mirror of
https://github.com/mpv-player/mpv
synced 2024-12-15 19:35:49 +00:00
36e1974be0
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@17013 b3059339-0415-0410-9bf9-f77b7e298cf2
321 lines
9.8 KiB
C
321 lines
9.8 KiB
C
#include <stdio.h>
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#include <stdlib.h>
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#include <unistd.h>
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#include <math.h>
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#include <assert.h>
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#include "config.h"
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#ifdef USE_LIBA52
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#include "mp_msg.h"
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#include "help_mp.h"
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#include "ad_internal.h"
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#include "cpudetect.h"
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#include "libaf/af_format.h"
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#include "liba52/a52.h"
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#include "liba52/mm_accel.h"
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static sample_t * a52_samples;
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static a52_state_t a52_state;
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static uint32_t a52_flags=0;
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/** Used by a52_resample_float, it defines the mapping between liba52
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* channels and output channels. The ith nibble from the right in the
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* hex representation of channel_map is the index of the source
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* channel corresponding to the ith output channel. Source channels are
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* indexed 1-6. Silent output channels are marked by 0xf. */
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static uint32_t channel_map;
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#define DRC_NO_ACTION 0
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#define DRC_NO_COMPRESSION 1
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#define DRC_CALLBACK 2
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/** The output is multiplied by this var. Used for volume control */
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static sample_t a52_level = 1;
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/** The value of the -a52drc switch. */
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float a52_drc_level = 1.0;
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static int a52_drc_action = DRC_NO_ACTION;
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#include "bswap.h"
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static ad_info_t info =
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{
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"AC3 decoding with liba52",
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"liba52",
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"Nick Kurshev",
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"Michel LESPINASSE",
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""
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};
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LIBAD_EXTERN(liba52)
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extern int audio_output_channels;
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int a52_fillbuff(sh_audio_t *sh_audio){
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int length=0;
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int flags=0;
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int sample_rate=0;
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int bit_rate=0;
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sh_audio->a_in_buffer_len=0;
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/* sync frame:*/
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while(1){
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while(sh_audio->a_in_buffer_len<8){
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int c=demux_getc(sh_audio->ds);
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if(c<0) return -1; /* EOF*/
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sh_audio->a_in_buffer[sh_audio->a_in_buffer_len++]=c;
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}
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if(sh_audio->format!=0x2000) swab(sh_audio->a_in_buffer,sh_audio->a_in_buffer,8);
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length = a52_syncinfo (sh_audio->a_in_buffer, &flags, &sample_rate, &bit_rate);
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if(length>=7 && length<=3840) break; /* we're done.*/
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/* bad file => resync*/
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if(sh_audio->format!=0x2000) swab(sh_audio->a_in_buffer,sh_audio->a_in_buffer,8);
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memmove(sh_audio->a_in_buffer,sh_audio->a_in_buffer+1,7);
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--sh_audio->a_in_buffer_len;
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}
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mp_msg(MSGT_DECAUDIO,MSGL_DBG2,"a52: len=%d flags=0x%X %d Hz %d bit/s\n",length,flags,sample_rate,bit_rate);
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sh_audio->samplerate=sample_rate;
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sh_audio->i_bps=bit_rate/8;
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sh_audio->samplesize=sh_audio->sample_format==AF_FORMAT_FLOAT_NE ? 4 : 2;
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demux_read_data(sh_audio->ds,sh_audio->a_in_buffer+8,length-8);
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if(sh_audio->format!=0x2000)
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swab(sh_audio->a_in_buffer+8,sh_audio->a_in_buffer+8,length-8);
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if(crc16_block(sh_audio->a_in_buffer+2,length-2)!=0)
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mp_msg(MSGT_DECAUDIO,MSGL_STATUS,"a52: CRC check failed! \n");
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return length;
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}
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/* returns: number of available channels*/
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static int a52_printinfo(sh_audio_t *sh_audio){
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int flags, sample_rate, bit_rate;
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char* mode="unknown";
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int channels=0;
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a52_syncinfo (sh_audio->a_in_buffer, &flags, &sample_rate, &bit_rate);
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switch(flags&A52_CHANNEL_MASK){
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case A52_CHANNEL: mode="channel"; channels=2; break;
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case A52_MONO: mode="mono"; channels=1; break;
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case A52_STEREO: mode="stereo"; channels=2; break;
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case A52_3F: mode="3f";channels=3;break;
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case A52_2F1R: mode="2f+1r";channels=3;break;
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case A52_3F1R: mode="3f+1r";channels=4;break;
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case A52_2F2R: mode="2f+2r";channels=4;break;
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case A52_3F2R: mode="3f+2r";channels=5;break;
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case A52_CHANNEL1: mode="channel1"; channels=2; break;
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case A52_CHANNEL2: mode="channel2"; channels=2; break;
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case A52_DOLBY: mode="dolby"; channels=2; break;
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}
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mp_msg(MSGT_DECAUDIO,MSGL_INFO,"AC3: %d.%d (%s%s) %d Hz %3.1f kbit/s\n",
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channels, (flags&A52_LFE)?1:0,
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mode, (flags&A52_LFE)?"+lfe":"",
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sample_rate, bit_rate*0.001f);
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return (flags&A52_LFE) ? (channels+1) : channels;
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}
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sample_t dynrng_call (sample_t c, void *data) {
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// fprintf(stderr, "(%lf, %lf): %lf\n", (double)c, (double)a52_drc_level, (double)pow((double)c, a52_drc_level));
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return pow((double)c, a52_drc_level);
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}
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static int preinit(sh_audio_t *sh)
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{
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/* Dolby AC3 audio: */
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/* however many channels, 2 bytes in a word, 256 samples in a block, 6 blocks in a frame */
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if (sh->samplesize < 2) sh->samplesize = 2;
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sh->audio_out_minsize=audio_output_channels*sh->samplesize*256*6;
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sh->audio_in_minsize=3840;
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a52_level = 1.0;
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return 1;
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}
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/**
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* \brief Function to convert the "planar" float format used by liba52
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* into the interleaved float format used by libaf/libao2.
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* \param in the input buffer containing the planar samples.
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* \param out the output buffer where the interleaved result is stored.
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*/
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static int a52_resample_float(float *in, int16_t *out)
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{
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unsigned long i;
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float *p = (float*) out;
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for (i = 0; i != 256; i++) {
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unsigned long map = channel_map;
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do {
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unsigned long ch = map & 15;
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if (ch == 15)
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*p = 0;
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else
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*p = in[i + ((ch-1)<<8)];
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p++;
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} while ((map >>= 4));
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}
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return (int16_t*) p - out;
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}
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static int init(sh_audio_t *sh_audio)
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{
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uint32_t a52_accel=0;
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sample_t level=a52_level, bias=384;
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int flags=0;
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/* Dolby AC3 audio:*/
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if(gCpuCaps.hasSSE) a52_accel|=MM_ACCEL_X86_SSE;
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if(gCpuCaps.hasMMX) a52_accel|=MM_ACCEL_X86_MMX;
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if(gCpuCaps.hasMMX2) a52_accel|=MM_ACCEL_X86_MMXEXT;
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if(gCpuCaps.has3DNow) a52_accel|=MM_ACCEL_X86_3DNOW;
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if(gCpuCaps.has3DNowExt) a52_accel|=MM_ACCEL_X86_3DNOWEXT;
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if(gCpuCaps.hasAltiVec) a52_accel|=MM_ACCEL_PPC_ALTIVEC;
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a52_samples=a52_init (a52_accel);
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if (a52_samples == NULL) {
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mp_msg(MSGT_DECAUDIO,MSGL_ERR,"A52 init failed\n");
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return 0;
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}
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if(a52_fillbuff(sh_audio)<0){
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mp_msg(MSGT_DECAUDIO,MSGL_ERR,"A52 sync failed\n");
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return 0;
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}
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/* Init a52 dynrng */
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if (a52_drc_level < 0.001) {
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/* level == 0 --> no compression, init library without callback */
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a52_drc_action = DRC_NO_COMPRESSION;
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} else if (a52_drc_level > 0.999) {
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/* level == 1 --> full compression, do nothing at all (library default = full compression) */
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a52_drc_action = DRC_NO_ACTION;
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} else {
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a52_drc_action = DRC_CALLBACK;
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}
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/* Library init for dynrng has to be done for each frame, see decode_audio() */
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/* 'a52 cannot upmix' hotfix:*/
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a52_printinfo(sh_audio);
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sh_audio->channels=audio_output_channels;
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while(sh_audio->channels>0){
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switch(sh_audio->channels){
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case 1: a52_flags=A52_MONO; break;
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/* case 2: a52_flags=A52_STEREO; break;*/
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case 2: a52_flags=A52_DOLBY; break;
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/* case 3: a52_flags=A52_3F; break;*/
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case 3: a52_flags=A52_2F1R; break;
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case 4: a52_flags=A52_2F2R; break; /* 2+2*/
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case 5: a52_flags=A52_3F2R; break;
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case 6: a52_flags=A52_3F2R|A52_LFE; break; /* 5.1*/
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}
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/* test:*/
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flags=a52_flags|A52_ADJUST_LEVEL;
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mp_msg(MSGT_DECAUDIO,MSGL_V,"A52 flags before a52_frame: 0x%X\n",flags);
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if (a52_frame (&a52_state, sh_audio->a_in_buffer, &flags, &level, bias)){
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mp_msg(MSGT_DECAUDIO,MSGL_ERR,"a52: error decoding frame -> nosound\n");
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return 0;
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}
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mp_msg(MSGT_DECAUDIO,MSGL_V,"A52 flags after a52_frame: 0x%X\n",flags);
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/* frame decoded, let's init resampler:*/
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channel_map = 0;
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if (sh_audio->sample_format == AF_FORMAT_FLOAT_NE) {
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if (!(flags & A52_LFE)) {
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switch ((flags<<3) | sh_audio->channels) {
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case (A52_MONO << 3) | 1: channel_map = 0x1; break;
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case (A52_CHANNEL << 3) | 2:
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case (A52_STEREO << 3) | 2:
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case (A52_DOLBY << 3) | 2: channel_map = 0x21; break;
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case (A52_2F1R << 3) | 3: channel_map = 0x321; break;
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case (A52_2F2R << 3) | 4: channel_map = 0x4321; break;
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case (A52_3F << 3) | 5: channel_map = 0x2ff31; break;
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case (A52_3F2R << 3) | 5: channel_map = 0x25431; break;
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}
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} else if (sh_audio->channels == 6) {
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switch (flags & ~A52_LFE) {
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case A52_MONO : channel_map = 0x12ffff; break;
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case A52_CHANNEL:
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case A52_STEREO :
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case A52_DOLBY : channel_map = 0x1fff32; break;
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case A52_3F : channel_map = 0x13ff42; break;
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case A52_2F1R : channel_map = 0x1f4432; break;
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case A52_2F2R : channel_map = 0x1f5432; break;
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case A52_3F2R : channel_map = 0x136542; break;
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}
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}
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if (channel_map) {
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a52_resample = a52_resample_float;
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break;
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}
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} else
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if(a52_resample_init(a52_accel,flags,sh_audio->channels)) break;
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--sh_audio->channels; /* try to decrease no. of channels*/
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}
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if(sh_audio->channels<=0){
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mp_msg(MSGT_DECAUDIO,MSGL_ERR,"a52: no resampler. try different channel setup!\n");
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return 0;
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}
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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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{
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}
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static int control(sh_audio_t *sh,int cmd,void* arg, ...)
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{
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switch(cmd)
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{
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case ADCTRL_RESYNC_STREAM:
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case ADCTRL_SKIP_FRAME:
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a52_fillbuff(sh);
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return CONTROL_TRUE;
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case ADCTRL_SET_VOLUME: {
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float vol = *(float*)arg;
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if (vol > 60.0) vol = 60.0;
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a52_level = vol <= -200.0 ? 0 : pow(10.0,vol/20.0);
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return CONTROL_TRUE;
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}
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case ADCTRL_QUERY_FORMAT:
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if (*(int*)arg == AF_FORMAT_S16_NE ||
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*(int*)arg == AF_FORMAT_FLOAT_NE)
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return CONTROL_TRUE;
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return CONTROL_FALSE;
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}
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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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{
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sample_t level=a52_level, bias=384;
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int flags=a52_flags|A52_ADJUST_LEVEL;
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int i,len=-1;
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if (sh_audio->sample_format == AF_FORMAT_FLOAT_NE)
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bias = 0;
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if(!sh_audio->a_in_buffer_len)
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if(a52_fillbuff(sh_audio)<0) return len; /* EOF */
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sh_audio->a_in_buffer_len=0;
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if (a52_frame (&a52_state, sh_audio->a_in_buffer, &flags, &level, bias)){
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mp_msg(MSGT_DECAUDIO,MSGL_WARN,"a52: error decoding frame\n");
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return len;
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}
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/* handle dynrng */
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if (a52_drc_action != DRC_NO_ACTION) {
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if (a52_drc_action == DRC_NO_COMPRESSION)
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a52_dynrng(&a52_state, NULL, NULL);
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else
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a52_dynrng(&a52_state, dynrng_call, NULL);
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}
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len=0;
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for (i = 0; i < 6; i++) {
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if (a52_block (&a52_state, a52_samples)){
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mp_msg(MSGT_DECAUDIO,MSGL_WARN,"a52: error at resampling\n");
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break;
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}
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len+=2*a52_resample(a52_samples,(int16_t *)&buf[len]);
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}
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assert(len <= maxlen);
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return len;
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}
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#endif
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