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Use model sample rate in client.cc

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This commit is contained in:
Reuben Morais 2019-10-10 21:46:01 +02:00
parent 0241f725cd
commit c1ed6d711d
1 changed files with 10 additions and 8 deletions
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18
native_client/client.cc
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@ -111,7 +111,7 @@ typedef struct {
} ds_audio_buffer; } ds_audio_buffer;
ds_audio_buffer ds_audio_buffer
GetAudioBuffer(const char* path) GetAudioBuffer(const char* path, int desired_sample_rate)
{ {
ds_audio_buffer res = {0}; 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 // Resample/reformat the audio so we can pass it through the MFCC functions
sox_signalinfo_t target_signal = { sox_signalinfo_t target_signal = {
16000, // Rate static_cast<sox_rate_t>(desired_sample_rate), // Rate
1, // Channels 1, // Channels
16, // Precision 16, // Precision
SOX_UNSPEC, // Length SOX_UNSPEC, // Length
@ -158,8 +158,10 @@ GetAudioBuffer(const char* path)
assert(output); assert(output);
if ((int)input->signal.rate < 16000) { if ((int)input->signal.rate < desired_sample_rate) {
fprintf(stderr, "Warning: original sample rate (%d) is lower than 16kHz. Up-sampling might produce erratic speech recognition.\n", (int)input->signal.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 // Setup the effects chain to decode/resample
@ -205,7 +207,7 @@ GetAudioBuffer(const char* path)
#endif // NO_SOX #endif // NO_SOX
#ifdef 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"); FILE* wave = fopen(path, "r");
size_t rv; size_t rv;
@ -224,12 +226,12 @@ GetAudioBuffer(const char* path)
assert(audio_format == 1); // 1 is PCM assert(audio_format == 1); // 1 is PCM
assert(num_channels == 1); // MONO 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 assert(bits_per_sample == 16); // 16 bits per sample
fprintf(stderr, "audio_format=%d\n", audio_format); fprintf(stderr, "audio_format=%d\n", audio_format);
fprintf(stderr, "num_channels=%d\n", num_channels); 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); fprintf(stderr, "bits_per_sample=%d\n", bits_per_sample);
fseek(wave, 40, SEEK_SET); rv = fread(&res.buffer_size, 4, 1, wave); fseek(wave, 40, SEEK_SET); rv = fread(&res.buffer_size, 4, 1, wave);
@ -257,7 +259,7 @@ GetAudioBuffer(const char* path)
void void
ProcessFile(ModelState* context, const char* path, bool show_times) 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 // Pass audio to DeepSpeech
// We take half of buffer_size because buffer is a char* while // We take half of buffer_size because buffer is a char* while