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authorBrandon Low <lostlogic@rockbox.org>2006-04-22 14:40:13 +0000
committerBrandon Low <lostlogic@rockbox.org>2006-04-22 14:40:13 +0000
commitf3bc1efc49826454a895c1c1f46fbe2cfd23f492 (patch)
tree676d501a9e75a9420f07a13f1d7fc1c5820b432a /apps/codecs
parentb5991b27cafa8cc8bd2e86e04110beec34a587e0 (diff)
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First commit of reworking voice to be mroe stable on swcodec
git-svn-id: svn://svn.rockbox.org/rockbox/trunk@9758 a1c6a512-1295-4272-9138-f99709370657
Diffstat (limited to 'apps/codecs')
-rw-r--r--apps/codecs/a52.c5
-rw-r--r--apps/codecs/aac.c14
-rw-r--r--apps/codecs/aiff.c24
-rw-r--r--apps/codecs/alac.c10
-rw-r--r--apps/codecs/flac.c11
-rw-r--r--apps/codecs/mpa.c8
-rw-r--r--apps/codecs/mpc.c14
-rw-r--r--apps/codecs/shorten.c3
-rw-r--r--apps/codecs/vorbis.c8
-rw-r--r--apps/codecs/wav.c34
-rw-r--r--apps/codecs/wavpack.c6
11 files changed, 69 insertions, 68 deletions
diff --git a/apps/codecs/a52.c b/apps/codecs/a52.c
index 8ad9d37d33..71e0fda432 100644
--- a/apps/codecs/a52.c
+++ b/apps/codecs/a52.c
@@ -141,7 +141,6 @@ enum codec_status codec_start(struct codec_api *api)
ci->memset(iedata, 0, iend - iedata);
#endif
- ci->configure(CODEC_DSP_ENABLE, (bool *)true);
ci->configure(DSP_DITHER, (bool *)false);
ci->configure(DSP_SET_STEREO_MODE, (long *)STEREO_NONINTERLEAVED);
ci->configure(DSP_SET_SAMPLE_DEPTH, (long *)28);
@@ -185,9 +184,11 @@ next_track:
a52_decode_data(filebuf, filebuf + n);
ci->advance_buffer(n);
}
+ retval = CODEC_OK;
+
if (ci->request_next_track())
goto next_track;
- retval = CODEC_OK;
+
exit:
a52_free(state);
return retval;
diff --git a/apps/codecs/aac.c b/apps/codecs/aac.c
index 02d460651c..532082ff05 100644
--- a/apps/codecs/aac.c
+++ b/apps/codecs/aac.c
@@ -63,7 +63,6 @@ enum codec_status codec_start(struct codec_api* api)
ci->configure(CODEC_SET_FILEBUF_CHUNKSIZE, (int *)(1024*16));
ci->configure(CODEC_SET_FILEBUF_WATERMARK, (int *)(1024*512));
- ci->configure(CODEC_DSP_ENABLE, (bool *)true);
ci->configure(DSP_DITHER, (bool *)false);
ci->configure(DSP_SET_STEREO_MODE, (int *)STEREO_NONINTERLEAVED);
ci->configure(DSP_SET_SAMPLE_DEPTH, (int *)(29));
@@ -88,7 +87,7 @@ next_track:
if (!qtmovie_read(&input_stream, &demux_res)) {
LOGF("FAAD: Error initialising file\n");
err = CODEC_ERROR;
- goto exit;
+ goto done;
}
/* initialise the sound converter */
@@ -98,7 +97,7 @@ next_track:
if (!hDecoder) {
LOGF("FAAD: Error opening decoder\n");
err = CODEC_ERROR;
- goto exit;
+ goto done;
}
NeAACDecConfigurationPtr conf = NeAACDecGetCurrentConfiguration(hDecoder);
@@ -112,7 +111,7 @@ next_track:
if (err) {
LOGF("FAAD: Error initialising decoder: %d, type=%d\n", err,hDecoder->object_type);
err = CODEC_ERROR;
- goto exit;
+ goto done;
}
ci->id3->frequency=s;
@@ -142,7 +141,7 @@ next_track:
&sample_byte_size)) {
LOGF("AAC: Error in get_sample_info\n");
err = CODEC_ERROR;
- goto exit;
+ goto done;
}
/* Request the required number of bytes from the input buffer */
@@ -156,7 +155,7 @@ next_track:
if (frameInfo.error > 0) {
LOGF("FAAD: decoding error \"%s\"\n", NeAACDecGetErrorMessage(frameInfo.error));
err = CODEC_ERROR;
- goto exit;
+ goto done;
}
/* Get the number of decoded samples */
@@ -182,13 +181,14 @@ next_track:
i++;
}
+ err = CODEC_OK;
+done:
LOGF("AAC: Decoded %d samples\n",samplesdone);
if (ci->request_next_track())
goto next_track;
- err = CODEC_OK;
exit:
return err;
}
diff --git a/apps/codecs/aiff.c b/apps/codecs/aiff.c
index 20d2dd3aa0..1e7adca220 100644
--- a/apps/codecs/aiff.c
+++ b/apps/codecs/aiff.c
@@ -98,11 +98,11 @@ next_track:
buf = ci->request_buffer(&n, 1024);
if (n < 44) {
i = CODEC_ERROR;
- goto exit;
+ goto done;
}
if ((memcmp(buf, "FORM", 4) != 0) || (memcmp(&buf[8], "AIFF", 4) != 0)) {
i = CODEC_ERROR;
- goto exit;
+ goto done;
}
buf += 12;
@@ -117,7 +117,7 @@ next_track:
if (i != 18) {
DEBUGF("CODEC_ERROR: 'COMM' chunk size=%lu != 18\n", i);
i = CODEC_ERROR;
- goto exit;
+ goto done;
}
/* num_channels */
num_channels = ((buf[8]<<8)|buf[9]);
@@ -130,7 +130,7 @@ next_track:
if (buf[16] != 0x40) {
DEBUGF("CODEC_ERROR: weird sampling rate (no @)\n", i);
i = CODEC_ERROR;
- goto exit;
+ goto done;
}
sample_rate = ((buf[18]<<24)|(buf[19]<<16)|(buf[20]<<8)|buf[21])+1;
sample_rate = sample_rate >> (16 + 14 - buf[17]);
@@ -140,7 +140,7 @@ next_track:
if (sample_size == 0) {
DEBUGF("CODEC_ERROR: unsupported chunk order\n");
i = CODEC_ERROR;
- goto exit;
+ goto done;
}
/* offset2snd */
offset2snd = ((buf[8]<<8)|buf[9]);
@@ -161,7 +161,7 @@ next_track:
if (n < (i + 8)) {
DEBUGF("CODEC_ERROR: AIFF header size > 1024\n");
i = CODEC_ERROR;
- goto exit;
+ goto done;
}
n -= i + 8;
} /* while 'SSND' */
@@ -169,21 +169,20 @@ next_track:
if (num_channels == 0) {
DEBUGF("CODEC_ERROR: 'COMM' chunk not found or 0-channels file\n");
i = CODEC_ERROR;
- goto exit;
+ goto done;
}
if (numbytes == 0) {
DEBUGF("CODEC_ERROR: 'SSND' chunk not found or has zero length\n");
i = CODEC_ERROR;
- goto exit;
+ goto done;
}
if (sample_size > 24) {
DEBUGF("CODEC_ERROR: PCM with more than 24 bits per sample "
"is unsupported\n");
i = CODEC_ERROR;
- goto exit;
+ goto done;
}
- ci->configure(CODEC_DSP_ENABLE, (bool *)true);
ci->configure(DSP_SET_FREQUENCY, (long *)(ci->id3->frequency));
ci->configure(DSP_SET_SAMPLE_DEPTH, (long *)28);
@@ -194,7 +193,7 @@ next_track:
} else {
DEBUGF("CODEC_ERROR: more than 2 channels unsupported\n");
i = CODEC_ERROR;
- goto exit;
+ goto done;
}
firstblockposn = 1024 - n;
@@ -277,11 +276,12 @@ next_track:
ci->set_elapsed(bytesdone*1000LL/avgbytespersec);
}
+ i = CODEC_OK;
+done:
if (ci->request_next_track())
goto next_track;
- i = CODEC_OK;
exit:
return i;
}
diff --git a/apps/codecs/alac.c b/apps/codecs/alac.c
index 73f45fc213..01172eed2b 100644
--- a/apps/codecs/alac.c
+++ b/apps/codecs/alac.c
@@ -64,7 +64,6 @@ enum codec_status codec_start(struct codec_api* api)
ci->configure(CODEC_SET_FILEBUF_WATERMARK, (int *)(1024*512));
ci->configure(CODEC_SET_FILEBUF_CHUNKSIZE, (int *)(1024*128));
- ci->configure(CODEC_DSP_ENABLE, (bool *)true);
ci->configure(DSP_DITHER, (bool *)false);
ci->configure(DSP_SET_STEREO_MODE, (int *)STEREO_NONINTERLEAVED);
ci->configure(DSP_SET_SAMPLE_DEPTH, (int *)(ALAC_OUTPUT_DEPTH-1));
@@ -89,7 +88,7 @@ enum codec_status codec_start(struct codec_api* api)
if (!qtmovie_read(&input_stream, &demux_res)) {
LOGF("ALAC: Error initialising file\n");
retval = CODEC_ERROR;
- goto exit;
+ goto done;
}
/* initialise the sound converter */
@@ -121,7 +120,7 @@ enum codec_status codec_start(struct codec_api* api)
&sample_byte_size)) {
LOGF("ALAC: Error in get_sample_info\n");
retval = CODEC_ERROR;
- goto exit;
+ goto done;
}
/* Request the required number of bytes from the input buffer */
@@ -129,7 +128,7 @@ enum codec_status codec_start(struct codec_api* api)
buffer=ci->request_buffer(&n,sample_byte_size);
if (n!=sample_byte_size) {
retval = CODEC_ERROR;
- goto exit;
+ goto done;
}
/* Decode one block - returned samples will be host-endian */
@@ -156,13 +155,14 @@ enum codec_status codec_start(struct codec_api* api)
i++;
}
+ retval = CODEC_OK;
+done:
LOGF("ALAC: Decoded %d samples\n",samplesdone);
if (ci->request_next_track())
goto next_track;
- retval = CODEC_OK;
exit:
return retval;
}
diff --git a/apps/codecs/flac.c b/apps/codecs/flac.c
index cc2ce636ab..5e392da8e7 100644
--- a/apps/codecs/flac.c
+++ b/apps/codecs/flac.c
@@ -224,7 +224,7 @@ enum codec_status codec_start(struct codec_api* api)
{
int8_t *buf;
FLACContext fc;
- uint32_t samplesdone;
+ uint32_t samplesdone = 0;
uint32_t elapsedtime;
size_t bytesleft;
int consumed;
@@ -244,7 +244,6 @@ enum codec_status codec_start(struct codec_api* api)
ci->configure(CODEC_SET_FILEBUF_WATERMARK, (int *)(1024*512));
ci->configure(CODEC_SET_FILEBUF_CHUNKSIZE, (int *)(1024*128));
- ci->configure(CODEC_DSP_ENABLE, (bool *)true);
ci->configure(DSP_DITHER, (bool *)false);
ci->configure(DSP_SET_STEREO_MODE, (long *)STEREO_NONINTERLEAVED);
ci->configure(DSP_SET_SAMPLE_DEPTH, (int *)(FLAC_OUTPUT_DEPTH-1));
@@ -260,7 +259,7 @@ enum codec_status codec_start(struct codec_api* api)
if (!flac_init(&fc,ci->id3->first_frame_offset)) {
LOGF("FLAC: Error initialising codec\n");
retval = CODEC_ERROR;
- goto exit;
+ goto done;
}
while (!*ci->taginfo_ready)
@@ -292,7 +291,7 @@ enum codec_status codec_start(struct codec_api* api)
bytesleft,ci->yield)) < 0) {
LOGF("FLAC: Frame %d, error %d\n",frame,res);
retval = CODEC_ERROR;
- goto exit;
+ goto done;
}
consumed=fc.gb.index/8;
frame++;
@@ -312,12 +311,14 @@ enum codec_status codec_start(struct codec_api* api)
buf = ci->request_buffer(&bytesleft, MAX_FRAMESIZE);
}
+ retval = CODEC_OK;
+
+done:
LOGF("FLAC: Decoded %d samples\n",samplesdone);
if (ci->request_next_track())
goto next_track;
- retval = CODEC_OK;
exit:
return retval;
}
diff --git a/apps/codecs/mpa.c b/apps/codecs/mpa.c
index 7219095591..51038c5244 100644
--- a/apps/codecs/mpa.c
+++ b/apps/codecs/mpa.c
@@ -91,7 +91,7 @@ void init_mad(void)
/* this is the codec entry point */
enum codec_status codec_start(struct codec_api *api)
{
- int status = CODEC_OK;
+ int status;
size_t size;
int file_end;
int frame_skip; /* samples to skip current frame */
@@ -110,7 +110,6 @@ enum codec_status codec_start(struct codec_api *api)
/* Create a decoder instance */
- ci->configure(CODEC_DSP_ENABLE, (bool *)true);
ci->configure(DSP_DITHER, (bool *)false);
ci->configure(DSP_SET_SAMPLE_DEPTH, (int *)(MAD_F_FRACBITS));
ci->configure(DSP_SET_CLIP_MIN, (int *)-MAD_F_ONE);
@@ -122,6 +121,7 @@ enum codec_status codec_start(struct codec_api *api)
* for gapless playback.
* Reinitializing seems to be necessary to avoid playback quircks when seeking. */
next_track:
+ status = CODEC_OK;
init_mad();
@@ -171,7 +171,7 @@ enum codec_status codec_start(struct codec_api *api)
ci->id3->first_frame_offset;
if (!ci->seek_buffer(newpos))
- goto next_track;
+ break;
ci->seek_complete();
init_mad();
}
@@ -192,7 +192,7 @@ enum codec_status codec_start(struct codec_api *api)
break;
/* Fill the buffer */
- if (stream.next_frame)
+ if (stream.next_frame && stream.next_frame != stream.this_frame)
ci->advance_buffer_loc((void *)stream.next_frame);
else
ci->advance_buffer(size);
diff --git a/apps/codecs/mpc.c b/apps/codecs/mpc.c
index 4201b2591c..ee639012b9 100644
--- a/apps/codecs/mpc.c
+++ b/apps/codecs/mpc.c
@@ -90,7 +90,6 @@ enum codec_status codec_start(struct codec_api *api)
ci->memset(iedata, 0, iend - iedata);
#endif
- ci->configure(CODEC_DSP_ENABLE, (bool *)true);
ci->configure(DSP_DITHER, (bool *)false);
ci->configure(DSP_SET_SAMPLE_DEPTH, (long *)(28));
ci->configure(CODEC_SET_FILEBUF_CHUNKSIZE, (long *)(1024*16));
@@ -113,7 +112,7 @@ next_track:
mpc_streaminfo_init(&info);
if (mpc_streaminfo_read(&info, &reader) != ERROR_CODE_OK) {
retval = CODEC_ERROR;
- goto exit;
+ goto done;
}
frequency = info.sample_freq;
ci->configure(DSP_SET_FREQUENCY, (long *)(long)info.sample_freq);
@@ -127,7 +126,7 @@ next_track:
ci->configure(DSP_SET_STEREO_MODE, (long *)STEREO_MONO);
else {
retval = CODEC_ERROR;
- goto exit;
+ goto done;
}
codec_set_replaygain(ci->id3);
@@ -135,7 +134,7 @@ next_track:
mpc_decoder_setup(&decoder, &reader);
if (!mpc_decoder_initialize(&decoder, &info)) {
retval = CODEC_ERROR;
- goto exit;
+ goto done;
}
/* This is the decoding loop. */
@@ -171,7 +170,7 @@ next_track:
ci->yield();
if (status == (unsigned)(-1)) { /* decode error */
retval = CODEC_ERROR;
- goto exit;
+ goto done;
} else {
while (!ci->pcmbuf_insert_split(sample_buffer,
sample_buffer + MPC_FRAME_LENGTH,
@@ -181,11 +180,12 @@ next_track:
ci->set_elapsed(samplesdone/(frequency/1000));
}
} while (status != 0);
-
+ retval = CODEC_OK;
+
+done:
if (ci->request_next_track())
goto next_track;
- retval = CODEC_OK;
exit:
return retval;
}
diff --git a/apps/codecs/shorten.c b/apps/codecs/shorten.c
index 03a08021a7..8d62a12f03 100644
--- a/apps/codecs/shorten.c
+++ b/apps/codecs/shorten.c
@@ -63,7 +63,6 @@ enum codec_status codec_start(struct codec_api* api)
ci->configure(CODEC_SET_FILEBUF_WATERMARK, (int *)(1024*512));
ci->configure(CODEC_SET_FILEBUF_CHUNKSIZE, (int *)(1024*128));
- ci->configure(CODEC_DSP_ENABLE, (bool *)true);
ci->configure(DSP_DITHER, (bool *)false);
ci->configure(DSP_SET_STEREO_MODE, (long *)STEREO_NONINTERLEAVED);
ci->configure(DSP_SET_SAMPLE_DEPTH, (int *)(SHN_OUTPUT_DEPTH-1));
@@ -146,7 +145,7 @@ seek_start:
if (res == FN_ERROR) {
LOGF("Shorten: shorten_decode_frames error (%d)\n", samplesdone);
- return CODEC_ERROR;
+ break;
} else {
/* Insert decoded samples in pcmbuf */
if (nsamples) {
diff --git a/apps/codecs/vorbis.c b/apps/codecs/vorbis.c
index a9a274500c..1900364a89 100644
--- a/apps/codecs/vorbis.c
+++ b/apps/codecs/vorbis.c
@@ -129,7 +129,6 @@ enum codec_status codec_start(struct codec_api *api)
rb->memset(iedata, 0, iend - iedata);
#endif
- rb->configure(CODEC_DSP_ENABLE, (bool *)true);
rb->configure(DSP_DITHER, (bool *)false);
rb->configure(DSP_SET_SAMPLE_DEPTH, (long *)24);
rb->configure(DSP_SET_CLIP_MAX, (long *)((1 << 24) - 1));
@@ -194,7 +193,7 @@ next_track:
} else {
//rb->logf("ov_open: %d", error);
error = CODEC_ERROR;
- goto exit;
+ goto done;
}
if (rb->id3->offset) {
@@ -224,7 +223,7 @@ next_track:
if (current_section != previous_section) {
if (!vorbis_set_codec_parameters(&vf)) {
error = CODEC_ERROR;
- goto exit;
+ goto done;
} else {
previous_section = current_section;
}
@@ -243,7 +242,9 @@ next_track:
rb->set_elapsed(ov_time_tell(&vf));
}
}
+ error = CODEC_OK;
+done:
if (rb->request_next_track()) {
/* Clean things up for the next track */
vf.dataoffsets = NULL;
@@ -255,7 +256,6 @@ next_track:
goto next_track;
}
- error = CODEC_OK;
exit:
return error;
}
diff --git a/apps/codecs/wav.c b/apps/codecs/wav.c
index c89d121c6a..d9be0a3420 100644
--- a/apps/codecs/wav.c
+++ b/apps/codecs/wav.c
@@ -242,7 +242,6 @@ enum codec_status codec_start(struct codec_api *api)
ci->memset(iedata, 0, iend - iedata);
#endif
- ci->configure(CODEC_DSP_ENABLE, (bool *)true);
ci->configure(DSP_SET_SAMPLE_DEPTH, (long *)28);
ci->configure(CODEC_SET_FILEBUF_WATERMARK, (int *)(1024*512));
ci->configure(CODEC_SET_FILEBUF_CHUNKSIZE, (int *)(1024*256));
@@ -261,11 +260,11 @@ next_track:
buf = ci->request_buffer(&n, 1024);
if (n < 44) {
i = CODEC_ERROR;
- goto exit;
+ goto done;
}
if ((memcmp(buf, "RIFF", 4) != 0) || (memcmp(&buf[8], "WAVE", 4) != 0)) {
i = CODEC_ERROR;
- goto exit;
+ goto done;
}
buf += 12;
@@ -281,7 +280,7 @@ next_track:
if (i < 16) {
DEBUGF("CODEC_ERROR: 'fmt ' chunk size=%lu < 16\n", i);
i = CODEC_ERROR;
- goto exit;
+ goto done;
}
/* wFormatTag */
formattag=buf[8]|(buf[9]<<8);
@@ -309,7 +308,7 @@ next_track:
DEBUGF("CODEC_ERROR: dvi_adpcm is missing "
"SamplesPerBlock value\n");
i = CODEC_ERROR;
- goto exit;
+ goto done;
}
samplesperblock = buf[26]|(buf[27]<<8);
} else if (formattag == WAVE_FORMAT_EXTENSIBLE) {
@@ -317,7 +316,7 @@ next_track:
DEBUGF("CODEC_ERROR: WAVE_FORMAT_EXTENSIBLE is "
"missing extension\n");
i = CODEC_ERROR;
- goto exit;
+ goto done;
}
/* wValidBitsPerSample */
bitspersample = buf[26]|(buf[27]<<8);
@@ -344,7 +343,7 @@ next_track:
if (n < (i + 8)) {
DEBUGF("CODEC_ERROR: WAVE header size > 1024\n");
i = CODEC_ERROR;
- goto exit;
+ goto done;
}
n -= i + 8;
}
@@ -352,12 +351,12 @@ next_track:
if (channels == 0) {
DEBUGF("CODEC_ERROR: 'fmt ' chunk not found or 0-channels file\n");
i = CODEC_ERROR;
- goto exit;
+ goto done;
}
if (numbytes == 0) {
DEBUGF("CODEC_ERROR: 'data' chunk not found or has zero-length\n");
i = CODEC_ERROR;
- goto exit;
+ goto done;
}
if (formattag != WAVE_FORMAT_PCM && totalsamples == 0) {
/* This is non-fatal for some formats */
@@ -368,7 +367,7 @@ next_track:
if (bitspersample != 8) {
DEBUGF("CODEC_ERROR: alaw and mulaw must have 8 bitspersample\n");
i = CODEC_ERROR;
- goto exit;
+ goto done;
}
bytespersample = channels;
}
@@ -376,13 +375,13 @@ next_track:
&& bitspersample != 4 && bitspersample != 3) {
DEBUGF("CODEC_ERROR: dvi_adpcm must have 3 or 4 bitspersample\n");
i = CODEC_ERROR;
- goto exit;
+ goto done;
}
if (formattag == WAVE_FORMAT_PCM && bitspersample > 32) {
DEBUGF("CODEC_ERROR: pcm with more than 32 bitspersample "
"is unsupported\n");
i = CODEC_ERROR;
- goto exit;
+ goto done;
}
ci->configure(DSP_SET_FREQUENCY, (long *)(ci->id3->frequency));
@@ -393,7 +392,7 @@ next_track:
} else {
DEBUGF("CODEC_ERROR: more than 2 channels\n");
i = CODEC_ERROR;
- goto exit;
+ goto done;
}
if (totalsamples == 0) {
@@ -406,7 +405,7 @@ next_track:
} else {
DEBUGF("CODEC_ERROR: cannot compute totalsamples\n");
i = CODEC_ERROR;
- goto exit;
+ goto done;
}
}
@@ -505,14 +504,14 @@ next_track:
samples + i*samplesperblock*channels,
&decodedsize) != CODEC_OK) {
i = CODEC_ERROR;
- goto exit;
+ goto done;
}
}
bufsize = nblocks*samplesperblock*channels*4;
} else {
DEBUGF("CODEC_ERROR: unsupported format %x\n", formattag);
i = CODEC_ERROR;
- goto exit;
+ goto done;
}
while (!ci->pcmbuf_insert((char *)samples, bufsize))
@@ -524,11 +523,12 @@ next_track:
endofstream = 1;
ci->set_elapsed(bytesdone*1000LL/avgbytespersec);
}
+ i = CODEC_OK;
+done:
if (ci->request_next_track())
goto next_track;
- i = CODEC_OK;
exit:
return i;
}
diff --git a/apps/codecs/wavpack.c b/apps/codecs/wavpack.c
index f864aa3ec6..1871b46f67 100644
--- a/apps/codecs/wavpack.c
+++ b/apps/codecs/wavpack.c
@@ -75,7 +75,6 @@ enum codec_status codec_start(struct codec_api* api)
while (!*ci->taginfo_ready && !ci->stop_codec)
ci->sleep(1);
- ci->configure(CODEC_DSP_ENABLE, (bool *)true);
ci->configure(DSP_SET_FREQUENCY, (long *)(ci->id3->frequency));
codec_set_replaygain(ci->id3);
@@ -84,7 +83,7 @@ enum codec_status codec_start(struct codec_api* api)
if (!wpc) {
retval = CODEC_ERROR;
- goto exit;
+ goto done;
}
bps = WavpackGetBytesPerSample (wpc);
@@ -143,11 +142,12 @@ enum codec_status codec_start(struct codec_api* api)
ci->set_elapsed (WavpackGetSampleIndex (wpc) / sr_100 * 10);
ci->yield ();
}
+ retval = CODEC_OK;
+done:
if (ci->request_next_track())
goto next_track;
- retval = CODEC_OK;
exit:
return retval;
}