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 "vd_ffmpeg.h"
  29 #include "libaf/reorder_ch.h"
  30
  31 #include "mpbswap.h"
  32
  33 static const ad_info_t info =
  34 {
  35         "FFmpeg/libavcodec audio decoders",
  36         "ffmpeg",
  37         "Nick Kurshev",
  38         "ffmpeg.sf.net",
  39         ""
  40 };
  41
  42 LIBAD_EXTERN(ffmpeg)
  43
  44 #define assert(x)
  45
  46 #include "libavcodec/avcodec.h"
  47
  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     init_avcodec();
 103
 104     lavc_codec = avcodec_find_decoder_by_name(sh_audio->codec->dll);
 105     if(!lavc_codec){
 106         mp_tmsg(MSGT_DECAUDIO,MSGL_ERR,"Cannot find codec '%s' in libavcodec...\n",sh_audio->codec->dll);
 107         return 0;
 108     }
 109
 110     lavc_context = avcodec_alloc_context();
 111     sh_audio->context=lavc_context;
 112
 113     lavc_context->drc_scale = opts->drc_level;
 114     lavc_context->sample_rate = sh_audio->samplerate;
 115     lavc_context->bit_rate = sh_audio->i_bps * 8;
 116     if(sh_audio->wf){
 117         lavc_context->channels = sh_audio->wf->nChannels;
 118         lavc_context->sample_rate = sh_audio->wf->nSamplesPerSec;
 119         lavc_context->bit_rate = sh_audio->wf->nAvgBytesPerSec * 8;
 120         lavc_context->block_align = sh_audio->wf->nBlockAlign;
 121         lavc_context->bits_per_coded_sample = sh_audio->wf->wBitsPerSample;
 122     }
 123     lavc_context->request_channels = opts->audio_output_channels;
 124     lavc_context->codec_tag = sh_audio->format; //FOURCC
 125     lavc_context->codec_type = CODEC_TYPE_AUDIO;
 126     lavc_context->codec_id = lavc_codec->id; // not sure if required, imho not --A'rpi
 127
 128     /* alloc extra data */
 129     if (sh_audio->wf && sh_audio->wf->cbSize > 0) {
 130         lavc_context->extradata = av_mallocz(sh_audio->wf->cbSize + FF_INPUT_BUFFER_PADDING_SIZE);
 131         lavc_context->extradata_size = sh_audio->wf->cbSize;
 132         memcpy(lavc_context->extradata, sh_audio->wf + 1,
 133                lavc_context->extradata_size);
 134     }
 135
 136     // for QDM2
 137     if (sh_audio->codecdata_len && sh_audio->codecdata && !lavc_context->extradata)
 138     {
 139         lavc_context->extradata = av_malloc(sh_audio->codecdata_len);
 140         lavc_context->extradata_size = sh_audio->codecdata_len;
 141         memcpy(lavc_context->extradata, (char *)sh_audio->codecdata,
 142                lavc_context->extradata_size);
 143     }
 144
 145     /* open it */
 146     if (avcodec_open(lavc_context, lavc_codec) < 0) {
 147         mp_tmsg(MSGT_DECAUDIO,MSGL_ERR, "Could not open codec.\n");
 148         return 0;
 149     }
 150    mp_msg(MSGT_DECAUDIO,MSGL_V,"INFO: libavcodec \"%s\" init OK!\n", lavc_codec->name);
 151
 152 //   printf("\nFOURCC: 0x%X\n",sh_audio->format);
 153    if(sh_audio->format==0x3343414D){
 154        // MACE 3:1
 155        sh_audio->ds->ss_div = 2*3; // 1 samples/packet
 156        sh_audio->ds->ss_mul = 2*sh_audio->wf->nChannels; // 1 byte*ch/packet
 157    } else
 158    if(sh_audio->format==0x3643414D){
 159        // MACE 6:1
 160        sh_audio->ds->ss_div = 2*6; // 1 samples/packet
 161        sh_audio->ds->ss_mul = 2*sh_audio->wf->nChannels; // 1 byte*ch/packet
 162    }
 163
 164    // Decode at least 1 byte:  (to get header filled)
 165    do {
 166        x=decode_audio(sh_audio,sh_audio->a_buffer,1,sh_audio->a_buffer_size);
 167    } while (x <= 0 && tries++ < 5);
 168    if(x>0) sh_audio->a_buffer_len=x;
 169
 170   sh_audio->i_bps=lavc_context->bit_rate/8;
 171   if (sh_audio->wf && sh_audio->wf->nAvgBytesPerSec)
 172       sh_audio->i_bps=sh_audio->wf->nAvgBytesPerSec;
 173
 174   switch (lavc_context->sample_fmt) {
 175       case SAMPLE_FMT_U8:
 176       case SAMPLE_FMT_S16:
 177       case SAMPLE_FMT_S32:
 178       case SAMPLE_FMT_FLT:
 179           break;
 180       default:
 181           return 0;
 182   }
 183   return 1;
 184 }
 185
 186 static void uninit(sh_audio_t *sh)
 187 {
 188     AVCodecContext *lavc_context = sh->context;
 189
 190     if (avcodec_close(lavc_context) < 0)
 191         mp_tmsg(MSGT_DECVIDEO, MSGL_ERR, "Could not close codec.\n");
 192     av_freep(&lavc_context->extradata);
 193     av_freep(&lavc_context);
 194 }
 195
 196 static int control(sh_audio_t *sh,int cmd,void* arg, ...)
 197 {
 198     AVCodecContext *lavc_context = sh->context;
 199     switch(cmd){
 200     case ADCTRL_RESYNC_STREAM:
 201         avcodec_flush_buffers(lavc_context);
 202         ds_clear_parser(sh->ds);
 203     return CONTROL_TRUE;
 204     }
 205     return CONTROL_UNKNOWN;
 206 }
 207
 208 static int decode_audio(sh_audio_t *sh_audio,unsigned char *buf,int minlen,int maxlen)
 209 {
 210     unsigned char *start=NULL;
 211     int y,len=-1;
 212     while(len<minlen){
 213         AVPacket pkt;
 214         int len2=maxlen;
 215         double pts;
 216         int x=ds_get_packet_pts(sh_audio->ds,&start, &pts);
 217         if(x<=0) {
 218             start = NULL;
 219             x = 0;
 220             ds_parse(sh_audio->ds, &start, &x, MP_NOPTS_VALUE, 0);
 221             if (x <= 0)
 222                 break; // error
 223         } else {
 224             int in_size = x;
 225             int consumed = ds_parse(sh_audio->ds, &start, &x, pts, 0);
 226             sh_audio->ds->buffer_pos -= in_size - consumed;
 227         }
 228         av_init_packet(&pkt);
 229         pkt.data = start;
 230         pkt.size = x;
 231         if (pts != MP_NOPTS_VALUE) {
 232             sh_audio->pts = pts;
 233             sh_audio->pts_bytes = 0;
 234         }
 235         y=avcodec_decode_audio3(sh_audio->context,(int16_t*)buf,&len2,&pkt);
 236 //printf("return:%d samples_out:%d bitstream_in:%d sample_sum:%d\n", y, len2, x, len); fflush(stdout);
 237         // LATM may need many packets to find mux info
 238         if (y == AVERROR(EAGAIN))
 239             continue;
 240         if(y<0){ mp_msg(MSGT_DECAUDIO,MSGL_V,"lavc_audio: error\n");break; }
 241         if(!sh_audio->parser && y<x)
 242             sh_audio->ds->buffer_pos+=y-x;  // put back data (HACK!)
 243         if(len2>0){
 244           if (((AVCodecContext *)sh_audio->context)->channels >= 5) {
 245             int samplesize = av_get_bits_per_sample_format(((AVCodecContext *)
 246                                     sh_audio->context)->sample_fmt) / 8;
 247             reorder_channel_nch(buf, AF_CHANNEL_LAYOUT_LAVC_DEFAULT,
 248                                 AF_CHANNEL_LAYOUT_MPLAYER_DEFAULT,
 249                                 ((AVCodecContext *)sh_audio->context)->channels,
 250                                 len2 / samplesize, samplesize);
 251           }
 252           //len=len2;break;
 253           if(len<0) len=len2; else len+=len2;
 254           buf+=len2;
 255           maxlen -= len2;
 256           sh_audio->pts_bytes += len2;
 257         }
 258         mp_dbg(MSGT_DECAUDIO,MSGL_DBG2,"Decoded %d -> %d  \n",y,len2);
 259
 260         if (setup_format(sh_audio, sh_audio->context))
 261             break;
 262     }
 263   return len;
 264 }