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 #include <stdbool.h>
  23 #include <assert.h>
  24
  25 #include <libavcodec/avcodec.h>
  26 #include <libavutil/opt.h>
  27
  28 #include "talloc.h"
  29
  30 #include "config.h"
  31 #include "mp_msg.h"
  32 #include "options.h"
  33
  34 #include "ad_internal.h"
  35 #include "libaf/reorder_ch.h"
  36
  37 #include "mpbswap.h"
  38
  39 static const ad_info_t info =
  40 {
  41     "libavcodec audio decoders",
  42     "ffmpeg",
  43     "",
  44     "",
  45     "",
  46     .print_name = "libavcodec",
  47 };
  48
  49 LIBAD_EXTERN(ffmpeg)
  50
  51 struct priv {
  52     AVCodecContext *avctx;
  53     AVFrame *avframe;
  54     char *output;
  55     int output_left;
  56     int unitsize;
  57     int previous_data_left;  // input demuxer packet data
  58 };
  59
  60 static int preinit(sh_audio_t *sh)
  61 {
  62     return 1;
  63 }
  64
  65 /* Prefer playing audio with the samplerate given in container data
  66  * if available, but take number the number of channels and sample format
  67  * from the codec, since if the codec isn't using the correct values for
  68  * those everything breaks anyway.
  69  */
  70 static int setup_format(sh_audio_t *sh_audio,
  71                         const AVCodecContext *lavc_context)
  72 {
  73     int sample_format = sh_audio->sample_format;
  74     switch (lavc_context->sample_fmt) {
  75     case AV_SAMPLE_FMT_U8:  sample_format = AF_FORMAT_U8;       break;
  76     case AV_SAMPLE_FMT_S16: sample_format = AF_FORMAT_S16_NE;   break;
  77     case AV_SAMPLE_FMT_S32: sample_format = AF_FORMAT_S32_NE;   break;
  78     case AV_SAMPLE_FMT_FLT: sample_format = AF_FORMAT_FLOAT_NE; break;
  79     default:
  80         mp_msg(MSGT_DECAUDIO, MSGL_FATAL, "Unsupported sample format\n");
  81         sample_format = AF_FORMAT_UNKNOWN;
  82     }
  83
  84     bool broken_srate        = false;
  85     int samplerate           = lavc_context->sample_rate;
  86     int container_samplerate = sh_audio->container_out_samplerate;
  87     if (!container_samplerate && sh_audio->wf)
  88         container_samplerate = sh_audio->wf->nSamplesPerSec;
  89     if (lavc_context->codec_id == CODEC_ID_AAC
  90         && samplerate == 2 * container_samplerate)
  91         broken_srate = true;
  92     else if (container_samplerate)
  93         samplerate = container_samplerate;
  94
  95     if (lavc_context->channels != sh_audio->channels ||
  96         samplerate != sh_audio->samplerate ||
  97         sample_format != sh_audio->sample_format) {
  98         sh_audio->channels = lavc_context->channels;
  99         sh_audio->samplerate = samplerate;
 100         sh_audio->sample_format = sample_format;
 101         sh_audio->samplesize = af_fmt2bits(sh_audio->sample_format) / 8;
 102         if (broken_srate)
 103             mp_msg(MSGT_DECAUDIO, MSGL_WARN,
 104                    "Ignoring broken container sample rate for AAC with SBR\n");
 105         return 1;
 106     }
 107     return 0;
 108 }
 109
 110 static int init(sh_audio_t *sh_audio)
 111 {
 112     struct MPOpts *opts = sh_audio->opts;
 113     AVCodecContext *lavc_context;
 114     AVCodec *lavc_codec;
 115
 116     if (sh_audio->codec->dll) {
 117         lavc_codec = avcodec_find_decoder_by_name(sh_audio->codec->dll);
 118         if (!lavc_codec) {
 119             mp_tmsg(MSGT_DECAUDIO, MSGL_ERR,
 120                     "Cannot find codec '%s' in libavcodec...\n",
 121                     sh_audio->codec->dll);
 122             return 0;
 123         }
 124     } else if (!sh_audio->libav_codec_id) {
 125         mp_tmsg(MSGT_DECAUDIO, MSGL_INFO, "No Libav codec ID known. "
 126                 "Generic lavc decoder is not applicable.\n");
 127         return 0;
 128     } else {
 129         lavc_codec = avcodec_find_decoder(sh_audio->libav_codec_id);
 130         if (!lavc_codec) {
 131             mp_tmsg(MSGT_DECAUDIO, MSGL_INFO, "Libavcodec has no decoder "
 132                    "for this codec\n");
 133             return 0;
 134         }
 135     }
 136
 137     sh_audio->codecname = lavc_codec->long_name;
 138     if (!sh_audio->codecname)
 139         sh_audio->codecname = lavc_codec->name;
 140
 141     struct priv *ctx = talloc_zero(NULL, struct priv);
 142     sh_audio->context = ctx;
 143     lavc_context = avcodec_alloc_context3(lavc_codec);
 144     ctx->avctx = lavc_context;
 145     ctx->avframe = avcodec_alloc_frame();
 146
 147     // Always try to set - option only exists for AC3 at the moment
 148     av_opt_set_double(lavc_context, "drc_scale", opts->drc_level,
 149                       AV_OPT_SEARCH_CHILDREN);
 150     lavc_context->sample_rate = sh_audio->samplerate;
 151     lavc_context->bit_rate = sh_audio->i_bps * 8;
 152     if (sh_audio->wf) {
 153         lavc_context->channels = sh_audio->wf->nChannels;
 154         lavc_context->sample_rate = sh_audio->wf->nSamplesPerSec;
 155         lavc_context->bit_rate = sh_audio->wf->nAvgBytesPerSec * 8;
 156         lavc_context->block_align = sh_audio->wf->nBlockAlign;
 157         lavc_context->bits_per_coded_sample = sh_audio->wf->wBitsPerSample;
 158     }
 159     lavc_context->request_channels = opts->audio_output_channels;
 160     lavc_context->codec_tag = sh_audio->format; //FOURCC
 161     lavc_context->codec_type = AVMEDIA_TYPE_AUDIO;
 162     lavc_context->codec_id = lavc_codec->id; // not sure if required, imho not --A'rpi
 163
 164     /* alloc extra data */
 165     if (sh_audio->wf && sh_audio->wf->cbSize > 0) {
 166         lavc_context->extradata = av_mallocz(sh_audio->wf->cbSize + FF_INPUT_BUFFER_PADDING_SIZE);
 167         lavc_context->extradata_size = sh_audio->wf->cbSize;
 168         memcpy(lavc_context->extradata, sh_audio->wf + 1,
 169                lavc_context->extradata_size);
 170     }
 171
 172     // for QDM2
 173     if (sh_audio->codecdata_len && sh_audio->codecdata &&
 174             !lavc_context->extradata) {
 175         lavc_context->extradata = av_malloc(sh_audio->codecdata_len +
 176                                             FF_INPUT_BUFFER_PADDING_SIZE);
 177         lavc_context->extradata_size = sh_audio->codecdata_len;
 178         memcpy(lavc_context->extradata, (char *)sh_audio->codecdata,
 179                lavc_context->extradata_size);
 180     }
 181
 182     /* open it */
 183     if (avcodec_open2(lavc_context, lavc_codec, NULL) < 0) {
 184         mp_tmsg(MSGT_DECAUDIO, MSGL_ERR, "Could not open codec.\n");
 185         uninit(sh_audio);
 186         return 0;
 187     }
 188     mp_msg(MSGT_DECAUDIO, MSGL_V, "INFO: libavcodec \"%s\" init OK!\n",
 189            lavc_codec->name);
 190
 191     if (sh_audio->format == 0x3343414D) {
 192         // MACE 3:1
 193         sh_audio->ds->ss_div = 2 * 3; // 1 samples/packet
 194         sh_audio->ds->ss_mul = 2 * sh_audio->wf->nChannels; // 1 byte*ch/packet
 195     } else if (sh_audio->format == 0x3643414D) {
 196         // MACE 6:1
 197         sh_audio->ds->ss_div = 2 * 6; // 1 samples/packet
 198         sh_audio->ds->ss_mul = 2 * sh_audio->wf->nChannels; // 1 byte*ch/packet
 199     }
 200
 201     // Decode at least 1 byte:  (to get header filled)
 202     for (int tries = 0;;) {
 203         int x = decode_audio(sh_audio, sh_audio->a_buffer, 1,
 204                              sh_audio->a_buffer_size);
 205         if (x > 0) {
 206             sh_audio->a_buffer_len = x;
 207             break;
 208         }
 209         if (++tries >= 5) {
 210             mp_msg(MSGT_DECAUDIO, MSGL_ERR,
 211                    "ad_ffmpeg: initial decode failed\n");
 212             uninit(sh_audio);
 213             return 0;
 214         }
 215     }
 216
 217     sh_audio->i_bps = lavc_context->bit_rate / 8;
 218     if (sh_audio->wf && sh_audio->wf->nAvgBytesPerSec)
 219         sh_audio->i_bps = sh_audio->wf->nAvgBytesPerSec;
 220
 221     switch (lavc_context->sample_fmt) {
 222     case AV_SAMPLE_FMT_U8:
 223     case AV_SAMPLE_FMT_S16:
 224     case AV_SAMPLE_FMT_S32:
 225     case AV_SAMPLE_FMT_FLT:
 226         break;
 227     default:
 228         uninit(sh_audio);
 229         return 0;
 230     }
 231     return 1;
 232 }
 233
 234 static void uninit(sh_audio_t *sh)
 235 {
 236     sh->codecname = NULL;
 237     struct priv *ctx = sh->context;
 238     if (!ctx)
 239         return;
 240     AVCodecContext *lavc_context = ctx->avctx;
 241
 242     if (lavc_context) {
 243         if (avcodec_close(lavc_context) < 0)
 244             mp_tmsg(MSGT_DECVIDEO, MSGL_ERR, "Could not close codec.\n");
 245         av_freep(&lavc_context->extradata);
 246         av_freep(&lavc_context);
 247     }
 248     av_free(ctx->avframe);
 249     talloc_free(ctx);
 250     sh->context = NULL;
 251 }
 252
 253 static int control(sh_audio_t *sh, int cmd, void *arg, ...)
 254 {
 255     struct priv *ctx = sh->context;
 256     switch (cmd) {
 257     case ADCTRL_RESYNC_STREAM:
 258         avcodec_flush_buffers(ctx->avctx);
 259         ds_clear_parser(sh->ds);
 260         ctx->previous_data_left = 0;
 261         ctx->output_left = 0;
 262         return CONTROL_TRUE;
 263     }
 264     return CONTROL_UNKNOWN;
 265 }
 266
 267 static int decode_new_packet(struct sh_audio *sh)
 268 {
 269     struct priv *priv = sh->context;
 270     AVCodecContext *avctx = priv->avctx;
 271     double pts = MP_NOPTS_VALUE;
 272     int insize;
 273     bool packet_already_used = priv->previous_data_left;
 274     struct demux_packet *mpkt = ds_get_packet2(sh->ds,
 275                                                priv->previous_data_left);
 276     unsigned char *start;
 277     if (!mpkt) {
 278         assert(!priv->previous_data_left);
 279         start = NULL;
 280         insize = 0;
 281         ds_parse(sh->ds, &start, &insize, pts, 0);
 282         if (insize <= 0)
 283             return -1;  // error or EOF
 284     } else {
 285         assert(mpkt->len >= priv->previous_data_left);
 286         if (!priv->previous_data_left) {
 287             priv->previous_data_left = mpkt->len;
 288             pts = mpkt->pts;
 289         }
 290         insize = priv->previous_data_left;
 291         start = mpkt->buffer + mpkt->len - priv->previous_data_left;
 292         int consumed = ds_parse(sh->ds, &start, &insize, pts, 0);
 293         priv->previous_data_left -= consumed;
 294     }
 295
 296     AVPacket pkt;
 297     av_init_packet(&pkt);
 298     pkt.data = start;
 299     pkt.size = insize;
 300     if (mpkt && mpkt->avpacket) {
 301         pkt.side_data = mpkt->avpacket->side_data;
 302         pkt.side_data_elems = mpkt->avpacket->side_data_elems;
 303     }
 304     if (pts != MP_NOPTS_VALUE && !packet_already_used) {
 305         sh->pts = pts;
 306         sh->pts_bytes = 0;
 307     }
 308     int got_frame = 0;
 309     int ret = avcodec_decode_audio4(avctx, priv->avframe, &got_frame, &pkt);
 310     // LATM may need many packets to find mux info
 311     if (ret == AVERROR(EAGAIN))
 312         return 0;
 313     if (ret < 0) {
 314         mp_msg(MSGT_DECAUDIO, MSGL_V, "lavc_audio: error\n");
 315         return -1;
 316     }
 317     if (!sh->parser)
 318         priv->previous_data_left += insize - ret;
 319     if (!got_frame)
 320         return 0;
 321     /* An error is reported later from output format checking, but make
 322      * sure we don't crash by overreading first plane. */
 323     if (av_sample_fmt_is_planar(avctx->sample_fmt) && avctx->channels > 1)
 324         return 0;
 325     uint64_t unitsize = (uint64_t)av_get_bytes_per_sample(avctx->sample_fmt) *
 326                         avctx->channels;
 327     if (unitsize > 100000)
 328         abort();
 329     priv->unitsize = unitsize;
 330     uint64_t output_left = unitsize * priv->avframe->nb_samples;
 331     if (output_left > 500000000)
 332         abort();
 333     priv->output_left = output_left;
 334     priv->output = priv->avframe->data[0];
 335     mp_dbg(MSGT_DECAUDIO, MSGL_DBG2, "Decoded %d -> %d  \n", insize,
 336            priv->output_left);
 337     return 0;
 338 }
 339
 340
 341 static int decode_audio(sh_audio_t *sh_audio, unsigned char *buf, int minlen,
 342                         int maxlen)
 343 {
 344     struct priv *priv = sh_audio->context;
 345     AVCodecContext *avctx = priv->avctx;
 346
 347     int len = -1;
 348     while (len < minlen) {
 349         if (!priv->output_left) {
 350             if (decode_new_packet(sh_audio) < 0)
 351                 break;
 352             continue;
 353         }
 354         if (setup_format(sh_audio, avctx))
 355             return len;
 356         int size = (minlen - len + priv->unitsize - 1);
 357         size -= size % priv->unitsize;
 358         size = FFMIN(size, priv->output_left);
 359         if (size > maxlen)
 360             abort();
 361         memcpy(buf, priv->output, size);
 362         priv->output += size;
 363         priv->output_left -= size;
 364         if (avctx->channels >= 5) {
 365             int samplesize = av_get_bytes_per_sample(avctx->sample_fmt);
 366             reorder_channel_nch(buf, AF_CHANNEL_LAYOUT_LAVC_DEFAULT,
 367                                 AF_CHANNEL_LAYOUT_MPLAYER_DEFAULT,
 368                                 avctx->channels,
 369                                 size / samplesize, samplesize);
 370         }
 371         if (len < 0)
 372             len = size;
 373         else
 374             len += size;
 375         buf += size;
 376         maxlen -= size;
 377         sh_audio->pts_bytes += size;
 378     }
 379     return len;
 380 }