summaryrefslogtreecommitdiffstats
path: root/audio/decode/ad_lavc.c
blob: 8abc0a603504d34234948b202913321fd28c938d (plain)
1
2
3
4
5
6
7
8
9
10
11
12
13
14
15
16
17
18
19
20
21
22
23
24
25
26
27
28
29
30
31
32
33
34
35
36
37
38
39
40
41
42
43
44
45
46
47
48
49
50
51
52
53
54
55
56
57
58
59
60
61
62
63
64
65
66
67
68
69
70
71
72
73
74
75
76
77
78
79
80
81
82
83
84
85
86
87
88
89
90
91
92
93
94
95
96
97
98
99
100
101
102
103
104
105
106
107
108
109
110
111
112
113
114
115
116
117
118
119
120
121
122
123
124
125
126
127
128
129
130
131
132
133
134
135
136
137
138
139
140
141
142
143
144
145
146
147
148
149
150
151
152
153
154
155
156
157
158
159
160
161
162
163
164
165
166
167
168
169
170
171
172
173
174
175
176
177
178
179
180
181
182
183
184
185
186
187
188
189
190
191
192
193
194
195
196
197
198
199
200
201
202
203
204
205
206
207
208
209
210
211
212
213
214
215
216
217
218
219
220
221
222
223
224
225
226
227
228
229
230
231
232
233
234
235
236
237
238
239
240
241
242
243
244
245
246
247
248
249
250
251
252
253
254
255
256
257
258
259
260
261
262
263
264
265
266
267
268
269
270
271
272
273
274
275
276
277
278
279
280
281
282
283
284
285
286
287
288
289
290
291
292
293
294
295
296
297
298
299
300
301
302
303
304
305
306
307
308
309
310
311
312
313
314
315
316
317
318
319
320
321
322
323
324
325
326
327
328
329
330
331
332
333
334
335
336
337
338
339
340
341
342
343
344
345
346
347
348
349
350
351
352
353
354
355
356
357
358
359
360
361
362
363
364
365
366
367
368
369
370
371
372
373
374
375
376
377
378
379
380
381
382
383
384
385
386
387
388
389
390
391
392
393
394
395
396
397
398
399
400
401
402
403
404
405
406
407
408
409
410
411
412
413
414
415
416
417
418
419
420
421
422
423
424
425
426
427
428
429
430
431
432
433
434
435
436
437
438
439
440
441
442
443
444
445
446
447
448
449
450
451
452
453
454
455
456
457
458
459
460
461
462
463
464
465
466
467
468
469
470
471
472
473
474
475
476
477
478
479
480
481
482
483
484
485
486
487
488
489
/*
 * 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 <stdbool.h>
#include <assert.h>

#include <libavcodec/avcodec.h>
#include <libavutil/opt.h>

#include "talloc.h"

#include "config.h"
#include "core/av_common.h"
#include "core/codecs.h"
#include "core/mp_msg.h"
#include "core/options.h"
#include "core/av_opts.h"

#include "ad_internal.h"
#include "audio/reorder_ch.h"
#include "audio/fmt-conversion.h"

#include "compat/mpbswap.h"
#include "compat/libav.h"

LIBAD_EXTERN(lavc)

struct priv {
    AVCodecContext *avctx;
    AVFrame *avframe;
    uint8_t *output;
    uint8_t *output_packed; // used by deplanarize to store packed audio samples
    int output_left;
    int unitsize;
    int previous_data_left;  // input demuxer packet data
    bool force_channel_map;
};

#define OPT_BASE_STRUCT struct MPOpts

const m_option_t ad_lavc_decode_opts_conf[] = {
    OPT_FLOATRANGE("ac3drc", ad_lavc_param.ac3drc, 0, 0, 2),
    OPT_FLAG("downmix", ad_lavc_param.downmix, 0),
    OPT_STRING("o", ad_lavc_param.avopt, 0),
    {0}
};

struct pcm_map
{
    int tag;
    const char *codecs[5]; // {any, 1byte, 2bytes, 3bytes, 4bytes}
};

// NOTE: some of these are needed to make rawaudio with demux_mkv and others
//       work. ffmpeg does similar mapping internally, not part of the public
//       API. Some of these might be dead leftovers for demux_mov support.
static const struct pcm_map tag_map[] = {
    // Microsoft PCM
    {0x0,           {NULL, "pcm_u8", "pcm_s16le", "pcm_s24le", "pcm_s32le"}},
    {0x1,           {NULL, "pcm_u8", "pcm_s16le", "pcm_s24le", "pcm_s32le"}},
    // MS PCM, Extended
    {0xfffe,        {NULL, "pcm_u8", "pcm_s16le", "pcm_s24le", "pcm_s32le"}},
    // IEEE float
    {0x3,           {"pcm_f32le"}},
    // 'raw '
    {0x20776172,    {"pcm_s16be", [1] = "pcm_u8"}},
    // 'twos'/'sowt'
    {0x736F7774,    {"pcm_s16be", [1] = "pcm_s8"}},
    {0x74776F73,    {"pcm_s16be", [1] = "pcm_s8"}},
    // 'fl32'/'FL32'
    {0x32336c66,    {"pcm_f32be"}},
    {0x32334C46,    {"pcm_f32be"}},
    // '23lf'/'lpcm'
    {0x666c3332,    {"pcm_f32le"}},
    {0x6D63706C,    {"pcm_f32le"}},
    // 'in24', bigendian int24
    {0x34326e69,    {"pcm_s24be"}},
    // '42ni', little endian int24, MPlayer internal fourCC
    {0x696e3234,    {"pcm_s24le"}},
    // 'in32', bigendian int32
    {0x32336e69,    {"pcm_s32be"}},
    // '23ni', little endian int32, MPlayer internal fourCC
    {0x696e3332,    {"pcm_s32le"}},
    {-1},
};

// For demux_rawaudio.c; needed because ffmpeg doesn't have these sample
// formats natively.
static const struct pcm_map af_map[] = {
    {AF_FORMAT_U8,              {"pcm_u8"}},
    {AF_FORMAT_S8,              {"pcm_u8"}},
    {AF_FORMAT_U16_LE,          {"pcm_u16le"}},
    {AF_FORMAT_U16_BE,          {"pcm_u16be"}},
    {AF_FORMAT_S16_LE,          {"pcm_s16le"}},
    {AF_FORMAT_S16_BE,          {"pcm_s16be"}},
    {AF_FORMAT_U24_LE,          {"pcm_u24le"}},
    {AF_FORMAT_U24_BE,          {"pcm_u24be"}},
    {AF_FORMAT_S24_LE,          {"pcm_s24le"}},
    {AF_FORMAT_S24_BE,          {"pcm_s24be"}},
    {AF_FORMAT_U32_LE,          {"pcm_u32le"}},
    {AF_FORMAT_U32_BE,          {"pcm_u32be"}},
    {AF_FORMAT_S32_LE,          {"pcm_s32le"}},
    {AF_FORMAT_S32_BE,          {"pcm_s32be"}},
    {AF_FORMAT_FLOAT_LE,        {"pcm_f32le"}},
    {AF_FORMAT_FLOAT_BE,        {"pcm_f32be"}},
    {-1},
};

static const char *find_pcm_decoder(const struct pcm_map *map, int format,
                                    int bits_per_sample)
{
    int bytes = (bits_per_sample + 7) / 8;
    for (int n = 0; map[n].tag != -1; n++) {
        const struct pcm_map *entry = &map[n];
        if (entry->tag == format) {
            const char *dec = NULL;
            if (bytes >= 1 && bytes <= 4)
                dec = entry->codecs[bytes];
            if (!dec)
                dec = entry->codecs[0];
            if (dec)
                return dec;
        }
    }
    return NULL;
}

static int preinit(sh_audio_t *sh)
{
    return 1;
}

/* Prefer playing audio with the samplerate given in container data
 * if available, but take number the number of channels and sample format
 * from the codec, since if the codec isn't using the correct values for
 * those everything breaks anyway.
 */
static int setup_format(sh_audio_t *sh_audio,
                        const AVCodecContext *lavc_context)
{
    struct priv *priv = sh_audio->context;
    int sample_format        =
        af_from_avformat(av_get_packed_sample_fmt(lavc_context->sample_fmt));
    bool broken_srate        = false;
    int samplerate           = lavc_context->sample_rate;
    int container_samplerate = sh_audio->container_out_samplerate;
    if (!container_samplerate && sh_audio->wf)
        container_samplerate = sh_audio->wf->nSamplesPerSec;
    if (lavc_context->codec_id == AV_CODEC_ID_AAC
        && samplerate == 2 * container_samplerate)
        broken_srate = true;
    else if (container_samplerate)
        samplerate = container_samplerate;

    struct mp_chmap lavc_chmap;
    mp_chmap_from_lavc(&lavc_chmap, lavc_context->channel_layout);
    // No channel layout or layout disagrees with channel count
    if (lavc_chmap.num != lavc_context->channels)
        mp_chmap_from_channels(&lavc_chmap, lavc_context->channels);
    if (priv->force_channel_map) {
        if (lavc_chmap.num == sh_audio->channels.num)
            lavc_chmap = sh_audio->channels;
    }

    if (!mp_chmap_equals(&lavc_chmap, &sh_audio->channels) ||
        samplerate != sh_audio->samplerate ||
        sample_format != sh_audio->sample_format) {
        sh_audio->channels = lavc_chmap;
        sh_audio->samplerate = samplerate;
        sh_audio->sample_format = sample_format;
        sh_audio->samplesize = af_fmt2bits(sh_audio->sample_format) / 8;
        if (broken_srate)
            mp_msg(MSGT_DECAUDIO, MSGL_WARN,
                   "Ignoring broken container sample rate for AAC with SBR\n");
        return 1;
    }
    return 0;
}

static void set_from_wf(AVCodecContext *avctx, WAVEFORMATEX *wf)
{
    avctx->channels = wf->nChannels;
    avctx->sample_rate = wf->nSamplesPerSec;
    avctx->bit_rate = wf->nAvgBytesPerSec * 8;
    avctx->block_align = wf->nBlockAlign;
    avctx->bits_per_coded_sample = wf->wBitsPerSample;

    if (wf->cbSize > 0) {
        avctx->extradata = av_mallocz(wf->cbSize + FF_INPUT_BUFFER_PADDING_SIZE);
        avctx->extradata_size = wf->cbSize;
        memcpy(avctx->extradata, wf + 1, avctx->extradata_size);
    }
}

static int init(sh_audio_t *sh_audio, const char *decoder)
{
    struct MPOpts *mpopts = sh_audio->opts;
    struct ad_lavc_param *opts = &mpopts->ad_lavc_param;
    AVCodecContext *lavc_context;
    AVCodec *lavc_codec;

    struct priv *ctx = talloc_zero(NULL, struct priv);
    sh_audio->context = ctx;

    if (sh_audio->wf && strcmp(decoder, "pcm") == 0) {
        decoder = find_pcm_decoder(tag_map, sh_audio->format,
                                   sh_audio->wf->wBitsPerSample);
    } else if (sh_audio->wf && strcmp(decoder, "mp-pcm") == 0) {
        decoder = find_pcm_decoder(af_map, sh_audio->format, 0);
        ctx->force_channel_map = true;
    }

    lavc_codec = avcodec_find_decoder_by_name(decoder);
    if (!lavc_codec) {
        mp_tmsg(MSGT_DECAUDIO, MSGL_ERR,
                "Cannot find codec '%s' in libavcodec...\n", decoder);
        uninit(sh_audio);
        return 0;
    }

    lavc_context = avcodec_alloc_context3(lavc_codec);
    ctx->avctx = lavc_context;
    ctx->avframe = avcodec_alloc_frame();
    lavc_context->codec_type = AVMEDIA_TYPE_AUDIO;
    lavc_context->codec_id = lavc_codec->id;

    if (opts->downmix) {
        lavc_context->request_channels = mpopts->audio_output_channels.num;
        lavc_context->request_channel_layout =
            mp_chmap_to_lavc(&mpopts->audio_output_channels);
    }

    // Always try to set - option only exists for AC3 at the moment
    av_opt_set_double(lavc_context, "drc_scale", opts->ac3drc,
                      AV_OPT_SEARCH_CHILDREN);

    if (opts->avopt) {
        if (parse_avopts(lavc_context, opts->avopt) < 0) {
            mp_msg(MSGT_DECVIDEO, MSGL_ERR,
                   "ad_lavc: setting AVOptions '%s' failed.\n", opts->avopt);
            uninit(sh_audio);
            return 0;
        }
    }

    lavc_context->codec_tag = sh_audio->format;
    lavc_context->sample_rate = sh_audio->samplerate;
    lavc_context->bit_rate = sh_audio->i_bps * 8;
    lavc_context->channel_layout = mp_chmap_to_lavc(&sh_audio->channels);

    if (sh_audio->wf)
        set_from_wf(lavc_context, sh_audio->wf);

    // demux_mkv, demux_mpg
    if (sh_audio->codecdata_len && sh_audio->codecdata &&
            !lavc_context->extradata) {
        lavc_context->extradata = av_malloc(sh_audio->codecdata_len +
                                            FF_INPUT_BUFFER_PADDING_SIZE);
        lavc_context->extradata_size = sh_audio->codecdata_len;
        memcpy(lavc_context->extradata, (char *)sh_audio->codecdata,
               lavc_context->extradata_size);
    }

    if (sh_audio->gsh->lav_headers)
        mp_copy_lav_codec_headers(lavc_context, sh_audio->gsh->lav_headers);

    /* open it */
    if (avcodec_open2(lavc_context, lavc_codec, NULL) < 0) {
        mp_tmsg(MSGT_DECAUDIO, MSGL_ERR, "Could not open codec.\n");
        uninit(sh_audio);
        return 0;
    }
    mp_msg(MSGT_DECAUDIO, MSGL_V, "INFO: libavcodec \"%s\" init OK!\n",
           lavc_codec->name);

    // Decode at least 1 byte:  (to get header filled)
    for (int tries = 0;;) {
        int x = decode_audio(sh_audio, sh_audio->a_buffer, 1,
                             sh_audio->a_buffer_size);
        if (x > 0) {
            sh_audio->a_buffer_len = x;
            break;
        }
        if (++tries >= 5) {
            mp_msg(MSGT_DECAUDIO, MSGL_ERR,
                   "ad_lavc: initial decode failed\n");
            uninit(sh_audio);
            return 0;
        }
    }

    sh_audio->i_bps = lavc_context->bit_rate / 8;
    if (sh_audio->wf && sh_audio->wf->nAvgBytesPerSec)
        sh_audio->i_bps = sh_audio->wf->nAvgBytesPerSec;

    int af_sample_fmt =
        af_from_avformat(av_get_packed_sample_fmt(lavc_context->sample_fmt));
    if (af_sample_fmt == AF_FORMAT_UNKNOWN) {
        uninit(sh_audio);
        return 0;
    }
    return 1;
}

static void uninit(sh_audio_t *sh)
{
    struct priv *ctx = sh->context;
    if (!ctx)
        return;
    AVCodecContext *lavc_context = ctx->avctx;

    if (lavc_context) {
        if (avcodec_close(lavc_context) < 0)
            mp_tmsg(MSGT_DECVIDEO, MSGL_ERR, "Could not close codec.\n");
        av_freep(&lavc_context->extradata);
        av_freep(&lavc_context);
    }
    avcodec_free_frame(&ctx->avframe);
    talloc_free(ctx);
    sh->context = NULL;
}

static int control(sh_audio_t *sh, int cmd, void *arg)
{
    struct priv *ctx = sh->context;
    switch (cmd) {
    case ADCTRL_RESYNC_STREAM:
        avcodec_flush_buffers(ctx->avctx);
        ds_clear_parser(sh->ds);
        ctx->previous_data_left = 0;
        ctx->output_left = 0;
        return CONTROL_TRUE;
    }
    return CONTROL_UNKNOWN;
}

static av_always_inline void deplanarize(struct sh_audio *sh)
{
    struct priv *priv = sh->context;

    uint8_t **planes  = priv->avframe->extended_data;
    size_t bps        = av_get_bytes_per_sample(priv->avctx->sample_fmt);
    size_t nb_samples = priv->avframe->nb_samples;
    size_t channels   = priv->avctx->channels;
    size_t size       = bps * nb_samples * channels;

    if (talloc_get_size(priv->output_packed) != size)
        priv->output_packed =
            talloc_realloc_size(priv, priv->output_packed, size);

    reorder_to_packed(priv->output_packed, planes, bps, channels, nb_samples);

    priv->output = priv->output_packed;
}

static int decode_new_packet(struct sh_audio *sh)
{
    struct priv *priv = sh->context;
    AVCodecContext *avctx = priv->avctx;
    double pts = MP_NOPTS_VALUE;
    int insize;
    bool packet_already_used = priv->previous_data_left;
    struct demux_packet *mpkt = ds_get_packet2(sh->ds,
                                               priv->previous_data_left);
    unsigned char *start;
    if (!mpkt) {
        assert(!priv->previous_data_left);
        start = NULL;
        insize = 0;
        ds_parse(sh->ds, &start, &insize, pts, 0);
        if (insize <= 0)
            return -1;  // error or EOF
    } else {
        assert(mpkt->len >= priv->previous_data_left);
        if (!priv->previous_data_left) {
            priv->previous_data_left = mpkt->len;
            pts = mpkt->pts;
        }
        insize = priv->previous_data_left;
        start = mpkt->buffer + mpkt->len - priv->previous_data_left;
        int consumed = ds_parse(sh->ds, &start, &insize, pts, 0);
        priv->previous_data_left -= consumed;
        priv->previous_data_left = FFMAX(priv->previous_data_left, 0);
    }

    AVPacket pkt;
    av_init_packet(&pkt);
    pkt.data = start;
    pkt.size = insize;
    if (mpkt && mpkt->avpacket) {
        pkt.side_data = mpkt->avpacket->side_data;
        pkt.side_data_elems = mpkt->avpacket->side_data_elems;
    }
    if (pts != MP_NOPTS_VALUE && !packet_already_used) {
        sh->pts = pts;
        sh->pts_bytes = 0;
    }
    int got_frame = 0;
    int ret = avcodec_decode_audio4(avctx, priv->avframe, &got_frame, &pkt);
    // LATM may need many packets to find mux info
    if (ret == AVERROR(EAGAIN))
        return 0;
    if (ret < 0) {
        mp_msg(MSGT_DECAUDIO, MSGL_V, "lavc_audio: error\n");
        return -1;
    }
    // The "insize >= ret" test is sanity check against decoder overreads
    if (!sh->parser && insize >= ret)
        priv->previous_data_left = insize - ret;
    if (!got_frame)
        return 0;
    uint64_t unitsize = (uint64_t)av_get_bytes_per_sample(avctx->sample_fmt) *
                        avctx->channels;
    if (unitsize > 100000)
        abort();
    priv->unitsize = unitsize;
    uint64_t output_left = unitsize * priv->avframe->nb_samples;
    if (output_left > 500000000)
        abort();
    priv->output_left = output_left;
    if (av_sample_fmt_is_planar(avctx->sample_fmt) && avctx->channels > 1) {
        deplanarize(sh);
    } else {
        priv->output = priv->avframe->data[0];
    }
    mp_dbg(MSGT_DECAUDIO, MSGL_DBG2, "Decoded %d -> %d  \n", insize,
           priv->output_left);
    return 0;
}


static int decode_audio(sh_audio_t *sh_audio, unsigned char *buf, int minlen,
                        int maxlen)
{
    struct priv *priv = sh_audio->context;
    AVCodecContext *avctx = priv->avctx;

    int len = -1;
    while (len < minlen) {
        if (!priv->output_left) {
            if (decode_new_packet(sh_audio) < 0)
                break;
            continue;
        }
        if (setup_format(sh_audio, avctx))
            return len;
        int size = (minlen - len + priv->unitsize - 1);
        size -= size % priv->unitsize;
        size = FFMIN(size, priv->output_left);
        if (size > maxlen)
            abort();
        memcpy(buf, priv->output, size);
        priv->output += size;
        priv->output_left -= size;
        if (len < 0)
            len = size;
        else
            len += size;
        buf += size;
        maxlen -= size;
        sh_audio->pts_bytes += size;
    }
    return len;
}

static void add_decoders(struct mp_decoder_list *list)
{
    mp_add_lavc_decoders(list, AVMEDIA_TYPE_AUDIO);
    mp_add_decoder(list, "lavc", "pcm", "pcm", "Raw PCM");
    mp_add_decoder(list, "lavc", "mp-pcm", "mp-pcm", "Raw PCM");
}