libmpcodecs/ad_ffmpeg.c

   1 /*
   2  * This file is part of MPlayer.
   3  *
   4  * MPlayer is free software; you can redistribute it and/or modify
   5  * it under the terms of the GNU General Public License as published by
   6  * the Free Software Foundation; either version 2 of the License, or
   7  * (at your option) any later version.
   8  *
   9  * MPlayer is distributed in the hope that it will be useful,
  10  * but WITHOUT ANY WARRANTY; without even the implied warranty of
  11  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
  12  * GNU General Public License for more details.
  13  *
  14  * You should have received a copy of the GNU General Public License along
  15  * with MPlayer; if not, write to the Free Software Foundation, Inc.,
  16  * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
  17  */
  18
  19 #include <stdio.h>
  20 #include <stdlib.h>
  21 #include <unistd.h>
  22
  23 #include "config.h"
  24 #include "mp_msg.h"
  25 #include "options.h"
  26
  27 #include "ad_internal.h"
  28 #include "libaf/reorder_ch.h"
  29
  30 #include "mpbswap.h"
  31
  32 static const ad_info_t info =
  33 {
  34         "FFmpeg/libavcodec audio decoders",
  35         "ffmpeg",
  36         "Nick Kurshev",
  37         "ffmpeg.sf.net",
  38         ""
  39 };
  40
  41 LIBAD_EXTERN(ffmpeg)
  42
  43 #define assert(x)
  44
  45 #include "libavcodec/avcodec.h"
  46
  47 extern int avcodec_initialized;
  48
  49 static int preinit(sh_audio_t *sh)
  50 {
  51   sh->audio_out_minsize=AVCODEC_MAX_AUDIO_FRAME_SIZE;
  52   return 1;
  53 }
  54
  55 static int setup_format(sh_audio_t *sh_audio, const AVCodecContext *lavc_context)
  56 {
  57     int broken_srate = 0;
  58     int samplerate    = lavc_context->sample_rate;
  59     int sample_format = sh_audio->sample_format;
  60     switch (lavc_context->sample_fmt) {
  61         case SAMPLE_FMT_U8:  sample_format = AF_FORMAT_U8;       break;
  62         case SAMPLE_FMT_S16: sample_format = AF_FORMAT_S16_NE;   break;
  63         case SAMPLE_FMT_S32: sample_format = AF_FORMAT_S32_NE;   break;
  64         case SAMPLE_FMT_FLT: sample_format = AF_FORMAT_FLOAT_NE; break;
  65         default:
  66             mp_msg(MSGT_DECAUDIO, MSGL_FATAL, "Unsupported sample format\n");
  67     }
  68     if(sh_audio->wf){
  69         // If the decoder uses the wrong number of channels all is lost anyway.
  70         // sh_audio->channels=sh_audio->wf->nChannels;
  71
  72         if (lavc_context->codec_id == CODEC_ID_AAC &&
  73             samplerate == 2*sh_audio->wf->nSamplesPerSec) {
  74             broken_srate = 1;
  75         } else if (sh_audio->wf->nSamplesPerSec)
  76             samplerate=sh_audio->wf->nSamplesPerSec;
  77     }
  78     if (lavc_context->channels != sh_audio->channels ||
  79         samplerate != sh_audio->samplerate ||
  80         sample_format != sh_audio->sample_format) {
  81         sh_audio->channels=lavc_context->channels;
  82         sh_audio->samplerate=samplerate;
  83         sh_audio->sample_format = sample_format;
  84         sh_audio->samplesize=af_fmt2bits(sh_audio->sample_format)/ 8;
  85         if (broken_srate)
  86             mp_msg(MSGT_DECAUDIO, MSGL_WARN,
  87                    "Ignoring broken container sample rate for AAC with SBR\n");
  88         return 1;
  89     }
  90     return 0;
  91 }
  92
  93 static int init(sh_audio_t *sh_audio)
  94 {
  95     struct MPOpts *opts = sh_audio->opts;
  96     int tries = 0;
  97     int x;
  98     AVCodecContext *lavc_context;
  99     AVCodec *lavc_codec;
 100
 101     mp_msg(MSGT_DECAUDIO,MSGL_V,"FFmpeg's libavcodec audio codec\n");
 102     if(!avcodec_initialized){
 103       avcodec_init();
 104       avcodec_register_all();
 105       avcodec_initialized=1;
 106     }
 107
 108     lavc_codec = (AVCodec *)avcodec_find_decoder_by_name(sh_audio->codec->dll);
 109     if(!lavc_codec){
 110         mp_tmsg(MSGT_DECAUDIO,MSGL_ERR,"Cannot find codec '%s' in libavcodec...\n",sh_audio->codec->dll);
 111         return 0;
 112     }
 113
 114     lavc_context = avcodec_alloc_context();
 115     sh_audio->context=lavc_context;
 116
 117     lavc_context->drc_scale = opts->drc_level;
 118     lavc_context->sample_rate = sh_audio->samplerate;
 119     lavc_context->bit_rate = sh_audio->i_bps * 8;
 120     if(sh_audio->wf){
 121         lavc_context->channels = sh_audio->wf->nChannels;
 122         lavc_context->sample_rate = sh_audio->wf->nSamplesPerSec;
 123         lavc_context->bit_rate = sh_audio->wf->nAvgBytesPerSec * 8;
 124         lavc_context->block_align = sh_audio->wf->nBlockAlign;
 125         lavc_context->bits_per_coded_sample = sh_audio->wf->wBitsPerSample;
 126     }
 127     lavc_context->request_channels = audio_output_channels;
 128     lavc_context->codec_tag = sh_audio->format; //FOURCC
 129     lavc_context->codec_type = CODEC_TYPE_AUDIO;
 130     lavc_context->codec_id = lavc_codec->id; // not sure if required, imho not --A'rpi
 131
 132     /* alloc extra data */
 133     if (sh_audio->wf && sh_audio->wf->cbSize > 0) {
 134         lavc_context->extradata = av_mallocz(sh_audio->wf->cbSize + FF_INPUT_BUFFER_PADDING_SIZE);
 135         lavc_context->extradata_size = sh_audio->wf->cbSize;
 136         memcpy(lavc_context->extradata, (char *)sh_audio->wf + sizeof(WAVEFORMATEX),
 137                lavc_context->extradata_size);
 138     }
 139
 140     // for QDM2
 141     if (sh_audio->codecdata_len && sh_audio->codecdata && !lavc_context->extradata)
 142     {
 143         lavc_context->extradata = av_malloc(sh_audio->codecdata_len);
 144         lavc_context->extradata_size = sh_audio->codecdata_len;
 145         memcpy(lavc_context->extradata, (char *)sh_audio->codecdata,
 146                lavc_context->extradata_size);
 147     }
 148
 149     /* open it */
 150     if (avcodec_open(lavc_context, lavc_codec) < 0) {
 151         mp_tmsg(MSGT_DECAUDIO,MSGL_ERR, "Could not open codec.\n");
 152         return 0;
 153     }
 154    mp_msg(MSGT_DECAUDIO,MSGL_V,"INFO: libavcodec \"%s\" init OK!\n", lavc_codec->name);
 155
 156 //   printf("\nFOURCC: 0x%X\n",sh_audio->format);
 157    if(sh_audio->format==0x3343414D){
 158        // MACE 3:1
 159        sh_audio->ds->ss_div = 2*3; // 1 samples/packet
 160        sh_audio->ds->ss_mul = 2*sh_audio->wf->nChannels; // 1 byte*ch/packet
 161    } else
 162    if(sh_audio->format==0x3643414D){
 163        // MACE 6:1
 164        sh_audio->ds->ss_div = 2*6; // 1 samples/packet
 165        sh_audio->ds->ss_mul = 2*sh_audio->wf->nChannels; // 1 byte*ch/packet
 166    }
 167
 168    // Decode at least 1 byte:  (to get header filled)
 169    do {
 170        x=decode_audio(sh_audio,sh_audio->a_buffer,1,sh_audio->a_buffer_size);
 171    } while (x <= 0 && tries++ < 5);
 172    if(x>0) sh_audio->a_buffer_len=x;
 173
 174   sh_audio->i_bps=lavc_context->bit_rate/8;
 175   if (sh_audio->wf && sh_audio->wf->nAvgBytesPerSec)
 176       sh_audio->i_bps=sh_audio->wf->nAvgBytesPerSec;
 177
 178   switch (lavc_context->sample_fmt) {
 179       case SAMPLE_FMT_U8:
 180       case SAMPLE_FMT_S16:
 181       case SAMPLE_FMT_S32:
 182       case SAMPLE_FMT_FLT:
 183           break;
 184       default:
 185           return 0;
 186   }
 187   return 1;
 188 }
 189
 190 static void uninit(sh_audio_t *sh)
 191 {
 192     AVCodecContext *lavc_context = sh->context;
 193
 194     if (avcodec_close(lavc_context) < 0)
 195         mp_tmsg(MSGT_DECVIDEO, MSGL_ERR, "Could not close codec.\n");
 196     av_freep(&lavc_context->extradata);
 197     av_freep(&lavc_context);
 198 }
 199
 200 static int control(sh_audio_t *sh,int cmd,void* arg, ...)
 201 {
 202     AVCodecContext *lavc_context = sh->context;
 203     switch(cmd){
 204     case ADCTRL_RESYNC_STREAM:
 205         avcodec_flush_buffers(lavc_context);
 206         ds_clear_parser(sh->ds);
 207     return CONTROL_TRUE;
 208     }
 209     return CONTROL_UNKNOWN;
 210 }
 211
 212 static int decode_audio(sh_audio_t *sh_audio,unsigned char *buf,int minlen,int maxlen)
 213 {
 214     unsigned char *start=NULL;
 215     int y,len=-1;
 216     while(len<minlen){
 217         AVPacket pkt;
 218         int len2=maxlen;
 219         double pts;
 220         int x=ds_get_packet_pts(sh_audio->ds,&start, &pts);
 221         if(x<=0) {
 222             start = NULL;
 223             x = 0;
 224             ds_parse(sh_audio->ds, &start, &x, MP_NOPTS_VALUE, 0);
 225             if (x <= 0)
 226                 break; // error
 227         } else {
 228             int in_size = x;
 229             int consumed = ds_parse(sh_audio->ds, &start, &x, pts, 0);
 230             sh_audio->ds->buffer_pos -= in_size - consumed;
 231         }
 232         av_init_packet(&pkt);
 233         pkt.data = start;
 234         pkt.size = x;
 235         if (pts != MP_NOPTS_VALUE) {
 236             sh_audio->pts = pts;
 237             sh_audio->pts_bytes = 0;
 238         }
 239         y=avcodec_decode_audio3(sh_audio->context,(int16_t*)buf,&len2,&pkt);
 240 //printf("return:%d samples_out:%d bitstream_in:%d sample_sum:%d\n", y, len2, x, len); fflush(stdout);
 241         if(y<0){ mp_msg(MSGT_DECAUDIO,MSGL_V,"lavc_audio: error\n");break; }
 242         if(!sh_audio->parser && y<x)
 243             sh_audio->ds->buffer_pos+=y-x;  // put back data (HACK!)
 244         if(len2>0){
 245           if (((AVCodecContext *)sh_audio->context)->channels >= 5) {
 246             int samplesize = av_get_bits_per_sample_format(((AVCodecContext *)
 247                                     sh_audio->context)->sample_fmt) / 8;
 248             reorder_channel_nch(buf, AF_CHANNEL_LAYOUT_LAVC_DEFAULT,
 249                                 AF_CHANNEL_LAYOUT_MPLAYER_DEFAULT,
 250                                 ((AVCodecContext *)sh_audio->context)->channels,
 251                                 len2 / samplesize, samplesize);
 252           }
 253           //len=len2;break;
 254           if(len<0) len=len2; else len+=len2;
 255           buf+=len2;
 256           maxlen -= len2;
 257           sh_audio->pts_bytes += len2;
 258         }
 259         mp_dbg(MSGT_DECAUDIO,MSGL_DBG2,"Decoded %d -> %d  \n",y,len2);
 260
 261         if (setup_format(sh_audio, sh_audio->context))
 262             break;
 263     }
 264   return len;
 265 }