From 6a26b4a66504f701baf35e58467e55aea28c0ad5 Mon Sep 17 00:00:00 2001 From: wm4 Date: Sat, 18 Aug 2012 11:17:35 +0200 Subject: libmpcodecs: remove redundant audio and video decoders Probably all of these are supported by libavcodec. Missing things can be added back. Also remove qtpalette.h. It was used by demux_mov.c, and should have been deleted with commit 1fde09db6f4ce. --- libmpcodecs/ad_liba52.c | 339 ------------------------------------------------ 1 file changed, 339 deletions(-) delete mode 100644 libmpcodecs/ad_liba52.c (limited to 'libmpcodecs/ad_liba52.c') diff --git a/libmpcodecs/ad_liba52.c b/libmpcodecs/ad_liba52.c deleted file mode 100644 index 505532af6b..0000000000 --- a/libmpcodecs/ad_liba52.c +++ /dev/null @@ -1,339 +0,0 @@ -/* - * This file is part of MPlayer. - * - * MPlayer is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License as published by - * the Free Software Foundation; either version 2 of the License, or - * (at your option) any later version. - * - * MPlayer is distributed in the hope that it will be useful, - * but WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the - * GNU General Public License for more details. - * - * You should have received a copy of the GNU General Public License along - * with MPlayer; if not, write to the Free Software Foundation, Inc., - * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. - */ - -#define _GNU_SOURCE -#define _XOPEN_SOURCE 600 -#include -#include -#include -#include -#include - -#include "config.h" -#include "options.h" -#include "mp_msg.h" -#include "mpbswap.h" - -#include "ad_internal.h" - -#include "cpudetect.h" - -#include "libaf/af_format.h" - -#include -#include -int (* a52_resample) (float * _f, int16_t * s16); - -static a52_state_t *a52_state; -static uint32_t a52_flags=0; -/** Used by a52_resample_float, it defines the mapping between liba52 - * channels and output channels. The ith nibble from the right in the - * hex representation of channel_map is the index of the source - * channel corresponding to the ith output channel. Source channels are - * indexed 1-6. Silent output channels are marked by 0xf. */ -static uint32_t channel_map; - -#define DRC_NO_ACTION 0 -#define DRC_NO_COMPRESSION 1 -#define DRC_CALLBACK 2 - -/** The output is multiplied by this var. Used for volume control */ -static sample_t a52_level = 1; -static int a52_drc_action = DRC_NO_ACTION; - -static const ad_info_t info = -{ - "AC3 decoding with liba52", - "liba52", - "Nick Kurshev", - "Michel LESPINASSE", - "" -}; - -LIBAD_EXTERN(liba52) - -static int a52_fillbuff(sh_audio_t *sh_audio) -{ -int length=0; -int flags=0; -int sample_rate=0; -int bit_rate=0; - - sh_audio->a_in_buffer_len=0; - /* sync frame:*/ -while(1){ - while(sh_audio->a_in_buffer_len<8){ - int c=demux_getc(sh_audio->ds); - if(c<0) return -1; /* EOF*/ - sh_audio->a_in_buffer[sh_audio->a_in_buffer_len++]=c; - } - if(sh_audio->format==MKTAG('d','n','e','t')) swab(sh_audio->a_in_buffer,sh_audio->a_in_buffer,8); - length = a52_syncinfo (sh_audio->a_in_buffer, &flags, &sample_rate, &bit_rate); - if(length>=7 && length<=3840) break; /* we're done.*/ - /* bad file => resync*/ - if(sh_audio->format==MKTAG('d','n','e','t')) swab(sh_audio->a_in_buffer,sh_audio->a_in_buffer,8); - memmove(sh_audio->a_in_buffer,sh_audio->a_in_buffer+1,7); - --sh_audio->a_in_buffer_len; -} - mp_msg(MSGT_DECAUDIO,MSGL_DBG2,"a52: len=%d flags=0x%X %d Hz %d bit/s\n",length,flags,sample_rate,bit_rate); - sh_audio->samplerate=sample_rate; - sh_audio->i_bps=bit_rate/8; - sh_audio->samplesize=sh_audio->sample_format==AF_FORMAT_FLOAT_NE ? 4 : 2; - demux_read_data(sh_audio->ds,sh_audio->a_in_buffer+8,length-8); - if(sh_audio->format==MKTAG('d','n','e','t')) - swab(sh_audio->a_in_buffer+8,sh_audio->a_in_buffer+8,length-8); - -#ifdef CONFIG_LIBA52_INTERNAL - if(crc16_block(sh_audio->a_in_buffer+2,length-2)!=0) - mp_msg(MSGT_DECAUDIO,MSGL_STATUS,"a52: CRC check failed! \n"); -#endif - - return length; -} - -/* returns: number of available channels*/ -static int a52_printinfo(sh_audio_t *sh_audio){ -int flags, sample_rate, bit_rate; -char* mode="unknown"; -int channels=0; - a52_syncinfo (sh_audio->a_in_buffer, &flags, &sample_rate, &bit_rate); - switch(flags&A52_CHANNEL_MASK){ - case A52_CHANNEL: mode="channel"; channels=2; break; - case A52_MONO: mode="mono"; channels=1; break; - case A52_STEREO: mode="stereo"; channels=2; break; - case A52_3F: mode="3f";channels=3;break; - case A52_2F1R: mode="2f+1r";channels=3;break; - case A52_3F1R: mode="3f+1r";channels=4;break; - case A52_2F2R: mode="2f+2r";channels=4;break; - case A52_3F2R: mode="3f+2r";channels=5;break; - case A52_CHANNEL1: mode="channel1"; channels=2; break; - case A52_CHANNEL2: mode="channel2"; channels=2; break; - case A52_DOLBY: mode="dolby"; channels=2; break; - } - mp_msg(MSGT_DECAUDIO,MSGL_V,"AC3: %d.%d (%s%s) %d Hz %3.1f kbit/s\n", - channels, (flags&A52_LFE)?1:0, - mode, (flags&A52_LFE)?"+lfe":"", - sample_rate, bit_rate*0.001f); - return (flags&A52_LFE) ? (channels+1) : channels; -} - -static sample_t dynrng_call (sample_t c, void *data) -{ - struct MPOpts *opts = data; - //fprintf(stderr, "(%f, %f): %f\n", (double)c, (double)drc_level, (double)pow((double)c, drc_level)); - //fprintf(stderr, "(%lf, %lf): %lf\n", (double)c, opts->drc_level, pow(c, opts->drc_level)); - return pow(c, opts->drc_level); -} - - -static int preinit(sh_audio_t *sh) -{ - struct MPOpts *opts = sh->opts; - /* Dolby AC3 audio: */ - /* however many channels, 2 bytes in a word, 256 samples in a block, 6 blocks in a frame */ - if (sh->samplesize < 4) sh->samplesize = 4; - sh->audio_out_minsize=opts->audio_output_channels*sh->samplesize*256*6; - sh->audio_in_minsize=3840; - a52_level = 1.0; - return 1; -} - -/** - * \brief Function to convert the "planar" float format used by liba52 - * into the interleaved float format used by libaf/libao2. - * \param in the input buffer containing the planar samples. - * \param out the output buffer where the interleaved result is stored. - */ -static int a52_resample_float(float *in, int16_t *out) -{ - unsigned long i; - float *p = (float*) out; - for (i = 0; i != 256; i++) { - unsigned long map = channel_map; - do { - unsigned long ch = map & 15; - if (ch == 15) - *p = 0; - else - *p = in[i + ((ch-1)<<8)]; - p++; - } while ((map >>= 4)); - } - return (int16_t*) p - out; -} - -static int init(sh_audio_t *sh_audio) -{ - struct MPOpts *opts = sh_audio->opts; - uint32_t a52_accel=0; - sample_t level=a52_level, bias=384; - int flags=0; - /* Dolby AC3 audio:*/ -#ifdef MM_ACCEL_X86_SSE - if(gCpuCaps.hasSSE) a52_accel|=MM_ACCEL_X86_SSE; -#endif - if(gCpuCaps.hasMMX) a52_accel|=MM_ACCEL_X86_MMX; - if(gCpuCaps.hasMMX2) a52_accel|=MM_ACCEL_X86_MMXEXT; - a52_state=a52_init (a52_accel); - if (a52_state == NULL) { - mp_msg(MSGT_DECAUDIO,MSGL_ERR,"A52 init failed\n"); - return 0; - } - sh_audio->sample_format = AF_FORMAT_FLOAT_NE; - if(a52_fillbuff(sh_audio)<0){ - mp_msg(MSGT_DECAUDIO,MSGL_ERR,"A52 sync failed\n"); - return 0; - } - - /* Init a52 dynrng */ - if (opts->drc_level < 0.001) { - /* level == 0 --> no compression, init library without callback */ - a52_drc_action = DRC_NO_COMPRESSION; - } else if (opts->drc_level > 0.999 && opts->drc_level < 1.001) { - /* level == 1 --> full compression, do nothing at all (library default = full compression) */ - a52_drc_action = DRC_NO_ACTION; - } else { - a52_drc_action = DRC_CALLBACK; - } - /* Library init for dynrng has to be done for each frame, see decode_audio() */ - - - /* 'a52 cannot upmix' hotfix:*/ - a52_printinfo(sh_audio); - sh_audio->channels=opts->audio_output_channels; -while(sh_audio->channels>0){ - switch(sh_audio->channels){ - case 1: a52_flags=A52_MONO; break; -/* case 2: a52_flags=A52_STEREO; break;*/ - case 2: a52_flags=A52_DOLBY; break; -/* case 3: a52_flags=A52_3F; break;*/ - case 3: a52_flags=A52_2F1R; break; - case 4: a52_flags=A52_2F2R; break; /* 2+2*/ - case 5: a52_flags=A52_3F2R; break; - case 6: a52_flags=A52_3F2R|A52_LFE; break; /* 5.1*/ - } - /* test:*/ - flags=a52_flags|A52_ADJUST_LEVEL; - mp_msg(MSGT_DECAUDIO,MSGL_V,"A52 flags before a52_frame: 0x%X\n",flags); - if (a52_frame (a52_state, sh_audio->a_in_buffer, &flags, &level, bias)){ - mp_msg(MSGT_DECAUDIO,MSGL_ERR,"a52: error decoding frame -> nosound\n"); - return 0; - } - mp_msg(MSGT_DECAUDIO,MSGL_V,"A52 flags after a52_frame: 0x%X\n",flags); - /* frame decoded, let's init resampler:*/ - channel_map = 0; - if (sh_audio->sample_format == AF_FORMAT_FLOAT_NE) { - if (!(flags & A52_LFE)) { - switch ((flags<<3) | sh_audio->channels) { - case (A52_MONO << 3) | 1: channel_map = 0x1; break; - case (A52_CHANNEL << 3) | 2: - case (A52_STEREO << 3) | 2: - case (A52_DOLBY << 3) | 2: channel_map = 0x21; break; - case (A52_2F1R << 3) | 3: channel_map = 0x321; break; - case (A52_2F2R << 3) | 4: channel_map = 0x4321; break; - case (A52_3F << 3) | 5: channel_map = 0x2ff31; break; - case (A52_3F2R << 3) | 5: channel_map = 0x25431; break; - } - } else if (sh_audio->channels == 6) { - switch (flags & ~A52_LFE) { - case A52_MONO : channel_map = 0x12ffff; break; - case A52_CHANNEL: - case A52_STEREO : - case A52_DOLBY : channel_map = 0x1fff32; break; - case A52_3F : channel_map = 0x13ff42; break; - case A52_2F1R : channel_map = 0x1f4432; break; - case A52_2F2R : channel_map = 0x1f5432; break; - case A52_3F2R : channel_map = 0x136542; break; - } - } - if (channel_map) { - a52_resample = a52_resample_float; - break; - } - } else - break; -} - if(sh_audio->channels<=0){ - mp_msg(MSGT_DECAUDIO,MSGL_ERR,"a52: no resampler. try different channel setup!\n"); - return 0; - } - return 1; -} - -static void uninit(sh_audio_t *sh) -{ - a52_free(a52_state); -} - -static int control(sh_audio_t *sh,int cmd,void* arg, ...) -{ - switch(cmd) - { - case ADCTRL_RESYNC_STREAM: - case ADCTRL_SKIP_FRAME: - a52_fillbuff(sh); - return CONTROL_TRUE; - case ADCTRL_SET_VOLUME: { - float vol = *(float*)arg; - if (vol > 60.0) vol = 60.0; - a52_level = vol <= -200.0 ? 0 : pow(10.0,vol/20.0); - return CONTROL_TRUE; - } - case ADCTRL_QUERY_FORMAT: - if (*(int*)arg == AF_FORMAT_S16_NE || - *(int*)arg == AF_FORMAT_FLOAT_NE) - return CONTROL_TRUE; - return CONTROL_FALSE; - } - return CONTROL_UNKNOWN; -} - -static int decode_audio(sh_audio_t *sh_audio,unsigned char *buf,int minlen,int maxlen) -{ - sample_t level=a52_level, bias=384; - int flags=a52_flags|A52_ADJUST_LEVEL; - int i,len=-1; - if (sh_audio->sample_format == AF_FORMAT_FLOAT_NE) - bias = 0; - if(!sh_audio->a_in_buffer_len) - if(a52_fillbuff(sh_audio)<0) return len; /* EOF */ - sh_audio->a_in_buffer_len=0; - if (a52_frame (a52_state, sh_audio->a_in_buffer, &flags, &level, bias)){ - mp_msg(MSGT_DECAUDIO,MSGL_WARN,"a52: error decoding frame\n"); - return len; - } - - /* handle dynrng */ - if (a52_drc_action != DRC_NO_ACTION) { - if (a52_drc_action == DRC_NO_COMPRESSION) - a52_dynrng(a52_state, NULL, NULL); - else - a52_dynrng(a52_state, dynrng_call, sh_audio->opts); - } - - len=0; - for (i = 0; i < 6; i++) { - if (a52_block (a52_state)){ - mp_msg(MSGT_DECAUDIO,MSGL_WARN,"a52: error at resampling\n"); - break; - } - len+=2*a52_resample(a52_samples(a52_state),(int16_t *)&buf[len]); - } - assert(len <= maxlen); - return len; -} -- cgit v1.2.3