Merge pull request #2420 from mozilla/remove-sr-param

Remove unused sample rate param from API

evaluate_tflite.py

@@ -45,7 +45,7 @@ def tflite_worker(model, alphabet, lm, trie, queue_in, queue_out, gpu_mask):
         audio = np.frombuffer(fin.readframes(fin.getnframes()), np.int16)
         fin.close()
 
-        decoded = ds.stt(audio, fs)
+        decoded = ds.stt(audio)
 
         queue_out.put({'wav': wavname, 'prediction': decoded, 'ground_truth': msg['transcript']})
         print(queue_out.qsize(), end='\r') # Update the current progress

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) {

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;

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);
 });

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))

native_client/client.cc

@@ -54,23 +54,23 @@ char* JSONOutput(Metadata* metadata);
 
 ds_result
 LocalDsSTT(ModelState* aCtx, const short* aBuffer, size_t aBufferSize,
-           int aSampleRate, bool extended_output, bool json_output)
+           bool extended_output, bool json_output)
 {
   ds_result res = {0};
 
   clock_t ds_start_time = clock();
 
   if (extended_output) {
-    Metadata *metadata = DS_SpeechToTextWithMetadata(aCtx, aBuffer, aBufferSize, aSampleRate);
+    Metadata *metadata = DS_SpeechToTextWithMetadata(aCtx, aBuffer, aBufferSize);
     res.string = metadataToString(metadata);
     DS_FreeMetadata(metadata);
   } else if (json_output) {
-    Metadata *metadata = DS_SpeechToTextWithMetadata(aCtx, aBuffer, aBufferSize, aSampleRate);
+    Metadata *metadata = DS_SpeechToTextWithMetadata(aCtx, aBuffer, aBufferSize);
     res.string = JSONOutput(metadata);
     DS_FreeMetadata(metadata);
   } else if (stream_size > 0) {
     StreamingState* ctx;
-    int status = DS_CreateStream(aCtx, aSampleRate, &ctx);
+    int status = DS_CreateStream(aCtx, &ctx);
     if (status != DS_ERR_OK) {
       res.string = strdup("");
       return res;
@@ -94,7 +94,7 @@ LocalDsSTT(ModelState* aCtx, const short* aBuffer, size_t aBufferSize,
     }
     res.string = DS_FinishStream(ctx);
   } else {
-    res.string = DS_SpeechToText(aCtx, aBuffer, aBufferSize, aSampleRate);
+    res.string = DS_SpeechToText(aCtx, aBuffer, aBufferSize);
   }
 
   clock_t ds_end_infer = clock();
@@ -108,7 +108,6 @@ LocalDsSTT(ModelState* aCtx, const short* aBuffer, size_t aBufferSize,
 typedef struct {
   char*  buffer;
   size_t buffer_size;
-  int    sample_rate;
 } ds_audio_buffer;
 
 ds_audio_buffer
@@ -159,8 +158,6 @@ GetAudioBuffer(const char* path)
 
   assert(output);
 
-  res.sample_rate = (int)output->signal.rate;
-
   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);
   }
@@ -221,7 +218,6 @@ GetAudioBuffer(const char* path)
 
   unsigned int sample_rate;
   fseek(wave, 24, SEEK_SET); rv = fread(&sample_rate, 4, 1, wave);
-  res.sample_rate = (int)sample_rate;
 
   unsigned short bits_per_sample;
   fseek(wave, 34, SEEK_SET); rv = fread(&bits_per_sample, 2, 1, wave);
@@ -269,7 +265,6 @@ ProcessFile(ModelState* context, const char* path, bool show_times)
   ds_result result = LocalDsSTT(context,
                                 (const short*)audio.buffer,
                                 audio.buffer_size / 2,
-                                audio.sample_rate,
                                 extended_metadata,
                                 json_output);
   free(audio.buffer);

native_client/deepspeech.cc

@@ -318,7 +318,6 @@ DS_EnableDecoderWithLM(ModelState* aCtx,
 
 int
 DS_CreateStream(ModelState* aCtx,
-                unsigned int aSampleRate,
                 StreamingState** retval)
 {
   *retval = nullptr;
@@ -383,11 +382,10 @@ DS_FinishStreamWithMetadata(StreamingState* aSctx)
 StreamingState*
 CreateStreamAndFeedAudioContent(ModelState* aCtx,
                                 const short* aBuffer,
-                                unsigned int aBufferSize,
+                                unsigned int aBufferSize)
-                                unsigned int aSampleRate)
 {
   StreamingState* ctx;
-  int status = DS_CreateStream(aCtx, aSampleRate, &ctx);
+  int status = DS_CreateStream(aCtx, &ctx);
   if (status != DS_ERR_OK) {
     return nullptr;
   }
@@ -398,20 +396,18 @@ CreateStreamAndFeedAudioContent(ModelState* aCtx,
 char*
 DS_SpeechToText(ModelState* aCtx,
                 const short* aBuffer,
-                unsigned int aBufferSize,
+                unsigned int aBufferSize)
-                unsigned int aSampleRate)
 {
-  StreamingState* ctx = CreateStreamAndFeedAudioContent(aCtx, aBuffer, aBufferSize, aSampleRate);
+  StreamingState* ctx = CreateStreamAndFeedAudioContent(aCtx, aBuffer, aBufferSize);
   return DS_FinishStream(ctx);
 }
 
 Metadata*
 DS_SpeechToTextWithMetadata(ModelState* aCtx,
                             const short* aBuffer,
-                            unsigned int aBufferSize,
+                            unsigned int aBufferSize)
-                            unsigned int aSampleRate)
 {
-  StreamingState* ctx = CreateStreamAndFeedAudioContent(aCtx, aBuffer, aBufferSize, aSampleRate);
+  StreamingState* ctx = CreateStreamAndFeedAudioContent(aCtx, aBuffer, aBufferSize);
   return DS_FinishStreamWithMetadata(ctx);
 }
 

native_client/deepspeech.h

@@ -124,9 +124,8 @@ int DS_EnableDecoderWithLM(ModelState* aCtx,
  *
  * @param aCtx The ModelState pointer for the model to use.
  * @param aBuffer A 16-bit, mono raw audio signal at the appropriate
- *                sample rate.
+ *                sample rate (matching what the model was trained on).
  * @param aBufferSize The number of samples in the audio signal.
- * @param aSampleRate The sample-rate of the audio signal.
  *
  * @return The STT result. The user is responsible for freeing the string using
  *         {@link DS_FreeString()}. Returns NULL on error.
@@ -134,8 +133,7 @@ int DS_EnableDecoderWithLM(ModelState* aCtx,
 DEEPSPEECH_EXPORT
 char* DS_SpeechToText(ModelState* aCtx,
                       const short* aBuffer,
-                      unsigned int aBufferSize,
+                      unsigned int aBufferSize);
-                      unsigned int aSampleRate);
 
 /**
  * @brief Use the DeepSpeech model to perform Speech-To-Text and output metadata 
@@ -143,9 +141,8 @@ char* DS_SpeechToText(ModelState* aCtx,
  *
  * @param aCtx The ModelState pointer for the model to use.
  * @param aBuffer A 16-bit, mono raw audio signal at the appropriate
- *                sample rate.
+ *                sample rate (matching what the model was trained on).
  * @param aBufferSize The number of samples in the audio signal.
- * @param aSampleRate The sample-rate of the audio signal.
  *
  * @return Outputs a struct of individual letters along with their timing information. 
  *         The user is responsible for freeing Metadata by calling {@link DS_FreeMetadata()}. Returns NULL on error.
@@ -153,8 +150,7 @@ char* DS_SpeechToText(ModelState* aCtx,
 DEEPSPEECH_EXPORT
 Metadata* DS_SpeechToTextWithMetadata(ModelState* aCtx,
                                       const short* aBuffer,
-                                      unsigned int aBufferSize,
+                                      unsigned int aBufferSize);
-                                      unsigned int aSampleRate);
 
 /**
  * @brief Create a new streaming inference state. The streaming state returned
@@ -162,7 +158,6 @@ Metadata* DS_SpeechToTextWithMetadata(ModelState* aCtx,
  *        and {@link DS_FinishStream()}.
  *
  * @param aCtx The ModelState pointer for the model to use.
- * @param aSampleRate The sample-rate of the audio signal.
  * @param[out] retval an opaque pointer that represents the streaming state. Can
  *                    be NULL if an error occurs.
  *
@@ -170,7 +165,6 @@ Metadata* DS_SpeechToTextWithMetadata(ModelState* aCtx,
  */
 DEEPSPEECH_EXPORT
 int DS_CreateStream(ModelState* aCtx,
-                    unsigned int aSampleRate,
                     StreamingState** retval);
 
 /**
@@ -178,7 +172,7 @@ int DS_CreateStream(ModelState* aCtx,
  *
  * @param aSctx A streaming state pointer returned by {@link DS_CreateStream()}.
  * @param aBuffer An array of 16-bit, mono raw audio samples at the
- *                appropriate sample rate.
+ *                appropriate sample rate (matching what the model was trained on).
  * @param aBufferSize The number of samples in @p aBuffer.
  */
 DEEPSPEECH_EXPORT

native_client/deepspeech_compat.h

@@ -71,17 +71,17 @@ int DS_EnableDecoderWithLM(ModelState* aCtx,
  *        and {@link DS_FinishStream()}.
  *
  * @param aCtx The ModelState pointer for the model to use.
- * @param aSampleRate The sample-rate of the audio signal.
+ * @param aSampleRate UNUSED, DEPRECATED.
  * @param[out] retval an opaque pointer that represents the streaming state. Can
  *                    be NULL if an error occurs.
  *
  * @return Zero for success, non-zero on failure.
  */
 int DS_SetupStream(ModelState* aCtx,
-                   unsigned int aSampleRate,
+                   unsigned int /*aSampleRate*/,
                    StreamingState** retval)
 {
-  return DS_CreateStream(aCtx, aSampleRate, retval);
+  return DS_CreateStream(aCtx, retval);
 }
 
 /**
@@ -98,4 +98,45 @@ void DS_DiscardStream(StreamingState* aSctx)
   return DS_FreeStream(aSctx);
 }
 
+/**
+ * @brief Use the DeepSpeech model to perform Speech-To-Text.
+ *
+ * @param aCtx The ModelState pointer for the model to use.
+ * @param aBuffer A 16-bit, mono raw audio signal at the appropriate
+ *                sample rate (matching what the model was trained on).
+ * @param aBufferSize The number of samples in the audio signal.
+ * @param aSampleRate UNUSED, DEPRECATED.
+ *
+ * @return The STT result. The user is responsible for freeing the string using
+ *         {@link DS_FreeString()}. Returns NULL on error.
+ */
+char* DS_SpeechToText(ModelState* aCtx,
+                      const short* aBuffer,
+                      unsigned int aBufferSize,
+                      unsigned int /*aSampleRate*/)
+{
+  return DS_SpeechToText(aCtx, aBuffer, aBufferSize);
+}
+
+/**
+ * @brief Use the DeepSpeech model to perform Speech-To-Text and output metadata
+ * about the results.
+ *
+ * @param aCtx The ModelState pointer for the model to use.
+ * @param aBuffer A 16-bit, mono raw audio signal at the appropriate
+ *                sample rate (matching what the model was trained on).
+ * @param aBufferSize The number of samples in the audio signal.
+ * @param aSampleRate UNUSED, DEPRECATED.
+ *
+ * @return Outputs a struct of individual letters along with their timing information.
+ *         The user is responsible for freeing Metadata by calling {@link DS_FreeMetadata()}. Returns NULL on error.
+ */
+Metadata* DS_SpeechToTextWithMetadata(ModelState* aCtx,
+                                      const short* aBuffer,
+                                      unsigned int aBufferSize,
+                                      unsigned int /*aSampleRate*/)
+{
+  return DS_SpeechToTextWithMetadata(aCtx, aBuffer, aBufferSize);
+}
+
 #endif /* DEEPSPEECH_COMPAT_H */

native_client/dotnet/DeepSpeechClient/DeepSpeech.cs

@@ -148,7 +148,7 @@ namespace DeepSpeechClient
         /// <summary>
         /// Feeds audio samples to an ongoing streaming inference.
         /// </summary>
-        /// <param name="aBuffer">An array of 16-bit, mono raw audio samples at the appropriate sample rate.</param>
+        /// <param name="aBuffer">An array of 16-bit, mono raw audio samples at the appropriate sample rate (matching what the model was trained on).</param>
         public unsafe void FeedAudioContent(short[] aBuffer, uint aBufferSize)
         {
             NativeImp.DS_FeedAudioContent(_streamingStatePP, aBuffer, aBufferSize);
@@ -193,11 +193,10 @@ namespace DeepSpeechClient
         /// <summary>
         /// Creates a new streaming inference state.
         /// </summary>
-        /// <param name="aSampleRate">The sample-rate of the audio signal</param>
         /// <exception cref="ArgumentException">Thrown when the native binary failed to initialize the streaming mode.</exception>
-        public unsafe void CreateStream(uint aSampleRate)
+        public unsafe void CreateStream()
         {
-            var resultCode = NativeImp.DS_CreateStream(_modelStatePP, aSampleRate, ref _streamingStatePP);
+            var resultCode = NativeImp.DS_CreateStream(_modelStatePP, ref _streamingStatePP);
             EvaluateResultCode(resultCode);
         }
 
@@ -230,25 +229,23 @@ namespace DeepSpeechClient
         /// <summary>
         /// Use the DeepSpeech model to perform Speech-To-Text.
         /// </summary>
-        /// <param name="aBuffer">A 16-bit, mono raw audio signal at the appropriate sample rate.</param>
+        /// <param name="aBuffer">A 16-bit, mono raw audio signal at the appropriate sample rate (matching what the model was trained on).</param>
         /// <param name="aBufferSize">The number of samples in the audio signal.</param>
-        /// <param name="aSampleRate">The sample-rate of the audio signal.</param>
         /// <returns>The STT result. The user is responsible for freeing the string.  Returns NULL on error.</returns>
-        public unsafe string SpeechToText(short[] aBuffer, uint aBufferSize, uint aSampleRate)
+        public unsafe string SpeechToText(short[] aBuffer, uint aBufferSize)
         {
-            return NativeImp.DS_SpeechToText(_modelStatePP, aBuffer, aBufferSize, aSampleRate).PtrToString();
+            return NativeImp.DS_SpeechToText(_modelStatePP, aBuffer, aBufferSize).PtrToString();
         }
 
         /// <summary>
         /// Use the DeepSpeech model to perform Speech-To-Text.
         /// </summary>
-        /// <param name="aBuffer">A 16-bit, mono raw audio signal at the appropriate sample rate.</param>
+        /// <param name="aBuffer">A 16-bit, mono raw audio signal at the appropriate sample rate (matching what the model was trained on).</param>
         /// <param name="aBufferSize">The number of samples in the audio signal.</param>
-        /// <param name="aSampleRate">The sample-rate of the audio signal.</param>
         /// <returns>The extended metadata. The user is responsible for freeing the struct.  Returns NULL on error.</returns>
-        public unsafe Models.Metadata SpeechToTextWithMetadata(short[] aBuffer, uint aBufferSize, uint aSampleRate)
+        public unsafe Models.Metadata SpeechToTextWithMetadata(short[] aBuffer, uint aBufferSize)
         {
-            return NativeImp.DS_SpeechToTextWithMetadata(_modelStatePP, aBuffer, aBufferSize, aSampleRate).PtrToMetadata();
+            return NativeImp.DS_SpeechToTextWithMetadata(_modelStatePP, aBuffer, aBufferSize).PtrToMetadata();
         }
 
         #endregion

native_client/dotnet/DeepSpeechClient/Interfaces/IDeepSpeech.cs

@@ -40,24 +40,20 @@ namespace DeepSpeechClient.Interfaces
         /// <summary>
         /// Use the DeepSpeech model to perform Speech-To-Text.
         /// </summary>
-        /// <param name="aBuffer">A 16-bit, mono raw audio signal at the appropriate sample rate.</param>
+        /// <param name="aBuffer">A 16-bit, mono raw audio signal at the appropriate sample rate (matching what the model was trained on).</param>
         /// <param name="aBufferSize">The number of samples in the audio signal.</param>
-        /// <param name="aSampleRate">The sample-rate of the audio signal.</param>
         /// <returns>The STT result. The user is responsible for freeing the string.  Returns NULL on error.</returns>
         unsafe string SpeechToText(short[] aBuffer,
-                uint aBufferSize,
+                uint aBufferSize);
-                uint aSampleRate);
 
         /// <summary>
         /// Use the DeepSpeech model to perform Speech-To-Text.
         /// </summary>
-        /// <param name="aBuffer">A 16-bit, mono raw audio signal at the appropriate sample rate.</param>
+        /// <param name="aBuffer">A 16-bit, mono raw audio signal at the appropriate sample rate (matching what the model was trained on).</param>
         /// <param name="aBufferSize">The number of samples in the audio signal.</param>
-        /// <param name="aSampleRate">The sample-rate of the audio signal.</param>
         /// <returns>The extended metadata result. The user is responsible for freeing the struct.  Returns NULL on error.</returns>
         unsafe Metadata SpeechToTextWithMetadata(short[] aBuffer,
-                uint aBufferSize,
+                uint aBufferSize);
-                uint aSampleRate);
 
         /// <summary>
         /// Destroy a streaming state without decoding the computed logits.
@@ -79,14 +75,13 @@ namespace DeepSpeechClient.Interfaces
         /// <summary>
         /// Creates a new streaming inference state.
         /// </summary>
-        /// <param name="aSampleRate">The sample-rate of the audio signal</param>
         /// <exception cref="ArgumentException">Thrown when the native binary failed to initialize the streaming mode.</exception>
-        unsafe void CreateStream(uint aSampleRate);
+        unsafe void CreateStream();
 
         /// <summary>
         /// Feeds audio samples to an ongoing streaming inference.
         /// </summary>
-        /// <param name="aBuffer">An array of 16-bit, mono raw audio samples at the appropriate sample rate.</param>
+        /// <param name="aBuffer">An array of 16-bit, mono raw audio samples at the appropriate sample rate (matching what the model was trained on).</param>
         unsafe void FeedAudioContent(short[] aBuffer, uint aBufferSize);
 
         /// <summary>

native_client/dotnet/DeepSpeechClient/NativeImp.cs

@@ -31,21 +31,19 @@ namespace DeepSpeechClient
             CharSet = CharSet.Ansi, SetLastError = true)]
         internal static unsafe extern IntPtr DS_SpeechToText(IntPtr** aCtx,
                  short[] aBuffer,
-                uint aBufferSize,
+                uint aBufferSize);
-                uint aSampleRate);
 
         [DllImport("libdeepspeech.so", CallingConvention = CallingConvention.Cdecl, SetLastError = true)]
         internal static unsafe extern IntPtr DS_SpeechToTextWithMetadata(IntPtr** aCtx,
                  short[] aBuffer,
-                uint aBufferSize,
+                uint aBufferSize);
-                uint aSampleRate);
 
         [DllImport("libdeepspeech.so", CallingConvention = CallingConvention.Cdecl)]
         internal static unsafe extern void DS_FreeModel(IntPtr** aCtx);
 
         [DllImport("libdeepspeech.so", CallingConvention = CallingConvention.Cdecl)]
         internal static unsafe extern ErrorCodes DS_CreateStream(IntPtr** aCtx,
-               uint aSampleRate, ref IntPtr** retval);
+               ref IntPtr** retval);
 
         [DllImport("libdeepspeech.so", CallingConvention = CallingConvention.Cdecl)]
         internal static unsafe extern void DS_FreeStream(ref IntPtr** aSctx);

native_client/dotnet/DeepSpeechConsole/Program.cs

@@ -91,12 +91,12 @@ namespace CSharpExamples
                         string speechResult;
                         if (extended)
                         {
-                            Metadata metaResult = sttClient.SpeechToTextWithMetadata(waveBuffer.ShortBuffer, Convert.ToUInt32(waveBuffer.MaxSize / 2), 16000);
+                            Metadata metaResult = sttClient.SpeechToTextWithMetadata(waveBuffer.ShortBuffer, Convert.ToUInt32(waveBuffer.MaxSize / 2));
                             speechResult = MetadataToString(metaResult);
                         }
                         else
                         {
-                            speechResult = sttClient.SpeechToText(waveBuffer.ShortBuffer, Convert.ToUInt32(waveBuffer.MaxSize / 2), 16000);
+                            speechResult = sttClient.SpeechToText(waveBuffer.ShortBuffer, Convert.ToUInt32(waveBuffer.MaxSize / 2));
                         }
 
                         stopwatch.Stop();

native_client/java/app/src/main/java/org/mozilla/deepspeech/DeepSpeechActivity.java

@@ -100,7 +100,7 @@ public class DeepSpeechActivity extends AppCompatActivity {
 
             long inferenceStartTime = System.currentTimeMillis();
 
-            String decoded = this._m.stt(shorts, shorts.length, sampleRate);
+            String decoded = this._m.stt(shorts, shorts.length);
 
             inferenceExecTime = System.currentTimeMillis() - inferenceStartTime;
 

native_client/java/libdeepspeech/src/androidTest/java/org/mozilla/deepspeech/libdeepspeech/test/BasicTest.java

@@ -104,9 +104,9 @@ public class BasicTest {
             ByteBuffer.wrap(bytes).order(ByteOrder.LITTLE_ENDIAN).asShortBuffer().get(shorts);
 
             if (extendedMetadata) {
-                return metadataToString(m.sttWithMetadata(shorts, shorts.length, sampleRate));
+                return metadataToString(m.sttWithMetadata(shorts, shorts.length));
             } else {
-                return m.stt(shorts, shorts.length, sampleRate);
+                return m.stt(shorts, shorts.length);
             }
         } catch (FileNotFoundException ex) {
 

native_client/java/libdeepspeech/src/main/java/org/mozilla/deepspeech/libdeepspeech/DeepSpeechModel.java

@@ -57,14 +57,13 @@ public class DeepSpeechModel {
     * @brief Use the DeepSpeech model to perform Speech-To-Text.
     *
     * @param buffer A 16-bit, mono raw audio signal at the appropriate
-    *                sample rate.
+    *                sample rate (matching what the model was trained on).
     * @param buffer_size The number of samples in the audio signal.
-    * @param sample_rate The sample-rate of the audio signal.
     *
     * @return The STT result.
     */
-    public String stt(short[] buffer, int buffer_size, int sample_rate) {
+    public String stt(short[] buffer, int buffer_size) {
-        return impl.SpeechToText(this._msp, buffer, buffer_size, sample_rate);
+        return impl.SpeechToText(this._msp, buffer, buffer_size);
     }
 
    /**
@@ -72,14 +71,13 @@ public class DeepSpeechModel {
     * about the results.
     *
     * @param buffer A 16-bit, mono raw audio signal at the appropriate
-    *                sample rate.
+    *                sample rate (matching what the model was trained on).
     * @param buffer_size The number of samples in the audio signal.
-    * @param sample_rate The sample-rate of the audio signal.
     *
     * @return Outputs a Metadata object of individual letters along with their timing information.
     */
-    public Metadata sttWithMetadata(short[] buffer, int buffer_size, int sample_rate) {
+    public Metadata sttWithMetadata(short[] buffer, int buffer_size) {
-        return impl.SpeechToTextWithMetadata(this._msp, buffer, buffer_size, sample_rate);
+        return impl.SpeechToTextWithMetadata(this._msp, buffer, buffer_size);
     }
 
    /**
@@ -87,12 +85,11 @@ public class DeepSpeechModel {
     *        by this function can then be passed to feedAudioContent()
     *        and finishStream().
     *
-    * @param sample_rate The sample-rate of the audio signal.
     * @return An opaque object that represents the streaming state.
     */
-    public DeepSpeechStreamingState createStream(int sample_rate) {
+    public DeepSpeechStreamingState createStream() {
         SWIGTYPE_p_p_StreamingState ssp = impl.new_streamingstatep();
-        impl.CreateStream(this._msp, sample_rate, ssp);
+        impl.CreateStream(this._msp, ssp);
         return new DeepSpeechStreamingState(impl.streamingstatep_value(ssp));
     }
 
@@ -101,7 +98,7 @@ public class DeepSpeechModel {
     *
     * @param cctx A streaming state pointer returned by createStream().
     * @param buffer An array of 16-bit, mono raw audio samples at the
-    *                appropriate sample rate.
+    *                appropriate sample rate (matching what the model was trained on).
     * @param buffer_size The number of samples in @p buffer.
     */
     public void feedAudioContent(DeepSpeechStreamingState ctx, short[] buffer, int buffer_size) {

native_client/javascript/client.js

@@ -118,9 +118,9 @@ audioStream.on('finish', () => {
   // We take half of the buffer_size because buffer is a char* while
   // LocalDsSTT() expected a short*
   if (args['extended']) {
-    console.log(metadataToString(model.sttWithMetadata(audioBuffer.slice(0, audioBuffer.length / 2), 16000)));
+    console.log(metadataToString(model.sttWithMetadata(audioBuffer.slice(0, audioBuffer.length / 2))));
   } else {
-    console.log(model.stt(audioBuffer.slice(0, audioBuffer.length / 2), 16000));
+    console.log(model.stt(audioBuffer.slice(0, audioBuffer.length / 2)));
   }
   const inference_stop = process.hrtime(inference_start);
   console.error('Inference took %ds for %ds audio file.', totalTime(inference_stop), audioLength.toPrecision(4));

native_client/javascript/index.js

@@ -64,9 +64,8 @@ Model.prototype.enableDecoderWithLM = function() {
 /**
  * Use the DeepSpeech model to perform Speech-To-Text.
  *
- * @param {object} aBuffer A 16-bit, mono raw audio signal at the appropriate sample rate.
+ * @param {object} aBuffer A 16-bit, mono raw audio signal at the appropriate sample rate (matching what the model was trained on).
  * @param {number} aBufferSize The number of samples in the audio signal.
- * @param {number} aSampleRate The sample-rate of the audio signal.
  *
  * @return {string} The STT result. Returns undefined on error.
  */
@@ -79,9 +78,8 @@ Model.prototype.stt = function() {
  * Use the DeepSpeech model to perform Speech-To-Text and output metadata
  * about the results.
  *
- * @param {object} aBuffer A 16-bit, mono raw audio signal at the appropriate sample rate.
+ * @param {object} aBuffer A 16-bit, mono raw audio signal at the appropriate sample rate (matching what the model was trained on).
  * @param {number} aBufferSize The number of samples in the audio signal.
- * @param {number} aSampleRate The sample-rate of the audio signal.
  *
  * @return {object} Outputs a :js:func:`Metadata` struct of individual letters along with their timing information. The user is responsible for freeing Metadata by calling :js:func:`FreeMetadata`. Returns undefined on error.
  */
@@ -93,7 +91,6 @@ Model.prototype.sttWithMetadata = function() {
 /**
  * Create a new streaming inference state. The streaming state returned by this function can then be passed to :js:func:`Model.feedAudioContent` and :js:func:`Model.finishStream`.
  *
- * @param {number} aSampleRate The sample-rate of the audio signal.
  * @return {object} an opaque object that represents the streaming state.
  *
  * @throws on error
@@ -114,7 +111,7 @@ Model.prototype.createStream = function() {
  *
  * @param {object} aSctx A streaming state returned by :js:func:`Model.setupStream`.
  * @param {buffer} aBuffer An array of 16-bit, mono raw audio samples at the
- *                 appropriate sample rate.
+ *                 appropriate sample rate (matching what the model was trained on).
  * @param {number} aBufferSize The number of samples in @param aBuffer.
  */
 Model.prototype.feedAudioContent = function() {

native_client/python/__init__.py

@@ -69,15 +69,12 @@ class Model(object):
         """
         Use the DeepSpeech model to perform Speech-To-Text.
 
-        :param aBuffer: A 16-bit, mono raw audio signal at the appropriate sample rate.
+        :param aBuffer: A 16-bit, mono raw audio signal at the appropriate sample rate (matching what the model was trained on).
         :type aBuffer: int array
 
         :param aBufferSize: The number of samples in the audio signal.
         :type aBufferSize: int
 
-        :param aSampleRate: The sample-rate of the audio signal.
-        :type aSampleRate: int
-
         :return: The STT result.
         :type: str
         """
@@ -87,34 +84,27 @@ class Model(object):
         """
         Use the DeepSpeech model to perform Speech-To-Text and output metadata about the results.
 
-        :param aBuffer: A 16-bit, mono raw audio signal at the appropriate sample rate.
+        :param aBuffer: A 16-bit, mono raw audio signal at the appropriate sample rate (matching what the model was trained on).
         :type aBuffer: int array
 
         :param aBufferSize: The number of samples in the audio signal.
         :type aBufferSize: int
 
-        :param aSampleRate: The sample-rate of the audio signal.
-        :type aSampleRate: int
-
         :return: Outputs a struct of individual letters along with their timing information.
         :type: :func:`Metadata`
         """
         return deepspeech.impl.SpeechToTextWithMetadata(self._impl, *args, **kwargs)
 
-    def createStream(self, sample_rate=16000):
+    def createStream(self):
         """
         Create a new streaming inference state. The streaming state returned
         by this function can then be passed to :func:`feedAudioContent()` and :func:`finishStream()`.
 
-        :param aSampleRate: The sample-rate of the audio signal.
-        :type aSampleRate: int
-
         :return: Object holding the stream
 
         :throws: RuntimeError on error
         """
-        status, ctx = deepspeech.impl.CreateStream(self._impl,
+        status, ctx = deepspeech.impl.CreateStream(self._impl)
-                                                   aSampleRate=sample_rate)
         if status != 0:
             raise RuntimeError("CreateStream failed with error code {}".format(status))
         return ctx
@@ -127,7 +117,7 @@ class Model(object):
         :param aSctx: A streaming state pointer returned by :func:`createStream()`.
         :type aSctx: object
 
-        :param aBuffer: An array of 16-bit, mono raw audio samples at the appropriate sample rate.
+        :param aBuffer: An array of 16-bit, mono raw audio samples at the appropriate sample rate (matching what the model was trained on).
         :type aBuffer: int array
 
         :param aBufferSize: The number of samples in @p aBuffer.

native_client/python/client.py

@@ -102,9 +102,9 @@ def main():
     print('Running inference.', file=sys.stderr)
     inference_start = timer()
     if args.extended:
-        print(metadata_to_string(ds.sttWithMetadata(audio, fs)))
+        print(metadata_to_string(ds.sttWithMetadata(audio)))
     else:
-        print(ds.stt(audio, fs))
+        print(ds.stt(audio))
     inference_end = timer() - inference_start
     print('Inference took %0.3fs for %0.3fs audio file.' % (inference_end, audio_length), file=sys.stderr)
 

native_client/test/concurrent_streams.py

@@ -52,8 +52,8 @@ def main():
     audio2 = np.frombuffer(fin.readframes(fin.getnframes()), np.int16)
     fin.close()
 
-    stream1 = ds.createStream(sample_rate=fs1)
+    stream1 = ds.createStream()
-    stream2 = ds.createStream(sample_rate=fs2)
+    stream2 = ds.createStream()
 
     splits1 = np.array_split(audio1, 10)
     splits2 = np.array_split(audio2, 10)