diff options
Diffstat (limited to 'libmpcodecs')
| -rw-r--r-- | libmpcodecs/ad.c | 16 | ||||
| -rw-r--r-- | libmpcodecs/ad_dk3adpcm.c | 262 | ||||
| -rw-r--r-- | libmpcodecs/ad_libvorbis.c | 350 | ||||
| -rw-r--r-- | libmpcodecs/ad_mpc.c | 231 | ||||
| -rw-r--r-- | libmpcodecs/ad_speex.c | 179 | ||||
| -rw-r--r-- | libmpcodecs/ad_twin.c | 523 | ||||
| -rw-r--r-- | libmpcodecs/dec_video.c | 1 | ||||
| -rw-r--r-- | libmpcodecs/vd.c | 12 | ||||
| -rw-r--r-- | libmpcodecs/vd_mpegpes.c | 84 | ||||
| -rw-r--r-- | libmpcodecs/vd_sgi.c | 343 | ||||
| -rw-r--r-- | libmpcodecs/vd_theora.c | 207 | ||||
| -rw-r--r-- | libmpcodecs/vd_xvid4.c | 393 |
12 files changed, 0 insertions, 2601 deletions
diff --git a/libmpcodecs/ad.c b/libmpcodecs/ad.c index 0ab5d44971..dce8f9f710 100644 --- a/libmpcodecs/ad.c +++ b/libmpcodecs/ad.c @@ -41,20 +41,14 @@ extern const ad_functions_t mpcodecs_ad_dvdpcm; extern const ad_functions_t mpcodecs_ad_alaw; extern const ad_functions_t mpcodecs_ad_imaadpcm; extern const ad_functions_t mpcodecs_ad_msadpcm; -extern const ad_functions_t mpcodecs_ad_dk3adpcm; -extern const ad_functions_t mpcodecs_ad_dk4adpcm; extern const ad_functions_t mpcodecs_ad_dshow; extern const ad_functions_t mpcodecs_ad_dmo; extern const ad_functions_t mpcodecs_ad_acm; extern const ad_functions_t mpcodecs_ad_faad; -extern const ad_functions_t mpcodecs_ad_libvorbis; -extern const ad_functions_t mpcodecs_ad_speex; extern const ad_functions_t mpcodecs_ad_libmad; extern const ad_functions_t mpcodecs_ad_realaud; extern const ad_functions_t mpcodecs_ad_libdv; extern const ad_functions_t mpcodecs_ad_qtaudio; -extern const ad_functions_t mpcodecs_ad_twin; -extern const ad_functions_t mpcodecs_ad_libmusepack; extern const ad_functions_t mpcodecs_ad_libdca; const ad_functions_t * const mpcodecs_ad_drivers[] = @@ -73,7 +67,6 @@ const ad_functions_t * const mpcodecs_ad_drivers[] = &mpcodecs_ad_alaw, &mpcodecs_ad_imaadpcm, &mpcodecs_ad_msadpcm, - &mpcodecs_ad_dk3adpcm, #ifdef CONFIG_WIN32DLL &mpcodecs_ad_dshow, &mpcodecs_ad_dmo, @@ -86,12 +79,6 @@ const ad_functions_t * const mpcodecs_ad_drivers[] = #ifdef CONFIG_FAAD &mpcodecs_ad_faad, #endif -#ifdef CONFIG_OGGVORBIS - &mpcodecs_ad_libvorbis, -#endif -#ifdef CONFIG_SPEEX - &mpcodecs_ad_speex, -#endif #ifdef CONFIG_LIBMAD &mpcodecs_ad_libmad, #endif @@ -101,9 +88,6 @@ const ad_functions_t * const mpcodecs_ad_drivers[] = #ifdef CONFIG_LIBDV095 &mpcodecs_ad_libdv, #endif -#ifdef CONFIG_MUSEPACK - &mpcodecs_ad_libmusepack, -#endif #ifdef CONFIG_LIBDCA &mpcodecs_ad_libdca, #endif diff --git a/libmpcodecs/ad_dk3adpcm.c b/libmpcodecs/ad_dk3adpcm.c deleted file mode 100644 index 15027f88b1..0000000000 --- a/libmpcodecs/ad_dk3adpcm.c +++ /dev/null @@ -1,262 +0,0 @@ -/* - * DK3 ADPCM decoder - * - * "This format number was used by Duck Corp. but not officially - * registered with Microsoft" - * - * This file is responsible for decoding audio data encoded with - * Duck Corp's DK3 ADPCM algorithm. Details about the data format - * can be found here: - * http://www.pcisys.net/~melanson/codecs/ - * - * Copyright (c) 2002 Mike Melanson - * - * 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. - */ - -#include <stdio.h> -#include <stdlib.h> -#include <unistd.h> - -#include <libavutil/intreadwrite.h> - -#include "config.h" -#include "ad_internal.h" - -static const ad_info_t info = -{ - "Duck Corp. DK3 ADPCM decoder", - "dk3adpcm", - "Nick Kurshev", - "Mike Melanson", - "" -}; - -LIBAD_EXTERN(dk3adpcm) - -#define DK3_ADPCM_PREAMBLE_SIZE 16 - -// useful macros -// clamp a number between 0 and 88 -#define CLAMP_0_TO_88(x) if (x < 0) x = 0; else if (x > 88) x = 88; -// clamp a number within a signed 16-bit range -#define CLAMP_S16(x) if (x < -32768) x = -32768; \ - else if (x > 32767) x = 32767; -// clamp a number above 16 -#define CLAMP_ABOVE_16(x) if (x < 16) x = 16; -// sign extend a 16-bit value -#define SE_16BIT(x) if (x & 0x8000) x -= 0x10000; -// sign extend a 4-bit value -#define SE_4BIT(x) if (x & 0x8) x -= 0x10; - -// pertinent tables -static int adpcm_step[89] = -{ - 7, 8, 9, 10, 11, 12, 13, 14, 16, 17, - 19, 21, 23, 25, 28, 31, 34, 37, 41, 45, - 50, 55, 60, 66, 73, 80, 88, 97, 107, 118, - 130, 143, 157, 173, 190, 209, 230, 253, 279, 307, - 337, 371, 408, 449, 494, 544, 598, 658, 724, 796, - 876, 963, 1060, 1166, 1282, 1411, 1552, 1707, 1878, 2066, - 2272, 2499, 2749, 3024, 3327, 3660, 4026, 4428, 4871, 5358, - 5894, 6484, 7132, 7845, 8630, 9493, 10442, 11487, 12635, 13899, - 15289, 16818, 18500, 20350, 22385, 24623, 27086, 29794, 32767 -}; - -static int adpcm_index[16] = -{ - -1, -1, -1, -1, 2, 4, 6, 8, - -1, -1, -1, -1, 2, 4, 6, 8 -}; - -static int preinit(sh_audio_t *sh_audio) -{ - sh_audio->audio_out_minsize = sh_audio->wf->nBlockAlign * 6; - sh_audio->ds->ss_div = - (sh_audio->wf->nBlockAlign - DK3_ADPCM_PREAMBLE_SIZE) * 8 / 3; - sh_audio->audio_in_minsize= - sh_audio->ds->ss_mul = sh_audio->wf->nBlockAlign; - return 1; -} - -static int init(sh_audio_t *sh_audio) -{ - sh_audio->channels = sh_audio->wf->nChannels; - sh_audio->samplerate = sh_audio->wf->nSamplesPerSec; - sh_audio->i_bps = - (sh_audio->ds->ss_mul * sh_audio->samplerate) / sh_audio->ds->ss_div; - sh_audio->samplesize=2; - return 1; -} - -static void uninit(sh_audio_t *sh_audio) -{ -} - -static int control(sh_audio_t *sh_audio,int cmd,void* arg, ...) -{ - if(cmd==ADCTRL_SKIP_FRAME){ - demux_read_data(sh_audio->ds, sh_audio->a_in_buffer,sh_audio->ds->ss_mul); - return CONTROL_TRUE; - } - return CONTROL_UNKNOWN; -} - -#define DK3_GET_NEXT_NIBBLE() \ - if (decode_top_nibble_next) \ - { \ - nibble = (last_byte >> 4) & 0x0F; \ - decode_top_nibble_next = 0; \ - } \ - else \ - { \ - last_byte = input[in_ptr++]; \ - nibble = last_byte & 0x0F; \ - decode_top_nibble_next = 1; \ - } - -// note: This decoder assumes the format 0x62 data always comes in -// stereo flavor -static int dk3_adpcm_decode_block(unsigned short *output, unsigned char *input, - int block_size) -{ - int sum_pred; - int diff_pred; - int sum_index; - int diff_index; - int diff_channel; - int in_ptr = 0x10; - int out_ptr = 0; - - unsigned char last_byte = 0; - unsigned char nibble; - int decode_top_nibble_next = 0; - - // ADPCM work variables - int sign; - int delta; - int step; - int diff; - - sum_pred = AV_RL16(&input[10]); - diff_pred = AV_RL16(&input[12]); - SE_16BIT(sum_pred); - SE_16BIT(diff_pred); - diff_channel = diff_pred; - sum_index = input[14]; - diff_index = input[15]; - - while (in_ptr < block_size - !decode_top_nibble_next) -// while (in_ptr < 2048) - { - // process the first predictor of the sum channel - DK3_GET_NEXT_NIBBLE(); - - step = adpcm_step[sum_index]; - - sign = nibble & 8; - delta = nibble & 7; - - diff = step >> 3; - if (delta & 4) diff += step; - if (delta & 2) diff += step >> 1; - if (delta & 1) diff += step >> 2; - - if (sign) - sum_pred -= diff; - else - sum_pred += diff; - - CLAMP_S16(sum_pred); - - sum_index += adpcm_index[nibble]; - CLAMP_0_TO_88(sum_index); - - // process the diff channel predictor - DK3_GET_NEXT_NIBBLE(); - - step = adpcm_step[diff_index]; - - sign = nibble & 8; - delta = nibble & 7; - - diff = step >> 3; - if (delta & 4) diff += step; - if (delta & 2) diff += step >> 1; - if (delta & 1) diff += step >> 2; - - if (sign) - diff_pred -= diff; - else - diff_pred += diff; - - CLAMP_S16(diff_pred); - - diff_index += adpcm_index[nibble]; - CLAMP_0_TO_88(diff_index); - - // output the first pair of stereo PCM samples - diff_channel = (diff_channel + diff_pred) / 2; - output[out_ptr++] = sum_pred + diff_channel; - output[out_ptr++] = sum_pred - diff_channel; - - // process the second predictor of the sum channel - DK3_GET_NEXT_NIBBLE(); - - step = adpcm_step[sum_index]; - - sign = nibble & 8; - delta = nibble & 7; - - diff = step >> 3; - if (delta & 4) diff += step; - if (delta & 2) diff += step >> 1; - if (delta & 1) diff += step >> 2; - - if (sign) - sum_pred -= diff; - else - sum_pred += diff; - - CLAMP_S16(sum_pred); - - sum_index += adpcm_index[nibble]; - CLAMP_0_TO_88(sum_index); - - // output the second pair of stereo PCM samples - output[out_ptr++] = sum_pred + diff_channel; - output[out_ptr++] = sum_pred - diff_channel; - } - - return out_ptr; -} - -static int decode_audio(sh_audio_t *sh_audio,unsigned char *buf,int minlen,int maxlen) -{ - if (demux_read_data(sh_audio->ds, sh_audio->a_in_buffer, - sh_audio->ds->ss_mul) != - sh_audio->ds->ss_mul) - return -1; /* EOF */ - - if (maxlen < 2 * 4 * sh_audio->wf->nBlockAlign * 2 / 3) { - mp_msg(MSGT_DECAUDIO, MSGL_V, "dk3adpcm: maxlen too small in decode_audio\n"); - return -1; - } - return 2 * dk3_adpcm_decode_block( - (unsigned short*)buf, sh_audio->a_in_buffer, - sh_audio->ds->ss_mul); -} diff --git a/libmpcodecs/ad_libvorbis.c b/libmpcodecs/ad_libvorbis.c deleted file mode 100644 index a768095187..0000000000 --- a/libmpcodecs/ad_libvorbis.c +++ /dev/null @@ -1,350 +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. - */ - -#include <stdio.h> -#include <stdlib.h> -#include <unistd.h> -#include <stdarg.h> -#include <math.h> - -#include "config.h" -#include "ad_internal.h" -#include "libaf/reorder_ch.h" - -static const ad_info_t info = -{ - "Ogg/Vorbis audio decoder", -#ifdef CONFIG_TREMOR - "tremor", -#else - "libvorbis", -#endif - "Felix Buenemann, A'rpi", - "libvorbis", - "" -}; - -LIBAD_EXTERN(libvorbis) - -#ifdef CONFIG_TREMOR -#include <tremor/ivorbiscodec.h> -#else -#include <vorbis/codec.h> -#endif - -// This struct is also defined in demux_ogg.c => common header ? -typedef struct ov_struct_st { - vorbis_info vi; /* struct that stores all the static vorbis bitstream - settings */ - vorbis_comment vc; /* struct that stores all the bitstream user comments */ - vorbis_dsp_state vd; /* central working state for the packet->PCM decoder */ - vorbis_block vb; /* local working space for packet->PCM decode */ - float rg_scale; /* replaygain scale */ -#ifdef CONFIG_TREMOR - int rg_scale_int; -#endif -} ov_struct_t; - -static int read_vorbis_comment( char* ptr, const char* comment, const char* format, ... ) { - va_list va; - int clen, ret; - - va_start( va, format ); - clen = strlen( comment ); - ret = strncasecmp( ptr, comment, clen) == 0 ? vsscanf( ptr+clen, format, va ) : 0; - va_end( va ); - - return ret; -} - -static int preinit(sh_audio_t *sh) -{ - sh->audio_out_minsize=1024*4; // 1024 samples/frame - return 1; -} - -static int init(sh_audio_t *sh) -{ - unsigned int offset, i, length, hsizes[3]; - void *headers[3]; - unsigned char* extradata; - ogg_packet op; - vorbis_comment vc; - struct ov_struct_st *ov; -#define ERROR() { \ - vorbis_comment_clear(&vc); \ - vorbis_info_clear(&ov->vi); \ - free(ov); \ - return 0; \ - } - - /// Init the decoder with the 3 header packets - ov = malloc(sizeof(struct ov_struct_st)); - vorbis_info_init(&ov->vi); - vorbis_comment_init(&vc); - - if(! sh->wf) { - mp_msg(MSGT_DECAUDIO,MSGL_ERR,"ad_vorbis, extradata seems to be absent! exit\n"); - ERROR(); - } - - if(! sh->wf->cbSize) { - mp_msg(MSGT_DECAUDIO,MSGL_ERR,"ad_vorbis, extradata seems to be absent!, exit\n"); - ERROR(); - } - - mp_msg(MSGT_DECAUDIO,MSGL_V,"ad_vorbis, extradata seems is %d bytes long\n", sh->wf->cbSize); - extradata = (char*) (sh->wf+1); - if(!extradata) { - mp_msg(MSGT_DECAUDIO,MSGL_ERR,"ad_vorbis, extradata seems to be NULL!, exit\n"); - ERROR(); - } - - if(*extradata != 2) { - mp_msg (MSGT_DEMUX, MSGL_WARN, "ad_vorbis: Vorbis track does not contain valid headers.\n"); - ERROR(); - } - - offset = 1; - for (i=0; i < 2; i++) { - length = 0; - while ((extradata[offset] == (unsigned char) 0xFF) && length < sh->wf->cbSize) { - length += 255; - offset++; - } - if(offset >= (sh->wf->cbSize - 1)) { - mp_msg (MSGT_DEMUX, MSGL_WARN, "ad_vorbis: Vorbis track does not contain valid headers.\n"); - ERROR(); - } - length += extradata[offset]; - offset++; - mp_msg (MSGT_DEMUX, MSGL_V, "ad_vorbis, offset: %u, length: %u\n", offset, length); - hsizes[i] = length; - } - - headers[0] = &extradata[offset]; - headers[1] = &extradata[offset + hsizes[0]]; - headers[2] = &extradata[offset + hsizes[0] + hsizes[1]]; - hsizes[2] = sh->wf->cbSize - offset - hsizes[0] - hsizes[1]; - mp_msg (MSGT_DEMUX, MSGL_V, "ad_vorbis, header sizes: %d %d %d\n", hsizes[0], hsizes[1], hsizes[2]); - - for(i=0; i<3; i++) { - op.bytes = hsizes[i]; - op.packet = headers[i]; - op.b_o_s = (i == 0); - if(vorbis_synthesis_headerin(&ov->vi,&vc,&op) <0) { - mp_msg(MSGT_DECAUDIO,MSGL_ERR,"OggVorbis: header n. %d broken! len=%ld\n", i, op.bytes); - ERROR(); - } - if(i == 2) { - float rg_gain=0.f, rg_peak=0.f; - char **ptr=vc.user_comments; - while(*ptr){ - mp_msg(MSGT_DECAUDIO,MSGL_V,"OggVorbisComment: %s\n",*ptr); - /* replaygain */ - read_vorbis_comment( *ptr, "replaygain_album_gain=", "%f", &rg_gain ); - read_vorbis_comment( *ptr, "rg_audiophile=", "%f", &rg_gain ); - if( !rg_gain ) { - read_vorbis_comment( *ptr, "replaygain_track_gain=", "%f", &rg_gain ); - read_vorbis_comment( *ptr, "rg_radio=", "%f", &rg_gain ); - } - read_vorbis_comment( *ptr, "replaygain_album_peak=", "%f", &rg_peak ); - if( !rg_peak ) { - read_vorbis_comment( *ptr, "replaygain_track_peak=", "%f", &rg_peak ); - read_vorbis_comment( *ptr, "rg_peak=", "%f", &rg_peak ); - } - ++ptr; - } - /* replaygain: scale */ - if(!rg_gain) - ov->rg_scale = 1.f; /* just in case pow() isn't standard-conformant */ - else - ov->rg_scale = pow(10.f, rg_gain/20); - /* replaygain: anticlip */ - if(ov->rg_scale * rg_peak > 1.f) - ov->rg_scale = 1.f / rg_peak; - /* replaygain: security */ - if(ov->rg_scale > 15.) - ov->rg_scale = 15.; -#ifdef CONFIG_TREMOR - ov->rg_scale_int = (int)(ov->rg_scale*64.f); -#endif - mp_msg(MSGT_DECAUDIO,MSGL_V,"OggVorbis: Bitstream is %d channel%s, %dHz, %dbit/s %cBR\n",(int)ov->vi.channels,ov->vi.channels>1?"s":"",(int)ov->vi.rate,(int)ov->vi.bitrate_nominal, - (ov->vi.bitrate_lower!=ov->vi.bitrate_nominal)||(ov->vi.bitrate_upper!=ov->vi.bitrate_nominal)?'V':'C'); - if(rg_gain || rg_peak) - mp_msg(MSGT_DECAUDIO,MSGL_V,"OggVorbis: Gain = %+.2f dB, Peak = %.4f, Scale = %.2f\n", rg_gain, rg_peak, ov->rg_scale); - mp_msg(MSGT_DECAUDIO,MSGL_V,"OggVorbis: Encoded by: %s\n",vc.vendor); - } - } - - vorbis_comment_clear(&vc); - -// printf("lower=%d upper=%d \n",(int)ov->vi.bitrate_lower,(int)ov->vi.bitrate_upper); - - // Setup the decoder - sh->channels=ov->vi.channels; - sh->samplerate=ov->vi.rate; - sh->samplesize=2; - // assume 128kbit if bitrate not specified in the header - sh->i_bps=((ov->vi.bitrate_nominal>0) ? ov->vi.bitrate_nominal : 128000)/8; - sh->context = ov; - - /// Finish the decoder init - vorbis_synthesis_init(&ov->vd,&ov->vi); - vorbis_block_init(&ov->vd,&ov->vb); - mp_msg(MSGT_DECAUDIO,MSGL_V,"OggVorbis: Init OK!\n"); - - return 1; -} - -static void uninit(sh_audio_t *sh) -{ - struct ov_struct_st *ov = sh->context; - vorbis_dsp_clear(&ov->vd); - vorbis_block_clear(&ov->vb); - vorbis_info_clear(&ov->vi); - free(ov); -} - -static int control(sh_audio_t *sh,int cmd,void* arg, ...) -{ - switch(cmd) - { -#if 0 - case ADCTRL_RESYNC_STREAM: - return CONTROL_TRUE; - case ADCTRL_SKIP_FRAME: - return CONTROL_TRUE; -#endif - } - return CONTROL_UNKNOWN; -} - -static int decode_audio(sh_audio_t *sh,unsigned char *buf,int minlen,int maxlen) -{ - int len = 0; - int samples; -#ifdef CONFIG_TREMOR - ogg_int32_t **pcm; -#else - float scale; - float **pcm; -#endif - struct ov_struct_st *ov = sh->context; - while(len < minlen) { - while((samples=vorbis_synthesis_pcmout(&ov->vd,&pcm))<=0){ - ogg_packet op; - double pts; - memset(&op,0,sizeof(op)); //op.b_o_s = op.e_o_s = 0; - op.bytes = ds_get_packet_pts(sh->ds,&op.packet, &pts); - if(op.bytes<=0) break; - if (pts != MP_NOPTS_VALUE) { - sh->pts = pts; - sh->pts_bytes = 0; - } - if(vorbis_synthesis(&ov->vb,&op)==0) /* test for success! */ - vorbis_synthesis_blockin(&ov->vd,&ov->vb); - } - if(samples<=0) break; // error/EOF - while(samples>0){ - int i,j; - int clipflag=0; - int convsize=(maxlen-len)/(2*ov->vi.channels); // max size! - int bout=((samples<convsize)?samples:convsize); - - if(bout<=0) break; // no buffer space - - /* convert floats to 16 bit signed ints (host order) and - interleave */ -#ifdef CONFIG_TREMOR - if (ov->rg_scale_int == 64) { - for(i=0;i<ov->vi.channels;i++){ - ogg_int16_t *convbuffer=(ogg_int16_t *)(&buf[len]); - ogg_int16_t *ptr=convbuffer+i; - ogg_int32_t *mono=pcm[i]; - for(j=0;j<bout;j++){ - int val=mono[j]>>9; - /* might as well guard against clipping */ - if(val>32767){ - val=32767; - clipflag=1; - } - if(val<-32768){ - val=-32768; - clipflag=1; - } - *ptr=val; - ptr+=ov->vi.channels; - } - } - } else -#endif /* CONFIG_TREMOR */ - { -#ifndef CONFIG_TREMOR - scale = 32767.f * ov->rg_scale; -#endif - for(i=0;i<ov->vi.channels;i++){ - ogg_int16_t *convbuffer=(ogg_int16_t *)(&buf[len]); - ogg_int16_t *ptr=convbuffer+i; -#ifdef CONFIG_TREMOR - ogg_int32_t *mono=pcm[i]; - for(j=0;j<bout;j++){ - int val=(mono[j]*ov->rg_scale_int)>>(9+6); -#else - float *mono=pcm[i]; - for(j=0;j<bout;j++){ - int val=mono[j]*scale; - /* might as well guard against clipping */ - if(val>32767){ - val=32767; - clipflag=1; - } - if(val<-32768){ - val=-32768; - clipflag=1; - } -#endif /* CONFIG_TREMOR */ - *ptr=val; - ptr+=ov->vi.channels; - } - } - } - - if(clipflag) - mp_msg(MSGT_DECAUDIO,MSGL_DBG2,"Clipping in frame %ld\n",(long)(ov->vd.sequence)); - len+=2*ov->vi.channels*bout; - sh->pts_bytes += 2*ov->vi.channels*bout; - mp_msg(MSGT_DECAUDIO,MSGL_DBG2,"\n[decoded: %d / %d ]\n",bout,samples); - samples-=bout; - vorbis_synthesis_read(&ov->vd,bout); /* tell libvorbis how - many samples we - actually consumed */ - } //while(samples>0) -// if (!samples) break; // why? how? - } - - if (len > 0 && ov->vi.channels >= 5) { - reorder_channel_nch(buf, AF_CHANNEL_LAYOUT_VORBIS_DEFAULT, - AF_CHANNEL_LAYOUT_MPLAYER_DEFAULT, - ov->vi.channels, len / sh->samplesize, - sh->samplesize); - } - - - return len; -} diff --git a/libmpcodecs/ad_mpc.c b/libmpcodecs/ad_mpc.c deleted file mode 100644 index 979dce6178..0000000000 --- a/libmpcodecs/ad_mpc.c +++ /dev/null @@ -1,231 +0,0 @@ -/* - * Musepack audio files decoder for MPlayer - * by Reza Jelveh <reza.jelveh@tuhh.de> and - * Reimar Döffinger <Reimar.Doeffinger@stud.uni-karlsruhe.de> - * - * This code may be be relicensed under the terms of the GNU LGPL when it - * becomes part of the FFmpeg project (ffmpeg.org) - * - * 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. - */ - -#include <stdio.h> -#include <stdlib.h> -#include <unistd.h> - -#include "config.h" -#include "ad_internal.h" -#include "libaf/af_format.h" -#include "libvo/fastmemcpy.h" - -static const ad_info_t info = -{ - "Musepack audio decoder", - "mpcdec", - "Reza Jelveh and Reimar Döffinger", - "", - "" -}; - -LIBAD_EXTERN(libmusepack) - -#include <mpcdec/mpcdec.h> - -// BUFFER_LENGTH is in MPC_SAMPLE_FORMAT units -#define MAX_FRAMESIZE (4 * MPC_DECODER_BUFFER_LENGTH) -//! this many frames should decode good after seeking -#define MIN_SEEK_GOOD 5 -//! how many frames to discard at most after seeking -#define MAX_SEEK_DISCARD 50 - -typedef struct context_s { - char *header; - int header_len; - sh_audio_t *sh; - uint32_t pos; - mpc_decoder decoder; -} context_t; - -/** - * \brief mpc_reader callback function for reading the header - */ -static mpc_int32_t cb_read(void *data, void *buf, mpc_int32_t size) { - context_t *d = (context_t *)data; - char *p = (char *)buf; - int s = size; - if (d->pos < d->header_len) { - if (s > d->header_len - d->pos) - s = d->header_len - d->pos; - fast_memcpy(p, &d->header[d->pos], s); - } else - s = 0; - memset(&p[s], 0, size - s); - d->pos += size; - return size; -} - -/** - * \brief dummy mpc_reader callback function for seeking - */ -static mpc_bool_t cb_seek(void *data, mpc_int32_t offset ) { - context_t *d = (context_t *)data; - d->pos = offset; - return 1; -} - -/** - * \brief dummy mpc_reader callback function for getting stream position - */ -static mpc_int32_t cb_tell(void *data) { - context_t *d = (context_t *)data; - return d->pos; -} - -/** - * \brief dummy mpc_reader callback function for getting stream length - */ -static mpc_int32_t cb_get_size(void *data) { - return 1 << 30; -} - -/** - * \brief mpc_reader callback function, we cannot seek. - */ -static mpc_bool_t cb_canseek(void *data) { - return 0; -} - - -mpc_reader header_reader = { - .read = cb_read, .seek = cb_seek, .tell = cb_tell, - .get_size = cb_get_size, .canseek = cb_canseek -}; - -static int preinit(sh_audio_t *sh) { - sh->audio_out_minsize = MAX_FRAMESIZE; - return 1; -} - -static void uninit(sh_audio_t *sh) { - free(sh->context); - sh->context = NULL; -} - -static int init(sh_audio_t *sh) { - mpc_streaminfo info; - context_t *cd = malloc(sizeof(context_t)); - - if (!sh->wf || (sh->wf->cbSize < 6 * 4)) { - mp_msg(MSGT_DECAUDIO, MSGL_FATAL, "Missing extradata!\n"); - return 0; - } - cd->header = (char *)(sh->wf + 1); - cd->header_len = sh->wf->cbSize; - cd->sh = sh; - cd->pos = 0; - sh->context = (char *)cd; - - /* read file's streaminfo data */ - mpc_streaminfo_init(&info); - header_reader.data = cd; - if (mpc_streaminfo_read(&info, &header_reader) != ERROR_CODE_OK) { - mp_msg(MSGT_DECAUDIO, MSGL_FATAL, "Not a valid musepack file.\n"); - return 0; - } -// this value is nonsense, since it relies on the get_size function. -// use the value from the demuxer instead. -// sh->i_bps = info.average_bitrate / 8; - sh->channels = info.channels; - sh->samplerate = info.sample_freq; - sh->samplesize = 4; - sh->sample_format = -#if MPC_SAMPLE_FORMAT == float - AF_FORMAT_FLOAT_NE; -#elif MPC_SAMPLE_FORMAT == mpc_int32_t - AF_FORMAT_S32_NE; -#else - #error musepack lib must use either float or mpc_int32_t sample format -#endif - - mpc_decoder_setup(&cd->decoder, NULL); - mpc_decoder_set_streaminfo(&cd->decoder, &info); - return 1; -} - -// FIXME: minlen is currently ignored -static int decode_audio(sh_audio_t *sh, unsigned char *buf, - int minlen, int maxlen) { - int status, len; - MPC_SAMPLE_FORMAT *sample_buffer = (MPC_SAMPLE_FORMAT *)buf; - mpc_uint32_t *packet = NULL; - - context_t *cd = (context_t *) sh->context; - if (maxlen < MAX_FRAMESIZE) { - mp_msg(MSGT_DECAUDIO, MSGL_V, "maxlen too small in decode_audio\n"); - return -1; - } - len = ds_get_packet(sh->ds, (unsigned char **)&packet); - if (len <= 0) return -1; - status = mpc_decoder_decode_frame(&cd->decoder, packet, len, sample_buffer); - if (status == -1) // decode error - mp_msg(MSGT_DECAUDIO, MSGL_FATAL, "Error decoding file.\n"); - if (status <= 0) // error or EOF - return -1; - - status = MPC_FRAME_LENGTH * sh->channels; // one sample per channel -#if MPC_SAMPLE_FORMAT == float || MPC_SAMPLE_FORMAT == mpc_int32_t - status *= 4; -#else - // should not happen - status *= 2; -#endif - return status; -} - -/** - * \brief check if the decoded values are in a sane range - * \param buf decoded buffer - * \param len length of buffer in bytes - * \return 1 if all values are in (-1.01, 1.01) range, 0 otherwise - */ -static int check_clip(void *buf, int len) { -#if MPC_SAMPLE_FORMAT == float - float *p = buf; - if (len < 4) return 1; - len = -len / 4; - p = &p[-len]; - do { - if (p[len] < -1 || p[len] > 1) return 0; - } while (++len); -#endif - return 1; -} - -static int control(sh_audio_t *sh, int cmd, void* arg, ...) { - if (cmd == ADCTRL_RESYNC_STREAM) { - unsigned char *buf = malloc(MAX_FRAMESIZE); - int i; - int nr_ok = 0; - for (i = 0; i < MAX_SEEK_DISCARD; i++) { - int len = decode_audio(sh, buf, 0, MAX_FRAMESIZE); - if (check_clip(buf, len)) nr_ok++; else nr_ok = 0; - if (nr_ok > MIN_SEEK_GOOD) break; - } - free(buf); - } - return CONTROL_UNKNOWN; -} diff --git a/libmpcodecs/ad_speex.c b/libmpcodecs/ad_speex.c deleted file mode 100644 index c9ac1a42c6..0000000000 --- a/libmpcodecs/ad_speex.c +++ /dev/null @@ -1,179 +0,0 @@ -/* - * Speex decoder by Reimar Döffinger <Reimar.Doeffinger@stud.uni-karlsruhe.de> - * - * This code may be be relicensed under the terms of the GNU LGPL when it - * becomes part of the FFmpeg project (ffmpeg.org) - * - * 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. - */ - -#include "config.h" -#include <stdlib.h> -#include <speex/speex.h> -#include <speex/speex_stereo.h> -#include <speex/speex_header.h> -#include "ad_internal.h" - -static const ad_info_t info = { - "Speex audio decoder", - "speex", - "Reimar Döffinger", - "", - "" -}; - -LIBAD_EXTERN(speex) - -typedef struct { - SpeexBits bits; - void *dec_context; - SpeexStereoState stereo; - SpeexHeader *hdr; -} context_t; - -#define MAX_FRAMES_PER_PACKET 100 - -static int preinit(sh_audio_t *sh) { - sh->audio_out_minsize = 2 * 320 * MAX_FRAMES_PER_PACKET * 2 * sizeof(short); - return 1; -} - -static int read_le32(const uint8_t **src) { - const uint8_t *p = *src; - *src += 4; - return p[0] + (p[1] << 8) + (p[2] << 16) + (p[3] << 24); -} - -static int init(sh_audio_t *sh) { - context_t *ctx = calloc(1, sizeof(context_t)); - const uint8_t *hdr = (const uint8_t *)(sh->wf + 1); - const SpeexMode *spx_mode; - const SpeexStereoState st_st = SPEEX_STEREO_STATE_INIT; // hack - if (sh->wf && sh->wf->cbSize >= 80) - ctx->hdr = speex_packet_to_header((char *)&sh->wf[1], sh->wf->cbSize); - if (!ctx->hdr && sh->wf->cbSize == 0x72 && hdr[0] == 1 && hdr[1] == 0) { - // speex.acm format: raw SpeexHeader dump - ctx->hdr = calloc(1, sizeof(*ctx->hdr)); - hdr += 2; - hdr += 8; // identifier string - hdr += 20; // version string - ctx->hdr->speex_version_id = read_le32(&hdr); - ctx->hdr->header_size = read_le32(&hdr); - ctx->hdr->rate = read_le32(&hdr); - ctx->hdr->mode = read_le32(&hdr); - ctx->hdr->mode_bitstream_version = read_le32(&hdr); - ctx->hdr->nb_channels = read_le32(&hdr); - ctx->hdr->bitrate = read_le32(&hdr); - ctx->hdr->frame_size = read_le32(&hdr); - ctx->hdr->vbr = read_le32(&hdr); - ctx->hdr->frames_per_packet = read_le32(&hdr); - } - if (!ctx->hdr) { - mp_msg(MSGT_DECAUDIO, MSGL_ERR, "Invalid or missing extradata! Assuming defaults.\n"); - ctx->hdr = calloc(1, sizeof(*ctx->hdr)); - ctx->hdr->frames_per_packet = 1; - ctx->hdr->mode = 0; - if (sh->wf) { - ctx->hdr->nb_channels = sh->wf->nChannels; - ctx->hdr->rate = sh->wf->nSamplesPerSec; - if (ctx->hdr->rate > 16000) - ctx->hdr->mode = 2; - else if (ctx->hdr->rate > 8000) - ctx->hdr->mode = 1; - } - } - if (ctx->hdr->nb_channels != 1 && ctx->hdr-&g |