AndroidAudioConverter
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Convert pcm
Hi, im recording audio track in android with AudioRecord object instance. sample rate = 16 000, channel config = mono_in format = ENCODING_PCM_16BIT
I tried to convert recorded file into aac, m4a, flac. Every time conversion is failed. Is it actually possible to convert from poor pcm? P.S. recorded file extension *.pcm (tried *.ogg)
In order to do this I first had to convert PCM to WAV, as follows (found somewhere on SO):
private int frequency = 44100;
private mBufferSize = AudioRecord.getMinBufferSize(frequency, AudioFormat.CHANNEL_IN_MONO, AudioFormat.ENCODING_PCM_16BIT);
private static final int RECORDER_BPP = 16;
private void copyWaveFile(String inFilename /*pcm file*/, String outFilename /*wav file*/) {
FileInputStream in;
FileOutputStream out;
long totalAudioLen;
long totalDataLen;
long longSampleRate = frequency;
int channels = 1;
long byteRate = RECORDER_BPP * frequency * channels/8;
byte[] data = new byte[mBufferSize];
try {
in = getActivity().openFileInput(inFilename);
out = getActivity().openFileOutput(outFilename, Context.MODE_PRIVATE);
totalAudioLen = in.getChannel().size();
totalDataLen = totalAudioLen + 36;
writeWaveFileHeader(out, totalAudioLen, totalDataLen, longSampleRate, channels, byteRate);
while(in.read(data) != -1){
out.write(data);
}
in.close();
out.close();
} catch (FileNotFoundException e) {
e.printStackTrace();
} catch (IOException e) {
e.printStackTrace();
}
}
private void writeWaveFileHeader(
FileOutputStream out, long totalAudioLen,
long totalDataLen, long longSampleRate, int channels,
long byteRate) throws IOException {
byte[] header = new byte[44];
header[0] = 'R'; // RIFF/WAVE header
header[1] = 'I';
header[2] = 'F';
header[3] = 'F';
header[4] = (byte) (totalDataLen & 0xff);
header[5] = (byte) ((totalDataLen >> 8) & 0xff);
header[6] = (byte) ((totalDataLen >> 16) & 0xff);
header[7] = (byte) ((totalDataLen >> 24) & 0xff);
header[8] = 'W';
header[9] = 'A';
header[10] = 'V';
header[11] = 'E';
header[12] = 'f'; // 'fmt ' chunk
header[13] = 'm';
header[14] = 't';
header[15] = ' ';
header[16] = 16; // 4 bytes: size of 'fmt ' chunk
header[17] = 0;
header[18] = 0;
header[19] = 0;
header[20] = 1; // format = 1
header[21] = 0;
header[22] = (byte) channels;
header[23] = 0;
header[24] = (byte) (longSampleRate & 0xff);
header[25] = (byte) ((longSampleRate >> 8) & 0xff);
header[26] = (byte) ((longSampleRate >> 16) & 0xff);
header[27] = (byte) ((longSampleRate >> 24) & 0xff);
header[28] = (byte) (byteRate & 0xff);
header[29] = (byte) ((byteRate >> 8) & 0xff);
header[30] = (byte) ((byteRate >> 16) & 0xff);
header[31] = (byte) ((byteRate >> 24) & 0xff);
header[32] = (byte) (channels * 16 / 8); // block align
header[33] = 0;
header[34] = RECORDER_BPP; // bits per sample
header[35] = 0;
header[36] = 'd';
header[37] = 'a';
header[38] = 't';
header[39] = 'a';
header[40] = (byte) (totalAudioLen & 0xff);
header[41] = (byte) ((totalAudioLen >> 8) & 0xff);
header[42] = (byte) ((totalAudioLen >> 16) & 0xff);
header[43] = (byte) ((totalAudioLen >> 24) & 0xff);
out.write(header, 0, 44);
}