Add a test for interleaved/concurrent streams with a single model instance
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#!/usr/bin/env python
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# -*- coding: utf-8 -*-
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from __future__ import absolute_import, division, print_function
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import argparse
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import numpy as np
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import wave
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from deepspeech import Model
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# These constants control the beam search decoder
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# Beam width used in the CTC decoder when building candidate transcriptions
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BEAM_WIDTH = 500
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# The alpha hyperparameter of the CTC decoder. Language Model weight
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LM_ALPHA = 0.75
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# The beta hyperparameter of the CTC decoder. Word insertion bonus.
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LM_BETA = 1.85
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# These constants are tied to the shape of the graph used (changing them changes
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# the geometry of the first layer), so make sure you use the same constants that
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# were used during training
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# Number of MFCC features to use
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N_FEATURES = 26
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# Size of the context window used for producing timesteps in the input vector
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N_CONTEXT = 9
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def main():
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parser = argparse.ArgumentParser(description='Running DeepSpeech inference.')
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parser.add_argument('--model', required=True,
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help='Path to the model (protocol buffer binary file)')
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parser.add_argument('--alphabet', required=True,
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help='Path to the configuration file specifying the alphabet used by the network')
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parser.add_argument('--lm', nargs='?',
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help='Path to the language model binary file')
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parser.add_argument('--trie', nargs='?',
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help='Path to the language model trie file created with native_client/generate_trie')
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parser.add_argument('--audio1', required=True,
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help='First audio file to use in interleaved streams')
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parser.add_argument('--audio2', required=True,
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help='Second audio file to use in interleaved streams')
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args = parser.parse_args()
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ds = Model(args.model, N_FEATURES, N_CONTEXT, args.alphabet, BEAM_WIDTH)
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if args.lm and args.trie:
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ds.enableDecoderWithLM(args.alphabet, args.lm, args.trie, LM_ALPHA, LM_BETA)
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with wave.open(args.audio1, 'rb') as fin:
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fs1 = fin.getframerate()
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audio1 = np.frombuffer(fin.readframes(fin.getnframes()), np.int16)
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with wave.open(args.audio2, 'rb') as fin:
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fs2 = fin.getframerate()
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audio2 = np.frombuffer(fin.readframes(fin.getnframes()), np.int16)
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stream1 = ds.setupStream(sample_rate=fs1)
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stream2 = ds.setupStream(sample_rate=fs2)
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splits1 = np.array_split(audio1, 10)
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splits2 = np.array_split(audio2, 10)
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for part1, part2 in zip(splits1, splits2):
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ds.feedAudioContent(stream1, part1)
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ds.feedAudioContent(stream2, part2)
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print(ds.finishStream(stream1))
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print(ds.finishStream(stream2))
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if __name__ == '__main__':
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main()
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@ -39,4 +39,6 @@ LD_LIBRARY_PATH=${PY37_LDPATH}:$LD_LIBRARY_PATH pip install --verbose --only-bin
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run_prod_inference_tests
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run_prod_concurrent_stream_tests
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virtualenv_deactivate "${pyver}" "${PYENV_NAME}"
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@ -419,6 +419,26 @@ run_all_inference_tests()
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assert_correct_warning_upsampling "${phrase_pbmodel_withlm_mono_8k}"
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}
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run_prod_concurrent_stream_tests()
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{
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set +e
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output=$(python ${TASKCLUSTER_TMP_DIR}/test_sources/concurrent_streams.py \
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--model ${TASKCLUSTER_TMP_DIR}/${model_name_mmap} \
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--alphabet ${TASKCLUSTER_TMP_DIR}/alphabet.txt \
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--lm ${TASKCLUSTER_TMP_DIR}/lm.binary \
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--trie ${TASKCLUSTER_TMP_DIR}/trie \
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--audio1 ${TASKCLUSTER_TMP_DIR}/LDC93S1.wav \
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--audio2 ${TASKCLUSTER_TMP_DIR}/new-home-in-the-stars-16k.wav 2>/dev/null)
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status=$?
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set -e
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output1=$(echo ${output} | head -n 1)
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output2=$(echo ${output} | tail -n 1)
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assert_correct_ldc93s1_prodmodel "${output1}" "${status}"
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assert_correct_inference "${output2}" "i must find a new home in the stars" "${status}"
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}
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run_prod_inference_tests()
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{
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set +e
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@ -540,6 +560,7 @@ download_data()
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cp ${DS_ROOT_TASK}/DeepSpeech/ds/data/alphabet.txt ${TASKCLUSTER_TMP_DIR}/alphabet.txt
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cp ${DS_ROOT_TASK}/DeepSpeech/ds/data/smoke_test/vocab.pruned.lm ${TASKCLUSTER_TMP_DIR}/lm.binary
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cp ${DS_ROOT_TASK}/DeepSpeech/ds/data/smoke_test/vocab.trie ${TASKCLUSTER_TMP_DIR}/trie
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cp -R ${DS_ROOT_TASK}/DeepSpeech/ds/native_client/test ${TASKCLUSTER_TMP_DIR}/test_sources
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}
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download_material()
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