From baaa5842b28aa14829739ece4f40f7c1b8e05635 Mon Sep 17 00:00:00 2001
From: Reuben Morais <reuben.morais@gmail.com>
Date: Thu, 10 Oct 2019 14:20:44 +0200
Subject: [PATCH] Remove sample rate parameter usage from examples

---
 examples/ffmpeg_vad_streaming/index.js                  | 4 ++--
 examples/net_framework/DeepSpeechWPF/MainWindow.xaml.cs | 4 ++--
 examples/nodejs_wav/index.js                            | 2 +-
 examples/vad_transcriber/wavTranscriber.py              | 4 ++--
 4 files changed, 7 insertions(+), 7 deletions(-)

diff --git a/examples/ffmpeg_vad_streaming/index.js b/examples/ffmpeg_vad_streaming/index.js
index 8aef749b..d64cc9f8 100644
--- 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
index 8b38316c..e332da6d 100644
--- a/examples/net_framework/DeepSpeechWPF/MainWindow.xaml.cs
+++ b/examples/net_framework/DeepSpeechWPF/MainWindow.xaml.cs
@@ -130,7 +130,7 @@ namespace DeepSpeechWPF
                 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 @@ namespace DeepSpeechWPF
 
         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
index a5432217..20ccb2ab 100644
--- 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
index 9f21f362..727dc5cf 100644
--- a/examples/vad_transcriber/wavTranscriber.py
+++ b/examples/vad_transcriber/wavTranscriber.py
@@ -44,12 +44,12 @@ Returns a list [Inference, Inference Time, Audio Length]
 '''
 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))