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cosmetics: reindent ad_pcm.c

This commit is contained in:
Uoti Urpala 2010-05-15 05:45:02 +03:00
parent 9e02b2db40
commit 584b43e9f8

View File

@ -26,13 +26,12 @@
#include "libaf/af_format.h" #include "libaf/af_format.h"
#include "libaf/reorder_ch.h" #include "libaf/reorder_ch.h"
static const ad_info_t info = static const ad_info_t info = {
{ "Uncompressed PCM audio decoder",
"Uncompressed PCM audio decoder", "pcm",
"pcm", "Nick Kurshev",
"Nick Kurshev", "A'rpi",
"A'rpi", ""
""
}; };
struct ad_pcm_context { struct ad_pcm_context {
@ -42,53 +41,55 @@ struct ad_pcm_context {
LIBAD_EXTERN(pcm) LIBAD_EXTERN(pcm)
static int init(sh_audio_t *sh_audio) static int init(sh_audio_t * sh_audio)
{ {
WAVEFORMATEX *h=sh_audio->wf; WAVEFORMATEX *h = sh_audio->wf;
if (!h) if (!h)
return 0; return 0;
sh_audio->i_bps=h->nAvgBytesPerSec; sh_audio->i_bps = h->nAvgBytesPerSec;
sh_audio->channels=h->nChannels; sh_audio->channels = h->nChannels;
sh_audio->samplerate=h->nSamplesPerSec; sh_audio->samplerate = h->nSamplesPerSec;
sh_audio->samplesize=(h->wBitsPerSample+7)/8; sh_audio->samplesize = (h->wBitsPerSample + 7) / 8;
sh_audio->sample_format=AF_FORMAT_S16_LE; // default sh_audio->sample_format = AF_FORMAT_S16_LE; // default
switch(sh_audio->format){ /* hardware formats: */ switch (sh_audio->format) { /* hardware formats: */
case 0x0: case 0x0:
case 0x1: // Microsoft PCM case 0x1: // Microsoft PCM
case 0xfffe: // Extended case 0xfffe: // Extended
switch (sh_audio->samplesize) { switch (sh_audio->samplesize) {
case 1: sh_audio->sample_format=AF_FORMAT_U8; break; case 1: sh_audio->sample_format = AF_FORMAT_U8; break;
case 2: sh_audio->sample_format=AF_FORMAT_S16_LE; break; case 2: sh_audio->sample_format = AF_FORMAT_S16_LE; break;
case 3: sh_audio->sample_format=AF_FORMAT_S24_LE; break; case 3: sh_audio->sample_format = AF_FORMAT_S24_LE; break;
case 4: sh_audio->sample_format=AF_FORMAT_S32_LE; break; case 4: sh_audio->sample_format = AF_FORMAT_S32_LE; break;
} }
break; break;
case 0x3: // IEEE float case 0x3: // IEEE float
sh_audio->sample_format=AF_FORMAT_FLOAT_LE; sh_audio->sample_format = AF_FORMAT_FLOAT_LE;
break; break;
case 0x6: sh_audio->sample_format=AF_FORMAT_A_LAW;break; case 0x6: sh_audio->sample_format = AF_FORMAT_A_LAW; break;
case 0x7: sh_audio->sample_format=AF_FORMAT_MU_LAW;break; case 0x7: sh_audio->sample_format = AF_FORMAT_MU_LAW; break;
case 0x11: sh_audio->sample_format=AF_FORMAT_IMA_ADPCM;break; case 0x11: sh_audio->sample_format = AF_FORMAT_IMA_ADPCM; break;
case 0x50: sh_audio->sample_format=AF_FORMAT_MPEG2;break; case 0x50: sh_audio->sample_format = AF_FORMAT_MPEG2; break;
/* case 0x2000: sh_audio->sample_format=AFMT_AC3; */ /* case 0x2000: sh_audio->sample_format=AFMT_AC3; */
case 0x20776172: // 'raw ' case 0x20776172: // 'raw '
sh_audio->sample_format=AF_FORMAT_S16_BE; sh_audio->sample_format = AF_FORMAT_S16_BE;
if(sh_audio->samplesize==1) sh_audio->sample_format=AF_FORMAT_U8; if (sh_audio->samplesize == 1)
break; sh_audio->sample_format = AF_FORMAT_U8;
break;
case 0x736F7774: // 'twos' case 0x736F7774: // 'twos'
sh_audio->sample_format=AF_FORMAT_S16_BE; sh_audio->sample_format = AF_FORMAT_S16_BE;
// intended fall-through // intended fall-through
case 0x74776F73: // 'sowt' case 0x74776F73: // 'sowt'
if(sh_audio->samplesize==1) sh_audio->sample_format=AF_FORMAT_S8; if (sh_audio->samplesize == 1)
break; sh_audio->sample_format = AF_FORMAT_S8;
break;
case 0x32336c66: // 'fl32', bigendian float32 case 0x32336c66: // 'fl32', bigendian float32
sh_audio->sample_format=AF_FORMAT_FLOAT_BE; sh_audio->sample_format = AF_FORMAT_FLOAT_BE;
sh_audio->samplesize=4; sh_audio->samplesize = 4;
break; break;
case 0x666c3332: // '23lf', little endian float32, MPlayer internal fourCC case 0x666c3332: // '23lf', little endian float32, MPlayer internal fourCC
sh_audio->sample_format=AF_FORMAT_FLOAT_LE; sh_audio->sample_format = AF_FORMAT_FLOAT_LE;
sh_audio->samplesize=4; sh_audio->samplesize = 4;
break; break;
/* case 0x34366c66: // 'fl64', bigendian float64 /* case 0x34366c66: // 'fl64', bigendian float64
sh_audio->sample_format=AF_FORMAT_FLOAT_BE; sh_audio->sample_format=AF_FORMAT_FLOAT_BE;
sh_audio->samplesize=8; sh_audio->samplesize=8;
@ -98,93 +99,95 @@ static int init(sh_audio_t *sh_audio)
sh_audio->samplesize=8; sh_audio->samplesize=8;
break;*/ break;*/
case 0x34326e69: // 'in24', bigendian int24 case 0x34326e69: // 'in24', bigendian int24
sh_audio->sample_format=AF_FORMAT_S24_BE; sh_audio->sample_format = AF_FORMAT_S24_BE;
sh_audio->samplesize=3; sh_audio->samplesize = 3;
break; break;
case 0x696e3234: // '42ni', little endian int24, MPlayer internal fourCC case 0x696e3234: // '42ni', little endian int24, MPlayer internal fourCC
sh_audio->sample_format=AF_FORMAT_S24_LE; sh_audio->sample_format = AF_FORMAT_S24_LE;
sh_audio->samplesize=3; sh_audio->samplesize = 3;
break; break;
case 0x32336e69: // 'in32', bigendian int32 case 0x32336e69: // 'in32', bigendian int32
sh_audio->sample_format=AF_FORMAT_S32_BE; sh_audio->sample_format = AF_FORMAT_S32_BE;
sh_audio->samplesize=4; sh_audio->samplesize = 4;
break; break;
case 0x696e3332: // '23ni', little endian int32, MPlayer internal fourCC case 0x696e3332: // '23ni', little endian int32, MPlayer internal fourCC
sh_audio->sample_format=AF_FORMAT_S32_LE; sh_audio->sample_format = AF_FORMAT_S32_LE;
sh_audio->samplesize=4; sh_audio->samplesize = 4;
break; break;
default: if(sh_audio->samplesize!=2) sh_audio->sample_format=AF_FORMAT_U8; default:
} if (sh_audio->samplesize != 2)
if (!sh_audio->samplesize) // this would cause MPlayer to hang later sh_audio->sample_format = AF_FORMAT_U8;
sh_audio->samplesize = 2; }
sh_audio->context = talloc_zero(NULL, struct ad_pcm_context); if (!sh_audio->samplesize) // this would cause MPlayer to hang later
return 1; sh_audio->samplesize = 2;
sh_audio->context = talloc_zero(NULL, struct ad_pcm_context);
return 1;
} }
static int preinit(sh_audio_t *sh) static int preinit(sh_audio_t * sh)
{ {
sh->audio_out_minsize=2048; sh->audio_out_minsize = 2048;
return 1; return 1;
} }
static void uninit(sh_audio_t *sh) static void uninit(sh_audio_t * sh)
{ {
talloc_free(sh->context); talloc_free(sh->context);
} }
static int control(sh_audio_t *sh,int cmd,void* arg, ...) static int control(sh_audio_t * sh, int cmd, void *arg, ...)
{ {
int skip; int skip;
switch(cmd) switch (cmd) {
{ case ADCTRL_SKIP_FRAME:
case ADCTRL_SKIP_FRAME: skip = sh->i_bps / 16;
skip=sh->i_bps/16; skip = skip & (~3);
skip=skip&(~3); demux_read_data(sh->ds, NULL, skip);
demux_read_data(sh->ds,NULL,skip); return CONTROL_TRUE;
return CONTROL_TRUE;
} }
return CONTROL_UNKNOWN; return CONTROL_UNKNOWN;
} }
static int decode_audio(sh_audio_t *sh_audio,unsigned char *buf,int minlen,int maxlen) static int decode_audio(sh_audio_t * sh_audio, unsigned char *buf, int minlen,
int maxlen)
{ {
unsigned len = sh_audio->channels*sh_audio->samplesize; unsigned len = sh_audio->channels * sh_audio->samplesize;
minlen = (minlen + len - 1) / len * len; minlen = (minlen + len - 1) / len * len;
if (minlen > maxlen) if (minlen > maxlen)
// if someone needs hundreds of channels adjust audio_out_minsize // if someone needs hundreds of channels adjust audio_out_minsize
// based on channels in preinit() // based on channels in preinit()
return -1; return -1;
len = 0; len = 0;
struct ad_pcm_context *ctx = sh_audio->context; struct ad_pcm_context *ctx = sh_audio->context;
while (len < minlen) { while (len < minlen) {
if (ctx->packet_len == 0) { if (ctx->packet_len == 0) {
double pts; double pts;
int plen = ds_get_packet_pts(sh_audio->ds, &ctx->packet_ptr, &pts); int plen = ds_get_packet_pts(sh_audio->ds, &ctx->packet_ptr, &pts);
if (plen < 0) if (plen < 0)
break; break;
ctx->packet_len = plen; ctx->packet_len = plen;
if (pts != MP_NOPTS_VALUE) { if (pts != MP_NOPTS_VALUE) {
sh_audio->pts = pts; sh_audio->pts = pts;
sh_audio->pts_bytes = 0; sh_audio->pts_bytes = 0;
} }
} }
int from_stored = ctx->packet_len; int from_stored = ctx->packet_len;
if (from_stored > minlen - len) if (from_stored > minlen - len)
from_stored = minlen - len; from_stored = minlen - len;
memcpy(buf + len, ctx->packet_ptr, from_stored); memcpy(buf + len, ctx->packet_ptr, from_stored);
ctx->packet_len -= from_stored; ctx->packet_len -= from_stored;
ctx->packet_ptr += from_stored; ctx->packet_ptr += from_stored;
sh_audio->pts_bytes += from_stored; sh_audio->pts_bytes += from_stored;
len += from_stored; len += from_stored;
} }
if (len == 0) if (len == 0)
len = -1; // The loop above only exits at error/EOF len = -1; // The loop above only exits at error/EOF
if (len > 0 && sh_audio->channels >= 5) { if (len > 0 && sh_audio->channels >= 5) {
reorder_channel_nch(buf, AF_CHANNEL_LAYOUT_WAVEEX_DEFAULT, reorder_channel_nch(buf, AF_CHANNEL_LAYOUT_WAVEEX_DEFAULT,
AF_CHANNEL_LAYOUT_MPLAYER_DEFAULT, AF_CHANNEL_LAYOUT_MPLAYER_DEFAULT,
sh_audio->channels, sh_audio->channels, len / sh_audio->samplesize,
len / sh_audio->samplesize, sh_audio->samplesize); sh_audio->samplesize);
} }
return len; return len;
} }