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https://github.com/mpv-player/mpv
synced 2025-03-11 04:44:32 +01:00
cosmetics: reindent ad_pcm.c
This commit is contained in:
Uoti Urpala
parent
9e02b2db40
commit
584b43e9f8
@ -26,13 +26,12 @@
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#include "libaf/af_format.h"
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#include "libaf/reorder_ch.h"
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static const ad_info_t info =
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{
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"Uncompressed PCM audio decoder",
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"pcm",
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"Nick Kurshev",
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"A'rpi",
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""
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static const ad_info_t info = {
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"Uncompressed PCM audio decoder",
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"pcm",
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"Nick Kurshev",
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"A'rpi",
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""
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};
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struct ad_pcm_context {
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@ -42,53 +41,55 @@ struct ad_pcm_context {
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LIBAD_EXTERN(pcm)
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static int init(sh_audio_t *sh_audio)
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static int init(sh_audio_t * sh_audio)
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{
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WAVEFORMATEX *h=sh_audio->wf;
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if (!h)
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return 0;
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sh_audio->i_bps=h->nAvgBytesPerSec;
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sh_audio->channels=h->nChannels;
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sh_audio->samplerate=h->nSamplesPerSec;
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sh_audio->samplesize=(h->wBitsPerSample+7)/8;
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sh_audio->sample_format=AF_FORMAT_S16_LE; // default
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switch(sh_audio->format){ /* hardware formats: */
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WAVEFORMATEX *h = sh_audio->wf;
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if (!h)
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return 0;
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sh_audio->i_bps = h->nAvgBytesPerSec;
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sh_audio->channels = h->nChannels;
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sh_audio->samplerate = h->nSamplesPerSec;
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sh_audio->samplesize = (h->wBitsPerSample + 7) / 8;
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sh_audio->sample_format = AF_FORMAT_S16_LE; // default
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switch (sh_audio->format) { /* hardware formats: */
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case 0x0:
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case 0x1: // Microsoft PCM
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case 0xfffe: // Extended
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switch (sh_audio->samplesize) {
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case 1: sh_audio->sample_format=AF_FORMAT_U8; break;
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case 2: sh_audio->sample_format=AF_FORMAT_S16_LE; break;
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case 3: sh_audio->sample_format=AF_FORMAT_S24_LE; break;
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case 4: sh_audio->sample_format=AF_FORMAT_S32_LE; break;
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}
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break;
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case 0x3: // IEEE float
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sh_audio->sample_format=AF_FORMAT_FLOAT_LE;
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break;
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case 0x6: sh_audio->sample_format=AF_FORMAT_A_LAW;break;
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case 0x7: sh_audio->sample_format=AF_FORMAT_MU_LAW;break;
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case 0x11: sh_audio->sample_format=AF_FORMAT_IMA_ADPCM;break;
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case 0x50: sh_audio->sample_format=AF_FORMAT_MPEG2;break;
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case 0x1: // Microsoft PCM
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case 0xfffe: // Extended
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switch (sh_audio->samplesize) {
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case 1: sh_audio->sample_format = AF_FORMAT_U8; break;
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case 2: sh_audio->sample_format = AF_FORMAT_S16_LE; break;
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case 3: sh_audio->sample_format = AF_FORMAT_S24_LE; break;
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case 4: sh_audio->sample_format = AF_FORMAT_S32_LE; break;
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}
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break;
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case 0x3: // IEEE float
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sh_audio->sample_format = AF_FORMAT_FLOAT_LE;
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break;
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case 0x6: sh_audio->sample_format = AF_FORMAT_A_LAW; break;
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case 0x7: sh_audio->sample_format = AF_FORMAT_MU_LAW; break;
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case 0x11: sh_audio->sample_format = AF_FORMAT_IMA_ADPCM; break;
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case 0x50: sh_audio->sample_format = AF_FORMAT_MPEG2; break;
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/* case 0x2000: sh_audio->sample_format=AFMT_AC3; */
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case 0x20776172: // 'raw '
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sh_audio->sample_format=AF_FORMAT_S16_BE;
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if(sh_audio->samplesize==1) sh_audio->sample_format=AF_FORMAT_U8;
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break;
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sh_audio->sample_format = AF_FORMAT_S16_BE;
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if (sh_audio->samplesize == 1)
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sh_audio->sample_format = AF_FORMAT_U8;
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break;
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case 0x736F7774: // 'twos'
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sh_audio->sample_format=AF_FORMAT_S16_BE;
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// intended fall-through
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sh_audio->sample_format = AF_FORMAT_S16_BE;
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// intended fall-through
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case 0x74776F73: // 'sowt'
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if(sh_audio->samplesize==1) sh_audio->sample_format=AF_FORMAT_S8;
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break;
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if (sh_audio->samplesize == 1)
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sh_audio->sample_format = AF_FORMAT_S8;
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break;
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case 0x32336c66: // 'fl32', bigendian float32
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sh_audio->sample_format=AF_FORMAT_FLOAT_BE;
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sh_audio->samplesize=4;
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break;
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sh_audio->sample_format = AF_FORMAT_FLOAT_BE;
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sh_audio->samplesize = 4;
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break;
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case 0x666c3332: // '23lf', little endian float32, MPlayer internal fourCC
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sh_audio->sample_format=AF_FORMAT_FLOAT_LE;
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sh_audio->samplesize=4;
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break;
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sh_audio->sample_format = AF_FORMAT_FLOAT_LE;
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sh_audio->samplesize = 4;
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break;
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/* case 0x34366c66: // 'fl64', bigendian float64
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sh_audio->sample_format=AF_FORMAT_FLOAT_BE;
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sh_audio->samplesize=8;
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@ -98,93 +99,95 @@ static int init(sh_audio_t *sh_audio)
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sh_audio->samplesize=8;
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break;*/
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case 0x34326e69: // 'in24', bigendian int24
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sh_audio->sample_format=AF_FORMAT_S24_BE;
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sh_audio->samplesize=3;
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break;
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sh_audio->sample_format = AF_FORMAT_S24_BE;
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sh_audio->samplesize = 3;
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break;
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case 0x696e3234: // '42ni', little endian int24, MPlayer internal fourCC
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sh_audio->sample_format=AF_FORMAT_S24_LE;
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sh_audio->samplesize=3;
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break;
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sh_audio->sample_format = AF_FORMAT_S24_LE;
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sh_audio->samplesize = 3;
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break;
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case 0x32336e69: // 'in32', bigendian int32
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sh_audio->sample_format=AF_FORMAT_S32_BE;
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sh_audio->samplesize=4;
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break;
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sh_audio->sample_format = AF_FORMAT_S32_BE;
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sh_audio->samplesize = 4;
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break;
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case 0x696e3332: // '23ni', little endian int32, MPlayer internal fourCC
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sh_audio->sample_format=AF_FORMAT_S32_LE;
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sh_audio->samplesize=4;
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break;
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default: if(sh_audio->samplesize!=2) sh_audio->sample_format=AF_FORMAT_U8;
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}
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if (!sh_audio->samplesize) // this would cause MPlayer to hang later
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sh_audio->samplesize = 2;
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sh_audio->context = talloc_zero(NULL, struct ad_pcm_context);
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return 1;
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sh_audio->sample_format = AF_FORMAT_S32_LE;
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sh_audio->samplesize = 4;
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break;
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default:
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if (sh_audio->samplesize != 2)
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sh_audio->sample_format = AF_FORMAT_U8;
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}
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if (!sh_audio->samplesize) // this would cause MPlayer to hang later
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sh_audio->samplesize = 2;
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sh_audio->context = talloc_zero(NULL, struct ad_pcm_context);
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return 1;
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}
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static int preinit(sh_audio_t *sh)
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static int preinit(sh_audio_t * sh)
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{
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sh->audio_out_minsize=2048;
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return 1;
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sh->audio_out_minsize = 2048;
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return 1;
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}
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static void uninit(sh_audio_t *sh)
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static void uninit(sh_audio_t * sh)
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{
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talloc_free(sh->context);
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}
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static int control(sh_audio_t *sh,int cmd,void* arg, ...)
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static int control(sh_audio_t * sh, int cmd, void *arg, ...)
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{
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int skip;
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switch(cmd)
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{
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case ADCTRL_SKIP_FRAME:
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skip=sh->i_bps/16;
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skip=skip&(~3);
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demux_read_data(sh->ds,NULL,skip);
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return CONTROL_TRUE;
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int skip;
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switch (cmd) {
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case ADCTRL_SKIP_FRAME:
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skip = sh->i_bps / 16;
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skip = skip & (~3);
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demux_read_data(sh->ds, NULL, skip);
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return CONTROL_TRUE;
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}
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return CONTROL_UNKNOWN;
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return CONTROL_UNKNOWN;
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}
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static int decode_audio(sh_audio_t *sh_audio,unsigned char *buf,int minlen,int maxlen)
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static int decode_audio(sh_audio_t * sh_audio, unsigned char *buf, int minlen,
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int maxlen)
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{
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unsigned len = sh_audio->channels*sh_audio->samplesize;
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minlen = (minlen + len - 1) / len * len;
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if (minlen > maxlen)
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// if someone needs hundreds of channels adjust audio_out_minsize
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// based on channels in preinit()
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return -1;
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unsigned len = sh_audio->channels * sh_audio->samplesize;
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minlen = (minlen + len - 1) / len * len;
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if (minlen > maxlen)
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// if someone needs hundreds of channels adjust audio_out_minsize
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// based on channels in preinit()
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return -1;
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len = 0;
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struct ad_pcm_context *ctx = sh_audio->context;
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while (len < minlen) {
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if (ctx->packet_len == 0) {
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double pts;
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int plen = ds_get_packet_pts(sh_audio->ds, &ctx->packet_ptr, &pts);
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if (plen < 0)
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break;
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ctx->packet_len = plen;
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if (pts != MP_NOPTS_VALUE) {
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sh_audio->pts = pts;
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sh_audio->pts_bytes = 0;
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}
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}
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int from_stored = ctx->packet_len;
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if (from_stored > minlen - len)
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from_stored = minlen - len;
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memcpy(buf + len, ctx->packet_ptr, from_stored);
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ctx->packet_len -= from_stored;
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ctx->packet_ptr += from_stored;
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sh_audio->pts_bytes += from_stored;
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len += from_stored;
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}
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if (len == 0)
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len = -1; // The loop above only exits at error/EOF
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if (len > 0 && sh_audio->channels >= 5) {
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reorder_channel_nch(buf, AF_CHANNEL_LAYOUT_WAVEEX_DEFAULT,
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AF_CHANNEL_LAYOUT_MPLAYER_DEFAULT,
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sh_audio->channels,
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len / sh_audio->samplesize, sh_audio->samplesize);
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}
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return len;
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len = 0;
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struct ad_pcm_context *ctx = sh_audio->context;
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while (len < minlen) {
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if (ctx->packet_len == 0) {
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double pts;
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int plen = ds_get_packet_pts(sh_audio->ds, &ctx->packet_ptr, &pts);
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if (plen < 0)
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break;
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ctx->packet_len = plen;
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if (pts != MP_NOPTS_VALUE) {
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sh_audio->pts = pts;
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sh_audio->pts_bytes = 0;
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}
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}
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int from_stored = ctx->packet_len;
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if (from_stored > minlen - len)
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from_stored = minlen - len;
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memcpy(buf + len, ctx->packet_ptr, from_stored);
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ctx->packet_len -= from_stored;
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ctx->packet_ptr += from_stored;
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sh_audio->pts_bytes += from_stored;
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len += from_stored;
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}
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if (len == 0)
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len = -1; // The loop above only exits at error/EOF
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if (len > 0 && sh_audio->channels >= 5) {
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reorder_channel_nch(buf, AF_CHANNEL_LAYOUT_WAVEEX_DEFAULT,
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AF_CHANNEL_LAYOUT_MPLAYER_DEFAULT,
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sh_audio->channels, len / sh_audio->samplesize,
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sh_audio->samplesize);
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}
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return len;
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}
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