deepspeech-gpu - npm Package Compare versions

Comparing version 0.6.0-alpha.8 to 0.6.0-alpha.9

client.js

@@ -65,7 +65,25 @@ #!/usr/bin/env node
 
+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.');
 }
@@ -88,3 +106,3 @@
       bits: 16,
-      rate: 16000,
+      rate: desired_sample_rate,
       channels: 1,
@@ -102,19 +120,5 @@ encoding: 'signed-integer',
 
-  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);
 
@@ -124,5 +128,5 @@ // We take half of the buffer_size because buffer is a char* while
   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)));
   }
@@ -129,0 +133,0 @@ const inference_stop = process.hrtime(inference_start);

index.js

@@ -49,2 +49,11 @@ 'use strict';
 /**
+ * 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.
@@ -68,5 +77,4 @@ *
  *
- * @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.
  *
@@ -84,5 +92,4 @@ * @return {string} The STT result. Returns undefined on error.
  *
- * @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.
  *
@@ -99,3 +106,2 @@ * @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.
  *
- * @param {number} aSampleRate The sample-rate of the audio signal.
  * @return {object} an opaque object that represents the streaming state.
@@ -121,3 +127,3 @@ *
  * @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.
@@ -124,0 +130,0 @@ */

package.json

 {
     "name" : "deepspeech-gpu",
-    "version" : "0.6.0-alpha.8",
+    "version" : "0.6.0-alpha.9",
     "description" : "DeepSpeech NodeJS bindings",
@@ -11,5 +11,5 @@ "main" : "./index",
     "license": "MPL-2.0",
-    "homepage": "https://github.com/mozilla/DeepSpeech/tree/v0.6.0-alpha.8#project-deepspeech",
+    "homepage": "https://github.com/mozilla/DeepSpeech/tree/v0.6.0-alpha.9#project-deepspeech",
     "files": [
-       "README.md",
+       "README.rst",
        "client.js",
@@ -16,0 +16,0 @@ "index.js",

No alert changes

Improved metrics

Total package byte prevSize

144994265

increased by0.47%

Number of package files

46

increased by4.55%

Lines of code

354

increased by2.31%

Worsened metrics

Number of lines in readme file

92

decreased by-82.14%

No dependency changes