBuffer(const char* path, int desired_sample_rate)
{
ds_audio_buffer res = {0};
@ -121,7 +121,7 @@ GetAudioBuffer(const char* path)
// Resample/reformat the audio so we can pass it through the MFCC functions
sox_signalinfo_t target_signal = {
16000, // Rate
static_cast<sox_rate_t>(desired_sample_rate), // Rate
1, // Channels
16, // Precision
SOX_UNSPEC, // Length
@ -158,8 +158,10 @@ GetAudioBuffer(const char* path)
assert(output);
if ((int)input->signal.rate < 16000) {
fprintf(stderr, "Warning: original sample rate (%d) is lower than 16kHz. Up-sampling might produce erratic speech recognition.\n", (int)input->signal.rate);
if ((int)input->signal.rate < desired_sample_rate) {
fprintf(stderr, "Warning: original sample rate (%d) is lower than %dkHz. "
"Up-sampling might produce erratic speech recognition.\n",
desired_sample_rate, (int)input->signal.rate);
}
// Setup the effects chain to decode/resample
@ -205,7 +207,7 @@ GetAudioBuffer(const char* path)
#endif // NO_SOX
#ifdef NO_SOX
// FIXME: Hack and support only 16kHz mono 16-bits PCM
// FIXME: Hack and support only mono 16-bits PCM with standard SoX header
FILE* wave = fopen(path, "r");
size_t rv;
@ -224,12 +226,12 @@ GetAudioBuffer(const char* path)
assert(audio_format == 1); // 1 is PCM
assert(num_channels == 1); // MONO
assert(sample_rate == 16000); // 16000 Hz
assert(sample_rate == desired_sample_rate); // at desired sample rate
assert(bits_per_sample == 16); // 16 bits per sample
fprintf(stderr, "audio_format=%d\n", audio_format);
fprintf(stderr, "num_channels=%d\n", num_channels);
fprintf(stderr, "sample_rate=%d\n", sample_rate);
fprintf(stderr, "sample_rate=%d (desired=%d)\n", sample_rate, desired_sample_rate);
fprintf(stderr, "bits_per_sample=%d\n", bits_per_sample);
fseek(wave, 40, SEEK_SET); rv = fread(&res.buffer_size, 4, 1, wave);
@ -257,7 +259,7 @@ GetAudioBuffer(const char* path)
void
ProcessFile(ModelState* context, const char* path, bool show_times)
{
ds_audio_buffer audio = GetAudioBuffer(path);
ds_audio_buffer audio = GetAudioBuffer(path, DS_GetModelSampleRate(context));
// Pass audio to DeepSpeech
// We take half of buffer_size because buffer is a char* while

6
native_client/deepspeech.cc
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@ -292,6 +292,12 @@ DS_CreateModel(const char* aModelPath,
return DS_ERR_OK;
}
int
DS_GetModelSampleRate(ModelState* aCtx)
{
return aCtx->sample_rate_;
}
void
DS_FreeModel(ModelState* ctx)
{

10
native_client/deepspeech.h
View File

@ -92,6 +92,16 @@ int DS_CreateModel(const char* aModelPath,
unsigned int aBeamWidth,
ModelState** retval);
/**
* @brief Return the sample rate expected by a model.
*
* @param aCtx A ModelState pointer created with {@link DS_CreateModel}.
*
* @return Sample rate expected by the model for its input.
*/
DEEPSPEECH_EXPORT
int DS_GetModelSampleRate(ModelState* aCtx);
/**
* @brief Frees associated resources and destroys model object.
*/

9
native_client/dotnet/DeepSpeechClient/DeepSpeech.cs
View File

@ -64,6 +64,15 @@ namespace DeepSpeechClient
EvaluateResultCode(resultCode);
}
/// <summary>
/// Return the sample rate expected by the model.
/// </summary>
/// <returns>Sample rate.</returns>
public unsafe int GetModelSampleRate()
{
return NativeImp.DS_GetModelSampleRate(_modelStatePP);
}
/// <summary>
/// Evaluate the result code and will raise an exception if necessary.
/// </summary>

6
native_client/dotnet/DeepSpeechClient/Interfaces/IDeepSpeech.cs
View File

@ -24,6 +24,12 @@ namespace DeepSpeechClient.Interfaces
string aAlphabetConfigPath,
uint aBeamWidth);
/// <summary>
/// Return the sample rate expected by the model.
/// </summary>
/// <returns>Sample rate.</returns>
unsafe int GetModelSampleRate();
/// <summary>
/// Enable decoding using beam scoring with a KenLM language model.
/// </summary>

3
native_client/dotnet/DeepSpeechClient/NativeImp.cs
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@ -20,6 +20,9 @@ namespace DeepSpeechClient
uint aBeamWidth,
ref IntPtr** pint);
[DllImport("libdeepspeech.so", CallingConvention = CallingConvention.Cdecl)]
internal unsafe static extern int DS_GetModelSampleRate(IntPtr** aCtx);
[DllImport("libdeepspeech.so", CallingConvention = CallingConvention.Cdecl)]
internal static unsafe extern ErrorCodes DS_EnableDecoderWithLM(IntPtr** aCtx,
string aLMPath,

2
native_client/java/app/src/main/java/org/mozilla/deepspeech/DeepSpeechActivity.java
View File

@ -77,7 +77,7 @@ public class DeepSpeechActivity extends AppCompatActivity {
// tv_numChannels.setText("numChannels=" + (numChannels == 1 ? "MONO" : "!MONO"));
wave.seek(24); int sampleRate = this.readLEInt(wave);
assert (sampleRate == 16000); // 16000 Hz
assert (sampleRate == this._m.sampleRate()); // desired sample rate
// tv_sampleRate.setText("sampleRate=" + (sampleRate == 16000 ? "16kHz" : "!16kHz"));
wave.seek(34); char bitsPerSample = this.readLEChar(wave);

9
native_client/java/libdeepspeech/src/main/java/org/mozilla/deepspeech/libdeepspeech/DeepSpeechModel.java
View File

@ -32,6 +32,15 @@ public class DeepSpeechModel {
this._msp = impl.modelstatep_value(this._mspp);
}
/**
* @brief Return the sample rate expected by the model.
*
* @return Sample rate.
*/
public int sampleRate() {
return impl.GetModelSampleRate(this._msp);
}
/**
* @brief Frees associated resources and destroys model object.
*/

40
native_client/javascript/client.js
View File

@ -62,11 +62,29 @@ function metadataToString(metadata) {
return retval;
}
console.error('Loading model from file %s', args['model']);
const model_load_start = process.hrtime();
var model = new Ds.Model(args['model'], args['alphabet'], BEAM_WIDTH);
const model_load_end = process.hrtime(model_load_start);
console.error('Loaded model in %ds.', totalTime(model_load_end));
var desired_sample_rate = model.sampleRate();
if (args['lm'] && args['trie']) {
console.error('Loading language model from files %s %s', args['lm'], args['trie']);
const lm_load_start = process.hrtime();
model.enableDecoderWithLM(args['lm'], args['trie'], LM_ALPHA, LM_BETA);
const lm_load_end = process.hrtime(lm_load_start);
console.error('Loaded language model in %ds.', totalTime(lm_load_end));
}
const buffer = Fs.readFileSync(args['audio']);
const result = Wav.decode(buffer);
if (result.sampleRate < 16000) {
console.error('Warning: original sample rate (' + result.sampleRate + ') is lower than 16kHz. Up-sampling might produce erratic speech recognition.');
if (result.sampleRate < desired_sample_rate) {
console.error('Warning: original sample rate (' + result.sampleRate + ') ' +
'is lower than ' + desired_sample_rate + 'Hz. ' +
'Up-sampling might produce erratic speech recognition.');
}
function bufferToStream(buffer) {
@ -84,7 +102,7 @@ bufferToStream(buffer).
},
output: {
bits: 16,
rate: 16000,
rate: desired_sample_rate,
channels: 1,
encoding: 'signed-integer',
endian: 'little',
@ -97,23 +115,9 @@ bufferToStream(buffer).
audioStream.on('finish', () => {
let audioBuffer = audioStream.toBuffer();
console.error('Loading model from file %s', args['model']);
const model_load_start = process.hrtime();
var model = new Ds.Model(args['model'], args['alphabet'], BEAM_WIDTH);
const model_load_end = process.hrtime(model_load_start);
console.error('Loaded model in %ds.', totalTime(model_load_end));
if (args['lm'] && args['trie']) {
console.error('Loading language model from files %s %s', args['lm'], args['trie']);
const lm_load_start = process.hrtime();
model.enableDecoderWithLM(args['lm'], args['trie'], LM_ALPHA, LM_BETA);
const lm_load_end = process.hrtime(lm_load_start);
console.error('Loaded language model in %ds.', totalTime(lm_load_end));
}
const inference_start = process.hrtime();
console.error('Running inference.');
const audioLength = (audioBuffer.length / 2) * ( 1 / 16000);
const audioLength = (audioBuffer.length / 2) * (1 / desired_sample_rate);
// We take half of the buffer_size because buffer is a char* while
// LocalDsSTT() expected a short*

9
native_client/javascript/index.js
View File

@ -45,6 +45,15 @@ function Model() {
this._impl = impl;
}
/**
* Return the sample rate expected by the model.
*
* @return {number} Sample rate.
*/
Model.prototype.sampleRate = function() {
return binding.GetModelSampleRate(this._impl);
}
/**
* Enable decoding using beam scoring with a KenLM language model.
*

9
native_client/python/__init__.py
View File

@ -44,6 +44,15 @@ class Model(object):
deepspeech.impl.FreeModel(self._impl)
self._impl = None
def sampleRate(self):
"""
Return the sample rate expected by the model.
:return: Sample rate.
:type: int
"""
return deepspeech.impl.GetModelSampleRate(self._impl)
def enableDecoderWithLM(self, *args, **kwargs):
"""
Enable decoding using beam scoring with a KenLM language model.

21
native_client/python/client.py
View File

@ -17,9 +17,6 @@ try:
except ImportError:
from pipes import quote
# Define the sample rate for audio
SAMPLE_RATE = 16000
# These constants control the beam search decoder
# Beam width used in the CTC decoder when building candidate transcriptions
@ -32,16 +29,16 @@ LM_ALPHA = 0.75
LM_BETA = 1.85
def convert_samplerate(audio_path):
sox_cmd = 'sox {} --type raw --bits 16 --channels 1 --rate {} --encoding signed-integer --endian little --compression 0.0 --no-dither - '.format(quote(audio_path), SAMPLE_RATE)
def convert_samplerate(audio_path, desired_sample_rate):
sox_cmd = 'sox {} --type raw --bits 16 --channels 1 --rate {} --encoding signed-integer --endian little --compression 0.0 --no-dither - '.format(quote(audio_path), desired_sample_rate)
try:
output = subprocess.check_output(shlex.split(sox_cmd), stderr=subprocess.PIPE)
except subprocess.CalledProcessError as e:
raise RuntimeError('SoX returned non-zero status: {}'.format(e.stderr))
except OSError as e:
raise OSError(e.errno, 'SoX not found, use {}hz files or install it: {}'.format(SAMPLE_RATE, e.strerror))
raise OSError(e.errno, 'SoX not found, use {}hz files or install it: {}'.format(desired_sample_rate, e.strerror))
return SAMPLE_RATE, np.frombuffer(output, np.int16)
return desired_sample_rate, np.frombuffer(output, np.int16)
def metadata_to_string(metadata):
@ -81,6 +78,8 @@ def main():
model_load_end = timer() - model_load_start
print('Loaded model in {:.3}s.'.format(model_load_end), file=sys.stderr)
desired_sample_rate = ds.sampleRate()
if args.lm and args.trie:
print('Loading language model from files {} {}'.format(args.lm, args.trie), file=sys.stderr)
lm_load_start = timer()
@ -90,13 +89,13 @@ def main():
fin = wave.open(args.audio, 'rb')
fs = fin.getframerate()
if fs != SAMPLE_RATE:
print('Warning: original sample rate ({}) is different than {}hz. Resampling might produce erratic speech recognition.'.format(fs, SAMPLE_RATE), file=sys.stderr)
fs, audio = convert_samplerate(args.audio)
if fs != desired_sample_rate:
print('Warning: original sample rate ({}) is different than {}hz. Resampling might produce erratic speech recognition.'.format(fs, desired_sample_rate), file=sys.stderr)
fs, audio = convert_samplerate(args.audio, desired_sample_rate)
else:
audio = np.frombuffer(fin.readframes(fin.getnframes()), np.int16)
audio_length = fin.getnframes() * (1/SAMPLE_RATE)
audio_length = fin.getnframes() * (1/fs)
fin.close()
print('Running inference.', file=sys.stderr)