forked from len0rd/rockbox
Code policed. Converted to generate internal s3.28 format directly (next
task is removing use of interleaved audio). Could not test ADPCM due to difficulty in finding files it could play, so tell me if it is broken. git-svn-id: svn://svn.rockbox.org/rockbox/trunk@9136 a1c6a512-1295-4272-9138-f99709370657
This commit is contained in:
Thom Johansen
parent
d8eaefe29a
commit
0968536c4b
1 changed files with 333 additions and 374 deletions
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@ -22,12 +22,15 @@
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CODEC_HEADER
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struct codec_api* rb;
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/* Macro that sign extends an unsigned byte */
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#define SE(x) ((int32_t)((int8_t)(x)))
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struct codec_api *rb;
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/* This codec support WAVE files with the following formats:
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* - PCM, up to 32 bits, supporting 32 bits playback when useful.
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* - ALAW and MULAW (16 bits compressed on 8 bits).
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* - DVI_ADPCM (16 bits compressed on 4 bits).
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* - DVI_ADPCM (16 bits compressed on 3 or 4 bits).
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*
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* For a good documentation on WAVE files, see:
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* http://www.tsp.ece.mcgill.ca/MMSP/Documents/AudioFormats/WAVE/WAVE.html
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@ -100,8 +103,6 @@ extern char iedata[];
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extern char iend[];
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#endif
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/* Those are lookup tables, so they should be in the idata section
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* (fast but small RAM on the coldfire processor) */
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static const int16_t alaw2linear16[256] ICONST_ATTR = {
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-5504, -5248, -6016, -5760, -4480, -4224, -4992,
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-4736, -7552, -7296, -8064, -7808, -6528, -6272,
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@ -182,7 +183,7 @@ static const int16_t ulaw2linear16[256] ICONST_ATTR = {
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24, 16, 8, 0
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};
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static const uint16_t dvi_adpcm_steptab[ 89 ] ICONST_ATTR = {
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static const uint16_t dvi_adpcm_steptab[89] ICONST_ATTR = {
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7, 8, 9, 10, 11, 12, 13, 14,
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16, 17, 19, 21, 23, 25, 28, 31,
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34, 37, 41, 45, 50, 55, 60, 66,
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@ -195,58 +196,59 @@ static const uint16_t dvi_adpcm_steptab[ 89 ] ICONST_ATTR = {
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7132, 7845, 8630, 9493, 10442, 11487, 12635, 13899,
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15289, 16818, 18500, 20350, 22385, 24623, 27086, 29794,
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32767 };
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static const int dvi_adpcm_indextab4[ 8 ] ICONST_ATTR = { -1, -1, -1, -1, 2, 4, 6, 8 };
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static const int dvi_adpcm_indextab3[ 4 ] ICONST_ATTR = { -1, -1, 1, 2 };
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static int16_t int16_samples[WAV_CHUNK_SIZE] IBSS_ATTR;
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static const int dvi_adpcm_indextab4[8] ICONST_ATTR = {
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-1, -1, -1, -1, 2, 4, 6, 8 };
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static const int dvi_adpcm_indextab3[4] ICONST_ATTR = { -1, -1, 1, 2 };
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static int32_t samples[WAV_CHUNK_SIZE] IBSS_ATTR;
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static enum codec_status
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decode_dvi_adpcm(struct codec_api* ci,
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decode_dvi_adpcm(struct codec_api *ci,
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const uint8_t *buf,
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int n,
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uint16_t channels, uint16_t bitspersample,
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int16_t *pcmout,
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int32_t *pcmout,
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size_t *pcmoutsize);
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/* this is the codec entry point */
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enum codec_status codec_start(struct codec_api* api)
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enum codec_status codec_start(struct codec_api *api)
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{
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struct codec_api* ci;
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struct codec_api *ci;
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uint32_t numbytes, bytesdone;
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uint32_t totalsamples = 0;
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uint16_t channels=0;
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uint16_t channels = 0;
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uint16_t samplesperblock = 0;
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int bytespersample=0;
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int bytespersample = 0;
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uint16_t bitspersample;
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uint32_t i;
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size_t n, wavbufsize;
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size_t n, bufsize;
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int endofstream;
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unsigned char* buf;
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uint16_t* wavbuf;
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unsigned char *buf;
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uint8_t *wavbuf;
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long chunksize;
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uint16_t formattag = 0;
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uint16_t blockalign = 0;
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uint32_t avgbytespersec = 0;
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off_t firstblockposn; /* position of the first block in file */
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int shortorlong = 1; /* do we output shorts (1) or longs (2)? */
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int32_t * const int32_samples = (int32_t*)int16_samples;
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/* Generic codec initialisation */
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rb = api;
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ci = api;
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#ifdef USE_IRAM
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ci->memcpy(iramstart, iramcopy, iramend-iramstart);
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ci->memcpy(iramstart, iramcopy, iramend - iramstart);
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ci->memset(iedata, 0, iend - iedata);
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#endif
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ci->configure(CODEC_DSP_ENABLE, (bool *)true);
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ci->configure(DSP_SET_SAMPLE_DEPTH, (long *)28);
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ci->configure(CODEC_SET_FILEBUF_WATERMARK, (int *)(1024*512));
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ci->configure(CODEC_SET_FILEBUF_CHUNKSIZE, (int *)(1024*256));
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ci->configure(DSP_DITHER, (bool *)false);
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next_track:
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next_track:
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if (codec_init(api)) {
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i = CODEC_ERROR;
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goto exit;
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@ -256,12 +258,12 @@ enum codec_status codec_start(struct codec_api* api)
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ci->yield();
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/* assume the WAV header is less than 1024 bytes */
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buf=ci->request_buffer((long *)&n,1024);
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if (n<44) {
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buf = ci->request_buffer((long *)&n, 1024);
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if (n < 44) {
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i = CODEC_ERROR;
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goto exit;
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}
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if ((memcmp(buf,"RIFF",4)!=0) || (memcmp(&buf[8],"WAVE",4)!=0)) {
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if ((memcmp(buf, "RIFF", 4) != 0) || (memcmp(&buf[8], "WAVE", 4) != 0)) {
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i = CODEC_ERROR;
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goto exit;
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}
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@ -272,12 +274,12 @@ enum codec_status codec_start(struct codec_api* api)
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numbytes = 0;
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totalsamples = 0;
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/* read until the data chunk, which should be last */
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while(numbytes == 0 && n >= 8) {
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while (numbytes == 0 && n >= 8) {
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/* chunkSize */
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i = (buf[4]|(buf[5]<<8)|(buf[6]<<16)|(buf[7]<<24));
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if (memcmp(buf,"fmt ",4)==0) {
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if (i<16) {
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DEBUGF("CODEC_ERROR: 'fmt ' chunk size=%lu < 16\n",i);
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if (memcmp(buf, "fmt ", 4) == 0) {
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if (i < 16) {
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DEBUGF("CODEC_ERROR: 'fmt ' chunk size=%lu < 16\n", i);
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i = CODEC_ERROR;
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goto exit;
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}
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@ -294,7 +296,7 @@ enum codec_status codec_start(struct codec_api* api)
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bitspersample=buf[22]|(buf[23]<<8);
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if (formattag != WAVE_FORMAT_PCM) {
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uint16_t size;
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if (i<18) {
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if (i < 18) {
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/* this is not a fatal error with some formats,
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* we'll see later if we can't decode it */
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DEBUGF("CODEC_WARNING: non-PCM WAVE (formattag=0x%x) "
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@ -310,8 +312,7 @@ enum codec_status codec_start(struct codec_api* api)
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goto exit;
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}
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samplesperblock = buf[26]|(buf[27]<<8);
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}
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else if (formattag == WAVE_FORMAT_EXTENSIBLE) {
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} else if (formattag == WAVE_FORMAT_EXTENSIBLE) {
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if (size < 22) {
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DEBUGF("CODEC_ERROR: WAVE_FORMAT_EXTENSIBLE is "
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"missing extension\n");
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@ -325,31 +326,27 @@ enum codec_status codec_start(struct codec_api* api)
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formattag = buf[32]|(buf[33]<<8);
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}
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}
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}
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else if (memcmp(buf,"data",4)==0) {
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numbytes=i;
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i=0; /* advance to the beginning of data */
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}
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else if (memcmp(buf,"fact",4)==0) {
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} else if (memcmp(buf, "data", 4) == 0) {
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numbytes = i;
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i = 0; /* advance to the beginning of data */
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} else if (memcmp(buf, "fact", 4) == 0) {
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/* dwSampleLength */
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if (i>=4) {
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if (i >= 4)
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totalsamples = (buf[8]|(buf[9]<<8)|(buf[10]<<16)|(buf[11]<<24));
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}
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}
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else {
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} else {
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DEBUGF("unknown WAVE chunk: '%c%c%c%c', size=%lu\n",
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buf[0], buf[1], buf[2], buf[3],i);
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buf[0], buf[1], buf[2], buf[3], i);
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}
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/* go to next chunk (even chunk sizes must be padded) */
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if (i & 0x01)
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i++;
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buf += i+8;
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if (n < (i+8)) {
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buf += i + 8;
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if (n < (i + 8)) {
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DEBUGF("CODEC_ERROR: WAVE header size > 1024\n");
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i = CODEC_ERROR;
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goto exit;
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}
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n -= i+8;
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n -= i + 8;
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}
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if (channels == 0) {
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@ -375,7 +372,7 @@ enum codec_status codec_start(struct codec_api* api)
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}
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bytespersample = channels;
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}
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if ( formattag == WAVE_FORMAT_DVI_ADPCM
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if (formattag == WAVE_FORMAT_DVI_ADPCM
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&& bitspersample != 4 && bitspersample != 3) {
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DEBUGF("CODEC_ERROR: dvi_adpcm must have 3 or 4 bitspersample\n");
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i = CODEC_ERROR;
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@ -388,23 +385,11 @@ enum codec_status codec_start(struct codec_api* api)
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goto exit;
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}
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ci->configure(CODEC_DSP_ENABLE, (bool *)true);
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ci->configure(DSP_SET_FREQUENCY, (long *)(ci->id3->frequency));
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if (bitspersample <= 16) {
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ci->configure(DSP_SET_SAMPLE_DEPTH, (int *)(16));
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} else {
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shortorlong = 2;
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ci->configure(DSP_DITHER, (bool *)false);
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ci->configure(DSP_SET_SAMPLE_DEPTH, (long *) (32));
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ci->configure(DSP_SET_CLIP_MAX, (long *) (2147483647));
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ci->configure(DSP_SET_CLIP_MIN, (long *) (-2147483647-1));
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}
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if (channels == 2) {
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ci->configure(DSP_SET_STEREO_MODE, (int *)STEREO_INTERLEAVED);
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ci->configure(DSP_SET_STEREO_MODE, (long *)STEREO_INTERLEAVED);
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} else if (channels == 1) {
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ci->configure(DSP_SET_STEREO_MODE, (int *)STEREO_MONO);
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ci->configure(DSP_SET_STEREO_MODE, (long *)STEREO_MONO);
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} else {
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DEBUGF("CODEC_ERROR: more than 2 channels\n");
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i = CODEC_ERROR;
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@ -416,40 +401,36 @@ enum codec_status codec_start(struct codec_api* api)
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formattag == WAVE_FORMAT_ALAW || formattag == WAVE_FORMAT_MULAW ||
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formattag == IBM_FORMAT_ALAW || formattag == IBM_FORMAT_MULAW) {
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/* for PCM and derived formats only */
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bytespersample=(((bitspersample-1)/8+1)*channels);
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totalsamples=numbytes/bytespersample;
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}
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else {
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bytespersample = (((bitspersample - 1)/8 + 1)*channels);
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totalsamples = numbytes/bytespersample;
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} else {
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DEBUGF("CODEC_ERROR: cannot compute totalsamples\n");
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i = CODEC_ERROR;
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goto exit;
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}
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}
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firstblockposn = (1024-n);
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firstblockposn = 1024 - n;
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ci->advance_buffer(firstblockposn);
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/* The main decoder loop */
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bytesdone=0;
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bytesdone = 0;
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ci->set_elapsed(0);
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endofstream=0;
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endofstream = 0;
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/* chunksize is computed so that one chunk is about 1/50s.
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* this make 4096 for 44.1kHz 16bits stereo.
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* It also has to be a multiple of blockalign */
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chunksize = (1 + avgbytespersec / (50*blockalign)) * blockalign;
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chunksize = (1 + avgbytespersec / (50*blockalign))*blockalign;
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/* check that the output buffer is big enough (convert to samplespersec,
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then round to the blockalign multiple below) */
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if (((uint64_t)chunksize*ci->id3->frequency*channels*shortorlong)
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/ (uint64_t)avgbytespersec >= WAV_CHUNK_SIZE) {
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chunksize = ((uint64_t)WAV_CHUNK_SIZE * avgbytespersec
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/ ((uint64_t)ci->id3->frequency * channels * shortorlong
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* blockalign)) * blockalign;
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if (((uint64_t)chunksize*ci->id3->frequency*channels*sizeof(long))
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/(uint64_t)avgbytespersec >= WAV_CHUNK_SIZE) {
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chunksize = ((uint64_t)WAV_CHUNK_SIZE*avgbytespersec
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/((uint64_t)ci->id3->frequency*channels*sizeof(long)
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*blockalign))*blockalign;
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}
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while (!endofstream) {
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uint8_t *wavbuf8;
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ci->yield();
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if (ci->stop_codec || ci->reload_codec) {
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break;
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@ -460,19 +441,17 @@ enum codec_status codec_start(struct codec_api* api)
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/* use avgbytespersec to round to the closest blockalign multiple,
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add firstblockposn. 64-bit casts to avoid overflows. */
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newpos = (((uint64_t)avgbytespersec * (ci->seek_time - 1))
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/ (1000LL*blockalign)) * blockalign;
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newpos = (((uint64_t)avgbytespersec*(ci->seek_time - 1))
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/ (1000LL*blockalign))*blockalign;
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if (newpos > numbytes)
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break;
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if (ci->seek_buffer(firstblockposn + newpos)) {
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if (ci->seek_buffer(firstblockposn + newpos))
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bytesdone = newpos;
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}
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ci->seek_complete();
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}
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wavbuf=ci->request_buffer((long *)&n,chunksize);
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wavbuf8 = (uint8_t*)wavbuf;
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wavbuf = (uint8_t *)ci->request_buffer((long *)&n, chunksize);
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if (n==0)
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if (n == 0)
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break; /* End of stream */
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if (bytesdone + n > numbytes) {
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@ -480,85 +459,68 @@ enum codec_status codec_start(struct codec_api* api)
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endofstream = 1;
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}
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wavbufsize = sizeof(int16_samples);
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if (formattag == WAVE_FORMAT_PCM) {
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if (bitspersample > 24) {
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for (i=0;i<n;i+=4) {
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int32_samples[i/4]=(int32_t)(wavbuf8[i]|(wavbuf8[i+1]<<8)|
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(wavbuf8[i+2]<<16)|(wavbuf8[i+3]<<24));
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for (i = 0; i < n; i += 4) {
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samples[i/4] = (wavbuf[i] >> 3)|
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(wavbuf[i + 1]<<5)|(wavbuf[i + 2]<<13)|
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(SE(wavbuf[i + 3])<<21);
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}
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wavbufsize = n;
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bufsize = n;
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} else if (bitspersample > 16) {
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for (i = 0; i < n; i += 3) {
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samples[i/3] = (wavbuf[i]<<5)|
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(wavbuf[i + 1]<<13)|(SE(wavbuf[i + 2])<<21);
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}
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else if (bitspersample > 16) {
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for (i=0;i<n;i+=3) {
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int32_samples[i/3]=(int32_t)((wavbuf8[i]<<8)|
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(wavbuf8[i+1]<<16)|(wavbuf8[i+2]<<24));
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bufsize = n*4/3;
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} else if (bitspersample > 8) {
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for (i = 0; i < n; i += 2) {
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samples[i/2] = (wavbuf[i]<<13)|(SE(wavbuf[i + 1])<<21);
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}
|
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wavbufsize = n*4/3;
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bufsize = n*2;
|
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} else {
|
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for (i = 0; i < n; i++) {
|
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samples[i] = (wavbuf[i] - 0x80)<<21;
|
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}
|
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else if (bitspersample > 8) {
|
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/* Byte-swap data. */
|
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for (i=0;i<n/2;i++) {
|
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int16_samples[i]=(int16_t)letoh16(wavbuf[i]);
|
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bufsize = n*4;
|
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}
|
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wavbufsize = n;
|
||||
}
|
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else {
|
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for (i=0;i<n;i++) {
|
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int16_samples[i] = (wavbuf8[i]<<8) - 0x8000;
|
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}
|
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wavbufsize = n*2;
|
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}
|
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}
|
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else if (formattag == WAVE_FORMAT_ALAW || formattag == IBM_FORMAT_ALAW) {
|
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for (i=0;i<n;i++) {
|
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int16_samples[i] = alaw2linear16[wavbuf8[i]];
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}
|
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wavbufsize = n*2;
|
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}
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else if (formattag == WAVE_FORMAT_MULAW || formattag == IBM_FORMAT_MULAW) {
|
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for (i=0;i<n;i++) {
|
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int16_samples[i] = ulaw2linear16[wavbuf8[i]];
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}
|
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wavbufsize = n*2;
|
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} else if (formattag == WAVE_FORMAT_ALAW
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|| formattag == IBM_FORMAT_ALAW) {
|
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for (i = 0; i < n; i++)
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samples[i] = alaw2linear16[wavbuf[i]] << 13;
|
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bufsize = n*4;
|
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} else if (formattag == WAVE_FORMAT_MULAW
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|| formattag == IBM_FORMAT_MULAW) {
|
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for (i = 0; i < n; i++)
|
||||
samples[i] = ulaw2linear16[wavbuf[i]] << 13;
|
||||
bufsize = n*4;
|
||||
}
|
||||
else if (formattag == WAVE_FORMAT_DVI_ADPCM) {
|
||||
unsigned int nblocks = chunksize/blockalign;
|
||||
|
||||
for (i=0; i<nblocks; i++) {
|
||||
for (i = 0; i < nblocks; i++) {
|
||||
size_t decodedsize = samplesperblock*channels;
|
||||
if (decode_dvi_adpcm(ci, ((uint8_t*)wavbuf)+i*blockalign,
|
||||
if (decode_dvi_adpcm(ci, ((uint8_t *)wavbuf) + i*blockalign,
|
||||
blockalign, channels, bitspersample,
|
||||
int16_samples+i*samplesperblock*channels,
|
||||
&decodedsize)
|
||||
!= CODEC_OK) {
|
||||
samples + i*samplesperblock*channels,
|
||||
&decodedsize) != CODEC_OK)
|
||||
i = CODEC_ERROR;
|
||||
goto exit;
|
||||
}
|
||||
if (decodedsize != samplesperblock) {
|
||||
i = CODEC_ERROR;
|
||||
goto exit;
|
||||
}
|
||||
}
|
||||
wavbufsize = nblocks*samplesperblock*channels*2;
|
||||
}
|
||||
else {
|
||||
bufsize = nblocks*samplesperblock*channels*2;
|
||||
} else {
|
||||
DEBUGF("CODEC_ERROR: unsupported format %x\n", formattag);
|
||||
i = CODEC_ERROR;
|
||||
goto exit;
|
||||
}
|
||||
|
||||
while (!ci->pcmbuf_insert((char*)int16_samples, wavbufsize)) {
|
||||
while (!ci->pcmbuf_insert((char *)samples, bufsize))
|
||||
ci->yield();
|
||||
}
|
||||
|
||||
ci->advance_buffer(n);
|
||||
bytesdone += n;
|
||||
if (bytesdone >= numbytes) {
|
||||
endofstream=1;
|
||||
}
|
||||
|
||||
if (bytesdone >= numbytes)
|
||||
endofstream = 1;
|
||||
ci->set_elapsed(bytesdone*1000LL/avgbytespersec);
|
||||
}
|
||||
|
||||
|
|
@ -571,11 +533,11 @@ exit:
|
|||
}
|
||||
|
||||
static enum codec_status
|
||||
decode_dvi_adpcm(struct codec_api* ci,
|
||||
decode_dvi_adpcm(struct codec_api *ci,
|
||||
const uint8_t *buf,
|
||||
int n,
|
||||
uint16_t channels, uint16_t bitspersample,
|
||||
int16_t *pcmout,
|
||||
int32_t *pcmout,
|
||||
size_t *pcmoutsize)
|
||||
{
|
||||
size_t nsamples = 0;
|
||||
|
|
@ -596,7 +558,7 @@ decode_dvi_adpcm(struct codec_api* ci,
|
|||
}
|
||||
|
||||
/* decode block header */
|
||||
for(c=0; c<channels && n>=4; c++) {
|
||||
for (c = 0; c < channels && n >= 4; c++) {
|
||||
/* decode + push first sample */
|
||||
sample[c] = (short)(buf[0]|(buf[1]<<8));/* need cast for sign-extend */
|
||||
pcmout[c] = sample[c];
|
||||
|
|
@ -612,8 +574,8 @@ decode_dvi_adpcm(struct codec_api* ci,
|
|||
n -= 4;
|
||||
}
|
||||
if (bitspersample == 4) {
|
||||
while( n>= channels*4 && (nsamples+8*channels) <= *pcmoutsize ) {
|
||||
for (c=0; c<channels; c++) {
|
||||
while (n>= channels*4 && (nsamples + 8*channels) <= *pcmoutsize) {
|
||||
for (c = 0; c < channels; c++) {
|
||||
samplecode[0][c] = buf[0]&0xf;
|
||||
samplecode[1][c] = buf[0]>>4;
|
||||
samplecode[2][c] = buf[1]&0xf;
|
||||
|
|
@ -625,9 +587,8 @@ decode_dvi_adpcm(struct codec_api* ci,
|
|||
buf += 4;
|
||||
n -= 4;
|
||||
}
|
||||
|
||||
for (i=0; i<8; i++) {
|
||||
for (c=0; c<channels; c++) {
|
||||
for (i = 0; i < 8; i++) {
|
||||
for (c = 0; c < channels; c++) {
|
||||
step = dvi_adpcm_steptab[stepindex[c]];
|
||||
codem = samplecode[i][c];
|
||||
code = codem & 0x07;
|
||||
|
|
@ -653,7 +614,7 @@ decode_dvi_adpcm(struct codec_api* ci,
|
|||
step = step >> 1;
|
||||
diff += step;
|
||||
#else
|
||||
diff = ((code+code+1) * step) >> 3; /* faster */
|
||||
diff = ((code + code + 1) * step) >> 3; /* faster */
|
||||
#endif
|
||||
/* check the sign bit */
|
||||
/* check for overflow and underflow errors */
|
||||
|
|
@ -661,25 +622,23 @@ decode_dvi_adpcm(struct codec_api* ci,
|
|||
sample[c] -= diff;
|
||||
if (sample[c] < -32768)
|
||||
sample[c] = -32768;
|
||||
}
|
||||
else {
|
||||
} else {
|
||||
sample[c] += diff;
|
||||
if (sample[c] > 32767)
|
||||
sample[c] = 32767;
|
||||
}
|
||||
/* output the new sample */
|
||||
pcmout[nsamples] = sample[c];
|
||||
pcmout[nsamples] = sample[c] << 13;
|
||||
nsamples++;
|
||||
}
|
||||
}
|
||||
}
|
||||
}
|
||||
else { /* bitspersample == 3 */
|
||||
while( n>= channels*12 && (nsamples+32*channels) <= *pcmoutsize) {
|
||||
for (c=0; c<channels; c++) {
|
||||
} else { /* bitspersample == 3 */
|
||||
while (n >= channels*12 && (nsamples + 32*channels) <= *pcmoutsize) {
|
||||
for (c = 0; c < channels; c++) {
|
||||
uint16_t bitstream = 0;
|
||||
int bitsread = 0;
|
||||
for (i=0; i<32 && n>0; i++) {
|
||||
for (i = 0; i < 32 && n > 0; i++) {
|
||||
if (bitsread < 3) {
|
||||
/* read 8 more bits */
|
||||
bitstream |= buf[0]<<bitsread;
|
||||
|
|
@ -698,8 +657,8 @@ decode_dvi_adpcm(struct codec_api* ci,
|
|||
}
|
||||
}
|
||||
|
||||
for (i=0; i<32; i++) {
|
||||
for (c=0; c<channels; c++) {
|
||||
for (i = 0; i < 32; i++) {
|
||||
for (c = 0; c < channels; c++) {
|
||||
step = dvi_adpcm_steptab[stepindex[c]];
|
||||
codem = samplecode[i][c];
|
||||
code = codem & 0x03;
|
||||
|
|
@ -722,7 +681,7 @@ decode_dvi_adpcm(struct codec_api* ci,
|
|||
step = step >> 1;
|
||||
diff += step;
|
||||
#else
|
||||
diff = ((code+code+1) * step) >> 3; /* faster */
|
||||
diff = ((code + code + 1) * step) >> 3; /* faster */
|
||||
#endif
|
||||
/* check the sign bit */
|
||||
/* check for overflow and underflow errors */
|
||||
|
|
@ -737,7 +696,7 @@ decode_dvi_adpcm(struct codec_api* ci,
|
|||
sample[c] = 32767;
|
||||
}
|
||||
/* output the new sample */
|
||||
pcmout[nsamples] = sample[c];
|
||||
pcmout[nsamples] = sample[c] << 13;
|
||||
nsamples++;
|
||||
}
|
||||
}
|
||||
|
|
@ -749,8 +708,8 @@ decode_dvi_adpcm(struct codec_api* ci,
|
|||
return CODEC_ERROR;
|
||||
}
|
||||
*pcmoutsize = nsamples;
|
||||
if (n!=0) {
|
||||
DEBUGF("decode_dvi_adpcm: n=%d unprocessed bytes\n",n);
|
||||
if (n != 0) {
|
||||
DEBUGF("decode_dvi_adpcm: n=%d unprocessed bytes\n", n);
|
||||
}
|
||||
return CODEC_OK;
|
||||
}
|
||||
|
|
|
|||
Loading…
Add table
Add a link
Reference in a new issue