mirror of https://github.com/mstorsjo/fdk-aac.git
TESTING: Convert aac-dec.c to feed bytes blindly
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57233c408d
commit
bf843ff3a6
63
aac-dec.c
63
aac-dec.c
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@ -55,53 +55,50 @@ int main(int argc, char *argv[]) {
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int n, i;
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UINT valid, packet_size;
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AAC_DECODER_ERROR err;
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n = fread(packet, 1, 7, in);
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if (n != 7)
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n = fread(packet, 1, 150, in);
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if (n <= 0)
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break;
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if (packet[0] != 0xff || (packet[1] & 0xf0) != 0xf0) {
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fprintf(stderr, "Not an ADTS packet\n");
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break;
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}
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packet_size = ((packet[3] & 0x03) << 11) | (packet[4] << 3) | (packet[5] >> 5);
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n = fread(packet + 7, 1, packet_size - 7, in);
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if (n != packet_size - 7) {
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fprintf(stderr, "Partial packet\n");
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break;
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}
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packet_size = n;
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valid = packet_size;
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err = aacDecoder_Fill(handle, &ptr, &packet_size, &valid);
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if (err != AAC_DEC_OK) {
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fprintf(stderr, "Fill failed: %x\n", err);
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break;
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}
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err = aacDecoder_DecodeFrame(handle, decode_buf, output_size / sizeof(INT_PCM), 0);
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if (err == AAC_DEC_NOT_ENOUGH_BITS)
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continue;
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if (err != AAC_DEC_OK) {
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fprintf(stderr, "Decode failed: %x\n", err);
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continue;
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if (valid != 0) {
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fprintf(stderr, "Unable to feed all %d input bytes, %d bytes left\n", n, valid);
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}
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if (!wav) {
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CStreamInfo *info = aacDecoder_GetStreamInfo(handle);
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if (!info || info->sampleRate <= 0) {
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fprintf(stderr, "No stream info\n");
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while (1) {
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err = aacDecoder_DecodeFrame(handle, decode_buf, output_size / sizeof(INT_PCM), 0);
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if (err == AAC_DEC_NOT_ENOUGH_BITS)
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break;
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if (err != AAC_DEC_OK) {
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fprintf(stderr, "Decode failed: %x\n", err);
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break;
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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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if (!wav) {
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perror(outfile);
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break;
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CStreamInfo *info = aacDecoder_GetStreamInfo(handle);
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if (!info || info->sampleRate <= 0) {
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fprintf(stderr, "No stream info\n");
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goto end;
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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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if (!wav) {
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perror(outfile);
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goto end;
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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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}
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wav_write_data(wav, output_buf, 2*frame_size);
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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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}
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wav_write_data(wav, output_buf, 2*frame_size);
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}
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end:
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free(output_buf);
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free(decode_buf);
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fclose(in);
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