mirror of https://github.com/mstorsjo/fdk-aac.git
m4a-dec: save to 32-bit wav if aac: INT_PCM==LONG (SAMPLE_BITS==32)
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parent
7bcf0d3d55
commit
aed77c2f1b
17
m4a-dec.c
17
m4a-dec.c
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@ -35,7 +35,7 @@ int main(int argc, char *argv[]) {
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void *wav = NULL;
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int output_size, ret;
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uint8_t *output_buf;
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int16_t *decode_buf;
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INT_PCM *decode_buf;
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HANDLE_AACDECODER handle;
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AAC_DECODER_ERROR err;
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unsigned frame_size = 0, i;
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@ -82,9 +82,9 @@ int main(int argc, char *argv[]) {
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return 1;
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}
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output_size = 8*2*2048;
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output_size = 8*sizeof(INT_PCM)*2048;
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output_buf = (uint8_t*) malloc(output_size);
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decode_buf = (int16_t*) malloc(output_size);
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decode_buf = (INT_PCM*) malloc(output_size);
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while (1) {
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UINT valid;
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@ -126,7 +126,7 @@ int main(int argc, char *argv[]) {
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}
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frame_size = info->frameSize * info->numChannels;
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// Note, this probably doesn't return channels > 2 in the right order for wav
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wav = wav_write_open(outfile, info->sampleRate, 16, info->numChannels);
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wav = wav_write_open(outfile, info->sampleRate, sizeof(INT_PCM)*8, info->numChannels);
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if (!wav) {
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perror(outfile);
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status = 1;
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@ -134,11 +134,12 @@ int main(int argc, char *argv[]) {
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}
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}
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for (i = 0; i < frame_size; i++) {
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uint8_t* out = &output_buf[2*i];
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out[0] = decode_buf[i] & 0xff;
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out[1] = decode_buf[i] >> 8;
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uint8_t* out = &output_buf[sizeof(INT_PCM)*i];
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unsigned j;
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for (j = 0; j < sizeof(INT_PCM); j++)
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out[j] = (uint8_t)(decode_buf[i] >> (8*j));
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}
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wav_write_data(wav, output_buf, 2*frame_size);
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wav_write_data(wav, output_buf, sizeof(INT_PCM)*frame_size);
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}
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free(output_buf);
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free(decode_buf);
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