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     "FFmpeg/libavcodec audio decoders",
  42     "ffmpeg",
  43     "Nick Kurshev",
  44     "ffmpeg.sf.net",
  45     ""
  46 };
  47
  48 LIBAD_EXTERN(ffmpeg)
  49
  50 struct priv {
  51     AVCodecContext *avctx;
  52     int previous_data_left;
  53 };
  54
  55 static int preinit(sh_audio_t *sh)
  56 {
  57     sh->audio_out_minsize = AVCODEC_MAX_AUDIO_FRAME_SIZE;
  58     return 1;
  59 }
  60
  61 /* Prefer playing audio with the samplerate given in container data
  62  * if available, but take number the number of channels and sample format
  63  * from the codec, since if the codec isn't using the correct values for
  64  * those everything breaks anyway.
  65  */
  66 static int setup_format(sh_audio_t *sh_audio,
  67                         const AVCodecContext *lavc_context)
  68 {
  69     int sample_format = sh_audio->sample_format;
  70     switch (lavc_context->sample_fmt) {
  71     case AV_SAMPLE_FMT_U8:  sample_format = AF_FORMAT_U8;       break;
  72     case AV_SAMPLE_FMT_S16: sample_format = AF_FORMAT_S16_NE;   break;
  73     case AV_SAMPLE_FMT_S32: sample_format = AF_FORMAT_S32_NE;   break;
  74     case AV_SAMPLE_FMT_FLT: sample_format = AF_FORMAT_FLOAT_NE; break;
  75     default:
  76         mp_msg(MSGT_DECAUDIO, MSGL_FATAL, "Unsupported sample format\n");
  77     }
  78
  79     bool broken_srate        = false;
  80     int samplerate           = lavc_context->sample_rate;
  81     int container_samplerate = sh_audio->container_out_samplerate;
  82     if (!container_samplerate && sh_audio->wf)
  83         container_samplerate = sh_audio->wf->nSamplesPerSec;
  84     if (lavc_context->codec_id == CODEC_ID_AAC
  85         && samplerate == 2 * container_samplerate)
  86         broken_srate = true;
  87     else if (container_samplerate)
  88         samplerate = container_samplerate;
  89
  90     if (lavc_context->channels != sh_audio->channels ||
  91         samplerate != sh_audio->samplerate ||
  92         sample_format != sh_audio->sample_format) {
  93         sh_audio->channels = lavc_context->channels;
  94         sh_audio->samplerate = samplerate;
  95         sh_audio->sample_format = sample_format;
  96         sh_audio->samplesize = af_fmt2bits(sh_audio->sample_format) / 8;
  97         if (broken_srate)
  98             mp_msg(MSGT_DECAUDIO, MSGL_WARN,
  99                    "Ignoring broken container sample rate for AAC with SBR\n");
 100         return 1;
 101     }
 102     return 0;
 103 }
 104
 105 static int init(sh_audio_t *sh_audio)
 106 {
 107     struct MPOpts *opts = sh_audio->opts;
 108     AVCodecContext *lavc_context;
 109     AVCodec *lavc_codec;
 110
 111     mp_msg(MSGT_DECAUDIO, MSGL_V, "FFmpeg's libavcodec audio codec\n");
 112
 113     lavc_codec = avcodec_find_decoder_by_name(sh_audio->codec->dll);
 114     if (!lavc_codec) {
 115         mp_tmsg(MSGT_DECAUDIO, MSGL_ERR,
 116                 "Cannot find codec '%s' in libavcodec...\n",
 117                 sh_audio->codec->dll);
 118         return 0;
 119     }
 120
 121     struct priv *ctx = talloc_zero(NULL, struct priv);
 122     sh_audio->context = ctx;
 123     lavc_context = avcodec_alloc_context3(lavc_codec);
 124     ctx->avctx = lavc_context;
 125
 126     // Always try to set - option only exists for AC3 at the moment
 127     av_opt_set_double(lavc_context, "drc_scale", opts->drc_level,
 128                       AV_OPT_SEARCH_CHILDREN);
 129     lavc_context->sample_rate = sh_audio->samplerate;
 130     lavc_context->bit_rate = sh_audio->i_bps * 8;
 131     if (sh_audio->wf) {
 132         lavc_context->channels = sh_audio->wf->nChannels;
 133         lavc_context->sample_rate = sh_audio->wf->nSamplesPerSec;
 134         lavc_context->bit_rate = sh_audio->wf->nAvgBytesPerSec * 8;
 135         lavc_context->block_align = sh_audio->wf->nBlockAlign;
 136         lavc_context->bits_per_coded_sample = sh_audio->wf->wBitsPerSample;
 137     }
 138     lavc_context->request_channels = opts->audio_output_channels;
 139     lavc_context->codec_tag = sh_audio->format; //FOURCC
 140     lavc_context->codec_type = AVMEDIA_TYPE_AUDIO;
 141     lavc_context->codec_id = lavc_codec->id; // not sure if required, imho not --A'rpi
 142
 143     /* alloc extra data */
 144     if (sh_audio->wf && sh_audio->wf->cbSize > 0) {
 145         lavc_context->extradata = av_mallocz(sh_audio->wf->cbSize + FF_INPUT_BUFFER_PADDING_SIZE);
 146         lavc_context->extradata_size = sh_audio->wf->cbSize;
 147         memcpy(lavc_context->extradata, sh_audio->wf + 1,
 148                lavc_context->extradata_size);
 149     }
 150
 151     // for QDM2
 152     if (sh_audio->codecdata_len && sh_audio->codecdata &&
 153             !lavc_context->extradata) {
 154         lavc_context->extradata = av_malloc(sh_audio->codecdata_len +
 155                                             FF_INPUT_BUFFER_PADDING_SIZE);
 156         lavc_context->extradata_size = sh_audio->codecdata_len;
 157         memcpy(lavc_context->extradata, (char *)sh_audio->codecdata,
 158                lavc_context->extradata_size);
 159     }
 160
 161     /* open it */
 162     if (avcodec_open2(lavc_context, lavc_codec, NULL) < 0) {
 163         mp_tmsg(MSGT_DECAUDIO, MSGL_ERR, "Could not open codec.\n");
 164         uninit(sh_audio);
 165         return 0;
 166     }
 167     mp_msg(MSGT_DECAUDIO, MSGL_V, "INFO: libavcodec \"%s\" init OK!\n",
 168            lavc_codec->name);
 169
 170     if (sh_audio->format == 0x3343414D) {
 171         // MACE 3:1
 172         sh_audio->ds->ss_div = 2 * 3; // 1 samples/packet
 173         sh_audio->ds->ss_mul = 2 * sh_audio->wf->nChannels; // 1 byte*ch/packet
 174     } else if (sh_audio->format == 0x3643414D) {
 175         // MACE 6:1
 176         sh_audio->ds->ss_div = 2 * 6; // 1 samples/packet
 177         sh_audio->ds->ss_mul = 2 * sh_audio->wf->nChannels; // 1 byte*ch/packet
 178     }
 179
 180     // Decode at least 1 byte:  (to get header filled)
 181     for (int tries = 0;;) {
 182         int x = decode_audio(sh_audio, sh_audio->a_buffer, 1,
 183                              sh_audio->a_buffer_size);
 184         if (x > 0) {
 185             sh_audio->a_buffer_len = x;
 186             break;
 187         }
 188         if (++tries >= 5) {
 189             mp_msg(MSGT_DECAUDIO, MSGL_ERR,
 190                    "ad_ffmpeg: initial decode failed\n");
 191             uninit(sh_audio);
 192             return 0;
 193         }
 194     }
 195
 196     sh_audio->i_bps = lavc_context->bit_rate / 8;
 197     if (sh_audio->wf && sh_audio->wf->nAvgBytesPerSec)
 198         sh_audio->i_bps = sh_audio->wf->nAvgBytesPerSec;
 199
 200     switch (lavc_context->sample_fmt) {
 201     case AV_SAMPLE_FMT_U8:
 202     case AV_SAMPLE_FMT_S16:
 203     case AV_SAMPLE_FMT_S32:
 204     case AV_SAMPLE_FMT_FLT:
 205         break;
 206     default:
 207         uninit(sh_audio);
 208         return 0;
 209     }
 210     return 1;
 211 }
 212
 213 static void uninit(sh_audio_t *sh)
 214 {
 215     struct priv *ctx = sh->context;
 216     if (!ctx)
 217         return;
 218     AVCodecContext *lavc_context = ctx->avctx;
 219
 220     if (lavc_context) {
 221         if (avcodec_close(lavc_context) < 0)
 222             mp_tmsg(MSGT_DECVIDEO, MSGL_ERR, "Could not close codec.\n");
 223         av_freep(&lavc_context->extradata);
 224         av_freep(&lavc_context);
 225     }
 226     talloc_free(ctx);
 227     sh->context = NULL;
 228 }
 229
 230 static int control(sh_audio_t *sh, int cmd, void *arg, ...)
 231 {
 232     struct priv *ctx = sh->context;
 233     switch (cmd) {
 234     case ADCTRL_RESYNC_STREAM:
 235         avcodec_flush_buffers(ctx->avctx);
 236         ds_clear_parser(sh->ds);
 237         ctx->previous_data_left = 0;
 238         return CONTROL_TRUE;
 239     }
 240     return CONTROL_UNKNOWN;
 241 }
 242
 243 static int decode_audio(sh_audio_t *sh_audio, unsigned char *buf, int minlen,
 244                         int maxlen)
 245 {
 246     struct priv *ctx = sh_audio->context;
 247     AVCodecContext *avctx = ctx->avctx;
 248
 249     unsigned char *start = NULL;
 250     int y, len = -1;
 251     while (len < minlen) {
 252         AVPacket pkt;
 253         int len2 = maxlen;
 254         double pts = MP_NOPTS_VALUE;
 255         int x;
 256         bool packet_already_used = ctx->previous_data_left;
 257         struct demux_packet *mpkt = ds_get_packet2(sh_audio->ds,
 258                                                    ctx->previous_data_left);
 259         if (!mpkt) {
 260             assert(!ctx->previous_data_left);
 261             start = NULL;
 262             x = 0;
 263             ds_parse(sh_audio->ds, &start, &x, pts, 0);
 264             if (x <= 0)
 265                 break;  // error
 266         } else {
 267             assert(mpkt->len >= ctx->previous_data_left);
 268             if (!ctx->previous_data_left) {
 269                 ctx->previous_data_left = mpkt->len;
 270                 pts = mpkt->pts;
 271             }
 272             x = ctx->previous_data_left;
 273             start = mpkt->buffer + mpkt->len - ctx->previous_data_left;
 274             int consumed = ds_parse(sh_audio->ds, &start, &x, pts, 0);
 275             ctx->previous_data_left -= consumed;
 276         }
 277         av_init_packet(&pkt);
 278         pkt.data = start;
 279         pkt.size = x;
 280         if (mpkt && mpkt->avpacket) {
 281             pkt.side_data = mpkt->avpacket->side_data;
 282             pkt.side_data_elems = mpkt->avpacket->side_data_elems;
 283         }
 284         if (pts != MP_NOPTS_VALUE && !packet_already_used) {
 285             sh_audio->pts = pts;
 286             sh_audio->pts_bytes = 0;
 287         }
 288         y = avcodec_decode_audio3(avctx, (int16_t *)buf, &len2, &pkt);
 289         // LATM may need many packets to find mux info
 290         if (y == AVERROR(EAGAIN))
 291             continue;
 292         if (y < 0) {
 293             mp_msg(MSGT_DECAUDIO, MSGL_V, "lavc_audio: error\n");
 294             break;
 295         }
 296         if (!sh_audio->parser)
 297             ctx->previous_data_left += x - y;
 298         if (len2 > 0) {
 299             if (avctx->channels >= 5) {
 300                 int samplesize = av_get_bytes_per_sample(avctx->sample_fmt);
 301                 reorder_channel_nch(buf, AF_CHANNEL_LAYOUT_LAVC_DEFAULT,
 302                                     AF_CHANNEL_LAYOUT_MPLAYER_DEFAULT,
 303                                     avctx->channels,
 304                                     len2 / samplesize, samplesize);
 305             }
 306             if (len < 0)
 307                 len = len2;
 308             else
 309                 len += len2;
 310             buf += len2;
 311             maxlen -= len2;
 312             sh_audio->pts_bytes += len2;
 313         }
 314         mp_dbg(MSGT_DECAUDIO, MSGL_DBG2, "Decoded %d -> %d  \n", y, len2);
 315
 316         if (setup_format(sh_audio, avctx))
 317             break;
 318     }
 319     return len;
 320 }