Merge branch 'more-metadata'

DeepSpeech.py

@@ -534,9 +534,9 @@ def create_inference_graph(batch_size=1, n_steps=16, tflite=False):
     batch_size = batch_size if batch_size > 0 else None
 
     # Create feature computation graph
-    input_samples = tf.placeholder(tf.float32, [512], 'input_samples')
+    input_samples = tf.placeholder(tf.float32, [Config.audio_window_samples], 'input_samples')
     samples = tf.expand_dims(input_samples, -1)
-    mfccs, _ = samples_to_mfccs(samples, 16000)
+    mfccs, _ = samples_to_mfccs(samples, FLAGS.audio_sample_rate)
     mfccs = tf.identity(mfccs, name='mfccs')
 
     # Input tensor will be of shape [batch_size, n_steps, 2*n_context+1, n_input]
@@ -710,6 +710,17 @@ def export():
         if not FLAGS.export_tflite:
             frozen_graph = do_graph_freeze(output_node_names=output_names, variables_blacklist='previous_state_c,previous_state_h')
             frozen_graph.version = int(file_relative_read('GRAPH_VERSION').strip())
+
+            # Add a no-op node to the graph with metadata information to be loaded by the native client
+            metadata = frozen_graph.node.add()
+            metadata.name = 'model_metadata'
+            metadata.op = 'NoOp'
+            metadata.attr['sample_rate'].i = FLAGS.audio_sample_rate
+            metadata.attr['feature_win_len'].i = FLAGS.feature_win_len
+            metadata.attr['feature_win_step'].i = FLAGS.feature_win_step
+            if FLAGS.export_model_language:
+                metadata.attr['language'].s = FLAGS.export_model_language.encode('ascii')
+
             with open(output_graph_path, 'wb') as fout:
                 fout.write(frozen_graph.SerializeToString())
         else:

native_client/deepspeech.cc

@@ -39,25 +39,9 @@
 //TODO: infer batch size from model/use dynamic batch size
 constexpr unsigned int BATCH_SIZE = 1;
 
-//TODO: use dynamic sample rate
+constexpr unsigned int DEFAULT_SAMPLE_RATE = 16000;
-constexpr unsigned int SAMPLE_RATE = 16000;
+constexpr unsigned int DEFAULT_WINDOW_LENGTH = DEFAULT_SAMPLE_RATE * 0.032;
-
+constexpr unsigned int DEFAULT_WINDOW_STEP = DEFAULT_SAMPLE_RATE * 0.02;
-constexpr float AUDIO_WIN_LEN = 0.032f;
-constexpr float AUDIO_WIN_STEP = 0.02f;
-constexpr unsigned int AUDIO_WIN_LEN_SAMPLES = (unsigned int)(AUDIO_WIN_LEN * SAMPLE_RATE);
-constexpr unsigned int AUDIO_WIN_STEP_SAMPLES = (unsigned int)(AUDIO_WIN_STEP * SAMPLE_RATE);
-
-constexpr size_t WINDOW_SIZE = AUDIO_WIN_LEN * SAMPLE_RATE;
-
-std::array<float, WINDOW_SIZE> calc_hamming_window() {
-  std::array<float, WINDOW_SIZE> a{0};
-  for (int i = 0; i < WINDOW_SIZE; ++i) {
-    a[i] = 0.54 - 0.46 * std::cos(2*M_PI*i/(WINDOW_SIZE-1));
-  }
-  return a;
-}
-
-std::array<float, WINDOW_SIZE> hamming_window = calc_hamming_window();
 
 #ifndef USE_TFLITE
   using namespace tensorflow;
@@ -134,6 +118,9 @@ struct ModelState {
   unsigned int n_context;
   unsigned int n_features;
   unsigned int mfcc_feats_per_timestep;
+  unsigned int sample_rate;
+  unsigned int audio_win_len;
+  unsigned int audio_win_step;
 
 #ifdef USE_TFLITE
   size_t previous_state_size;
@@ -220,6 +207,9 @@ ModelState::ModelState()
   , n_context(-1)
   , n_features(-1)
   , mfcc_feats_per_timestep(-1)
+  , sample_rate(DEFAULT_SAMPLE_RATE)
+  , audio_win_len(DEFAULT_WINDOW_LENGTH)
+  , audio_win_step(DEFAULT_WINDOW_STEP)
 #ifdef USE_TFLITE
   , previous_state_size(0)
   , previous_state_c_(nullptr)
@@ -258,7 +248,7 @@ StreamingState::feedAudioContent(const short* buffer,
 {
   // Consume all the data that was passed in, processing full buffers if needed
   while (buffer_size > 0) {
-    while (buffer_size > 0 && audio_buffer.size() < AUDIO_WIN_LEN_SAMPLES) {
+    while (buffer_size > 0 && audio_buffer.size() < model->audio_win_len) {
       // Convert i16 sample into f32
       float multiplier = 1.0f / (1 << 15);
       audio_buffer.push_back((float)(*buffer) * multiplier);
@@ -267,10 +257,10 @@ StreamingState::feedAudioContent(const short* buffer,
     }
 
     // If the buffer is full, process and shift it
-    if (audio_buffer.size() == AUDIO_WIN_LEN_SAMPLES) {
+    if (audio_buffer.size() == model->audio_win_len) {
       processAudioWindow(audio_buffer);
       // Shift data by one step
-      shift_buffer_left(audio_buffer, AUDIO_WIN_STEP_SAMPLES);
+      shift_buffer_left(audio_buffer, model->audio_win_step);
     }
 
     // Repeat until buffer empty
@@ -461,13 +451,13 @@ void
 ModelState::compute_mfcc(const vector<float>& samples, vector<float>& mfcc_output)
 {
 #ifndef USE_TFLITE
-  Tensor input(DT_FLOAT, TensorShape({AUDIO_WIN_LEN_SAMPLES}));
+  Tensor input(DT_FLOAT, TensorShape({audio_win_len}));
   auto input_mapped = input.flat<float>();
   int i;
   for (i = 0; i < samples.size(); ++i) {
     input_mapped(i) = samples[i];
   }
-  for (; i < AUDIO_WIN_LEN_SAMPLES; ++i) {
+  for (; i < audio_win_len; ++i) {
     input_mapped(i) = 0.f;
   }
 
@@ -556,8 +546,8 @@ ModelState::decode_metadata(const vector<float>& logits)
   for (int i = 0; i < out[0].tokens.size(); ++i) {
     metadata->items[i].character = (char*)alphabet->StringFromLabel(out[0].tokens[i]).c_str(); 
     metadata->items[i].timestep = out[0].timesteps[i]; 
-    metadata->items[i].start_time = static_cast<float>(out[0].timesteps[i] * AUDIO_WIN_STEP);
+    metadata->items[i].start_time = out[0].timesteps[i] * ((float)audio_win_step / sample_rate);
-    
+
     if (metadata->items[i].start_time < 0) {
       metadata->items[i].start_time = 0;
     }
@@ -700,6 +690,13 @@ DS_CreateModel(const char* aModelPath,
                   << std::endl;
         return DS_ERR_INVALID_ALPHABET;
       }
+    } else if (node.name() == "model_metadata") {
+      int sample_rate = node.attr().at("sample_rate").i();
+      model->sample_rate = sample_rate;
+      int win_len_ms = node.attr().at("feature_win_len").i();
+      int win_step_ms = node.attr().at("feature_win_step").i();
+      model->audio_win_len = sample_rate * (win_len_ms / 1000.0);
+      model->audio_win_step = sample_rate * (win_step_ms / 1000.0);
     }
   }
 
@@ -833,7 +830,7 @@ DS_SetupStream(ModelState* aCtx,
 
   ctx->accumulated_logits.reserve(aPreAllocFrames * BATCH_SIZE * num_classes);
 
-  ctx->audio_buffer.reserve(AUDIO_WIN_LEN_SAMPLES);
+  ctx->audio_buffer.reserve(aCtx->audio_win_len);
   ctx->mfcc_buffer.reserve(aCtx->mfcc_feats_per_timestep);
   ctx->mfcc_buffer.resize(aCtx->n_features*aCtx->n_context, 0.f);
   ctx->batch_buffer.reserve(aCtx->n_steps * aCtx->mfcc_feats_per_timestep);

util/config.py

@@ -92,6 +92,12 @@ def initialize_globals():
     # Units in the sixth layer = number of characters in the target language plus one
     c.n_hidden_6 = c.alphabet.size() + 1 # +1 for CTC blank label
 
+    # Size of audio window in samples
+    c.audio_window_samples = FLAGS.audio_sample_rate * (FLAGS.feature_win_len / 1000)
+
+    # Stride for feature computations in samples
+    c.audio_step_samples = FLAGS.audio_sample_rate * (FLAGS.feature_win_step / 1000)
+
     if FLAGS.one_shot_infer:
         if not os.path.exists(FLAGS.one_shot_infer):
             log_error('Path specified in --one_shot_infer is not a valid file.')

util/feeding.py

@@ -27,7 +27,10 @@ def read_csvs(csv_files):
 
 
 def samples_to_mfccs(samples, sample_rate):
-    spectrogram = contrib_audio.audio_spectrogram(samples, window_size=512, stride=320, magnitude_squared=True)
+    spectrogram = contrib_audio.audio_spectrogram(samples,
+                                                  window_size=Config.audio_window_samples,
+                                                  stride=Config.audio_step_samples,
+                                                  magnitude_squared=True)
     mfccs = contrib_audio.mfcc(spectrogram, sample_rate, dct_coefficient_count=Config.n_input)
     mfccs = tf.reshape(mfccs, [-1, Config.n_input])
 

util/flags.py

@@ -19,6 +19,10 @@ def create_flags():
     tf.app.flags.DEFINE_string  ('dev_cached_features_path',        '',          'comma separated list of files specifying the dataset used for validation. multiple files will get merged')
     tf.app.flags.DEFINE_string  ('test_cached_features_path',       '',          'comma separated list of files specifying the dataset used for testing. multiple files will get merged')
 
+    tf.app.flags.DEFINE_integer ('feature_win_len',  32,          'feature extraction audio window length in milliseconds')
+    tf.app.flags.DEFINE_integer ('feature_win_step', 20,          'feature extraction window step length in milliseconds')
+    tf.app.flags.DEFINE_integer ('audio_sample_rate',16000,       'sample rate value expected by model')
+
     # Global Constants
     # ================
 
@@ -73,6 +77,7 @@ def create_flags():
     tf.app.flags.DEFINE_boolean ('export_tflite',    False,       'export a graph ready for TF Lite engine')
     tf.app.flags.DEFINE_boolean ('use_seq_length',   True,        'have sequence_length in the exported graph (will make tfcompile unhappy)')
     tf.app.flags.DEFINE_integer ('n_steps',          16,          'how many timesteps to process at once by the export graph, higher values mean more latency')
+    tf.app.flags.DEFINE_string  ('export_language',  '',          'language the model was trained on e.g. "en" or "English". Gets embedded into exported model.')
 
     # Reporting