Remove sample rate parameter usage from examples

jizhi · Oct 10, 2019 · baaa584 · baaa584
1 parent 11ad23c
commit baaa584
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Showing 4 changed files with 7 additions and 7 deletions.
diff --git a/examples/ffmpeg_vad_streaming/index.js b/examples/ffmpeg_vad_streaming/index.js
@@ -95,7 +95,7 @@ const ffmpeg = spawn('ffmpeg', [
 ]);
 
 let audioLength = 0;
-let sctx = model.createStream(AUDIO_SAMPLE_RATE);
+let sctx = model.createStream();
 
 function finishStream() {
 	const model_load_start = process.hrtime();
@@ -108,7 +108,7 @@ function finishStream() {
 
 function intermediateDecode() {
 	finishStream();
-	sctx = model.createStream(AUDIO_SAMPLE_RATE);
+	sctx = model.createStream();
 }
 
 function feedAudioContent(chunk) {

diff --git a/examples/net_framework/DeepSpeechWPF/MainWindow.xaml.cs b/examples/net_framework/DeepSpeechWPF/MainWindow.xaml.cs
@@ -130,7 +130,7 @@ private async void BtnTranscript_Click(object sender, RoutedEventArgs e)
                 watch.Start();
                 await Task.Run(() =>
                 {
-                    string speechResult = _sttClient.SpeechToText(waveBuffer.ShortBuffer, Convert.ToUInt32(waveBuffer.MaxSize / 2), 16000);
+                    string speechResult = _sttClient.SpeechToText(waveBuffer.ShortBuffer, Convert.ToUInt32(waveBuffer.MaxSize / 2));
                     watch.Stop();
                     Dispatcher.Invoke(() =>
                     {
@@ -250,7 +250,7 @@ private void _capture_DataAvailable(object sender, DataAvailableEventArgs e)
 
         private void BtnStartRecording_Click(object sender, RoutedEventArgs e)
         {
-            _sttClient.CreateStream(16000);
+            _sttClient.CreateStream();
             _audioCapture.Start();
             btnStartRecording.IsEnabled = false;
             btnStopRecording.IsEnabled = true;

diff --git a/examples/nodejs_wav/index.js b/examples/nodejs_wav/index.js
@@ -64,7 +64,7 @@ audioStream.on('finish', () => {
 	const audioLength = (audioBuffer.length / 2) * ( 1 / 16000);
 	console.log('audio length', audioLength);
 
-	let result = model.stt(audioBuffer.slice(0, audioBuffer.length / 2), 16000);
+	let result = model.stt(audioBuffer.slice(0, audioBuffer.length / 2));
 
 	console.log('result:', result);
 });
diff --git a/examples/vad_transcriber/wavTranscriber.py b/examples/vad_transcriber/wavTranscriber.py
@@ -44,12 +44,12 @@ def load_model(models, alphabet, lm, trie):
 '''
 def stt(ds, audio, fs):
     inference_time = 0.0
-    audio_length = len(audio) * (1 / 16000)
+    audio_length = len(audio) * (1 / fs)
 
     # Run Deepspeech
     logging.debug('Running inference...')
     inference_start = timer()
-    output = ds.stt(audio, fs)
+    output = ds.stt(audio)
     inference_end = timer() - inference_start
     inference_time += inference_end
     logging.debug('Inference took %0.3fs for %0.3fs audio file.' % (inference_end, audio_length))