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POSaving audio input of Android Stock speech recognition engine
 

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  1. POSaving audio input of Android Stock speech recognition engine
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    <p>I am trying to save in a file the audio data listened by speech recognition service of android.</p> <p>Actually I implement <code>RecognitionListener</code> as explained here: <a href="https://stackoverflow.com/questions/5913773/speech-to-text-on-android">Speech to Text on Android</a></p> <p>save the data into a buffer as illustrated here: <a href="https://stackoverflow.com/questions/5925657/capturing-audio-sent-to-googles-speech-recognition-server">Capturing audio sent to Google&#39;s speech recognition server</a></p> <p>and write the buffer to a Wav file, as in here. <a href="https://stackoverflow.com/questions/7336570/android-record-raw-bytes-into-wave-file-for-http-streaming">Android Record raw bytes into WAVE file for Http Streaming</a></p> <p>My problem is how to get appropriate audio settings to save in the wav file's headers. In fact when I play the wav file only hear strange noise, with this parameters,</p> <pre><code>short nChannels=2;// audio channels int sRate=44100; // Sample rate short bSamples = 16;// byteSample </code></pre> <p>or nothing with this:</p> <pre><code>short nChannels=1;// audio channels int sRate=8000; // Sample rate short bSamples = 16;// byteSample </code></pre> <p>What is confusing is that looking at parameters of the speech recognition task from logcat I find first <strong>Set PLAYBACK sample rate to 44100 HZ</strong>:</p> <pre><code> 12-20 14:41:34.007: DEBUG/AudioHardwareALSA(2364): Set PLAYBACK PCM format to S16_LE (Signed 16 bit Little Endian) 12-20 14:41:34.007: DEBUG/AudioHardwareALSA(2364): Using 2 channels for PLAYBACK. 12-20 14:41:34.007: DEBUG/AudioHardwareALSA(2364): Set PLAYBACK sample rate to 44100 HZ 12-20 14:41:34.007: DEBUG/AudioHardwareALSA(2364): Buffer size: 2048 12-20 14:41:34.007: DEBUG/AudioHardwareALSA(2364): Latency: 46439 </code></pre> <p>and then <strong>aInfo.SampleRate = 8000</strong> when it plays the file to send to google server:</p> <pre><code> 12-20 14:41:36.152: DEBUG/(2364): PV_Wav_Parser::InitWavParser 12-20 14:41:36.152: DEBUG/(2364): File open Succes 12-20 14:41:36.152: DEBUG/(2364): File SEEK End Succes ... 12-20 14:41:36.152: DEBUG/(2364): PV_Wav_Parser::ReadData 12-20 14:41:36.152: DEBUG/(2364): Data Read buff = RIFF? 12-20 14:41:36.152: DEBUG/(2364): Data Read = RIFF? 12-20 14:41:36.152: DEBUG/(2364): PV_Wav_Parser::ReadData 12-20 14:41:36.152: DEBUG/(2364): Data Read buff = fmt ... 12-20 14:41:36.152: DEBUG/(2364): PVWAVPARSER_OK 12-20 14:41:36.156: DEBUG/(2364): aInfo.AudioFormat = 1 12-20 14:41:36.156: DEBUG/(2364): aInfo.NumChannels = 1 12-20 14:41:36.156: DEBUG/(2364): aInfo.SampleRate = 8000 12-20 14:41:36.156: DEBUG/(2364): aInfo.ByteRate = 16000 12-20 14:41:36.156: DEBUG/(2364): aInfo.BlockAlign = 2 12-20 14:41:36.156: DEBUG/(2364): aInfo.BitsPerSample = 16 12-20 14:41:36.156: DEBUG/(2364): aInfo.BytesPerSample = 2 12-20 14:41:36.156: DEBUG/(2364): aInfo.NumSamples = 2258 </code></pre> <p>So, how can I find out the right parameters to save the audio buffer in a good wav audio file?</p>
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    1. CODid you ever find your solution?
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    2. COseems like you have gotten the farthest regarding doing this. mmmx, were you able to solve this problem?
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