aboutsummaryrefslogtreecommitdiffstats
path: root/libavcodec/audiotoolboxenc.c
blob: 9245aa9dc41b9a15953749975347492b5933641e (plain) (blame)
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
490
491
492
493
494
495
496
497
498
499
500
501
502
503
504
505
506
507
508
509
510
511
512
513
514
515
516
517
518
519
520
521
522
523
524
525
526
527
528
529
530
531
532
533
534
535
536
537
538
539
540
541
542
543
544
545
546
547
548
549
550
551
552
553
554
555
556
557
558
559
560
561
562
563
564
565
566
567
568
569
570
571
572
573
574
575
576
577
578
579
580
581
582
583
584
585
586
587
588
589
590
591
592
593
594
595
596
597
598
599
600
601
602
603
604
605
606
607
608
609
610
611
612
613
614
615
616
617
618
619
620
621
622
623
624
625
626
627
628
629
630
631
632
633
634
635
636
637
638
639
640
641
642
643
644
645
646
647
648
649
650
651
652
653
654
655
656
657
658
659
660
661
662
663
664
/*
 * Audio Toolbox system codecs
 *
 * copyright (c) 2016 rcombs
 *
 * This file is part of FFmpeg.
 *
 * FFmpeg is free software; you can redistribute it and/or
 * modify it under the terms of the GNU Lesser General Public
 * License as published by the Free Software Foundation; either
 * version 2.1 of the License, or (at your option) any later version.
 *
 * FFmpeg 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
 * Lesser General Public License for more details.
 *
 * You should have received a copy of the GNU Lesser General Public
 * License along with FFmpeg; if not, write to the Free Software
 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
 */

#include <AudioToolbox/AudioToolbox.h>

#define FF_BUFQUEUE_SIZE 256
#include "libavfilter/bufferqueue.h"

#include "config.h"
#include "audio_frame_queue.h"
#include "avcodec.h"
#include "bytestream.h"
#include "encode.h"
#include "internal.h"
#include "libavformat/isom.h"
#include "libavutil/avassert.h"
#include "libavutil/channel_layout.h"
#include "libavutil/opt.h"
#include "libavutil/log.h"

typedef struct ATDecodeContext {
    AVClass *av_class;
    int mode;
    int quality;

    AudioConverterRef converter;
    struct FFBufQueue frame_queue;
    struct FFBufQueue used_frame_queue;

    unsigned pkt_size;
    AudioFrameQueue afq;
    int eof;
    int frame_size;

    AVFrame* encoding_frame;
} ATDecodeContext;

static UInt32 ffat_get_format_id(enum AVCodecID codec, int profile)
{
    switch (codec) {
    case AV_CODEC_ID_AAC:
        switch (profile) {
        case FF_PROFILE_AAC_LOW:
        default:
            return kAudioFormatMPEG4AAC;
        case FF_PROFILE_AAC_HE:
            return kAudioFormatMPEG4AAC_HE;
        case FF_PROFILE_AAC_HE_V2:
            return kAudioFormatMPEG4AAC_HE_V2;
        case FF_PROFILE_AAC_LD:
            return kAudioFormatMPEG4AAC_LD;
        case FF_PROFILE_AAC_ELD:
            return kAudioFormatMPEG4AAC_ELD;
        }
    case AV_CODEC_ID_ADPCM_IMA_QT:
        return kAudioFormatAppleIMA4;
    case AV_CODEC_ID_ALAC:
        return kAudioFormatAppleLossless;
    case AV_CODEC_ID_ILBC:
        return kAudioFormatiLBC;
    case AV_CODEC_ID_PCM_ALAW:
        return kAudioFormatALaw;
    case AV_CODEC_ID_PCM_MULAW:
        return kAudioFormatULaw;
    default:
        av_assert0(!"Invalid codec ID!");
        return 0;
    }
}

static void ffat_update_ctx(AVCodecContext *avctx)
{
    ATDecodeContext *at = avctx->priv_data;
    UInt32 size = sizeof(unsigned);
    AudioConverterPrimeInfo prime_info;
    AudioStreamBasicDescription out_format;

    AudioConverterGetProperty(at->converter,
                              kAudioConverterPropertyMaximumOutputPacketSize,
                              &size, &at->pkt_size);

    if (at->pkt_size <= 0)
        at->pkt_size = 1024 * 50;

    size = sizeof(prime_info);

    if (!AudioConverterGetProperty(at->converter,
                                   kAudioConverterPrimeInfo,
                                   &size, &prime_info)) {
        avctx->initial_padding = prime_info.leadingFrames;
    }

    size = sizeof(out_format);
    if (!AudioConverterGetProperty(at->converter,
                                   kAudioConverterCurrentOutputStreamDescription,
                                   &size, &out_format)) {
        if (out_format.mFramesPerPacket)
            avctx->frame_size = out_format.mFramesPerPacket;
        if (out_format.mBytesPerPacket && avctx->codec_id == AV_CODEC_ID_ILBC)
            avctx->block_align = out_format.mBytesPerPacket;
    }

    at->frame_size = avctx->frame_size;
    if (avctx->codec_id == AV_CODEC_ID_PCM_MULAW ||
        avctx->codec_id == AV_CODEC_ID_PCM_ALAW) {
        at->pkt_size *= 1024;
        avctx->frame_size *= 1024;
    }
}

static int read_descr(GetByteContext *gb, int *tag)
{
    int len = 0;
    int count = 4;
    *tag = bytestream2_get_byte(gb);
    while (count--) {
        int c = bytestream2_get_byte(gb);
        len = (len << 7) | (c & 0x7f);
        if (!(c & 0x80))
            break;
    }
    return len;
}

static int get_ilbc_mode(AVCodecContext *avctx)
{
    if (avctx->block_align == 38)
        return 20;
    else if (avctx->block_align == 50)
        return 30;
    else if (avctx->bit_rate > 0)
        return avctx->bit_rate <= 14000 ? 30 : 20;
    else
        return 30;
}

static av_cold int get_channel_label(int channel)
{
    uint64_t map = 1 << channel;
    if (map <= AV_CH_LOW_FREQUENCY)
        return channel + 1;
    else if (map <= AV_CH_BACK_RIGHT)
        return channel + 29;
    else if (map <= AV_CH_BACK_CENTER)
        return channel - 1;
    else if (map <= AV_CH_SIDE_RIGHT)
        return channel - 4;
    else if (map <= AV_CH_TOP_BACK_RIGHT)
        return channel + 1;
    else if (map <= AV_CH_STEREO_RIGHT)
        return -1;
    else if (map <= AV_CH_WIDE_RIGHT)
        return channel + 4;
    else if (map <= AV_CH_SURROUND_DIRECT_RIGHT)
        return channel - 23;
    else if (map == AV_CH_LOW_FREQUENCY_2)
        return kAudioChannelLabel_LFE2;
    else
        return -1;
}

static int remap_layout(AudioChannelLayout *layout, uint64_t in_layout, int count)
{
    int i;
    int c = 0;
    layout->mChannelLayoutTag = kAudioChannelLayoutTag_UseChannelDescriptions;
    layout->mNumberChannelDescriptions = count;
    for (i = 0; i < count; i++) {
        int label;
        while (!(in_layout & (1 << c)) && c < 64)
            c++;
        if (c == 64)
            return AVERROR(EINVAL); // This should never happen
        label = get_channel_label(c);
        layout->mChannelDescriptions[i].mChannelLabel = label;
        if (label < 0)
            return AVERROR(EINVAL);
        c++;
    }
    return 0;
}

static int get_aac_tag(uint64_t in_layout)
{
    switch (in_layout) {
    case AV_CH_LAYOUT_MONO:
        return kAudioChannelLayoutTag_Mono;
    case AV_CH_LAYOUT_STEREO:
        return kAudioChannelLayoutTag_Stereo;
    case AV_CH_LAYOUT_QUAD:
        return kAudioChannelLayoutTag_AAC_Quadraphonic;
    case AV_CH_LAYOUT_OCTAGONAL:
        return kAudioChannelLayoutTag_AAC_Octagonal;
    case AV_CH_LAYOUT_SURROUND:
        return kAudioChannelLayoutTag_AAC_3_0;
    case AV_CH_LAYOUT_4POINT0:
        return kAudioChannelLayoutTag_AAC_4_0;
    case AV_CH_LAYOUT_5POINT0:
        return kAudioChannelLayoutTag_AAC_5_0;
    case AV_CH_LAYOUT_5POINT1:
        return kAudioChannelLayoutTag_AAC_5_1;
    case AV_CH_LAYOUT_6POINT0:
        return kAudioChannelLayoutTag_AAC_6_0;
    case AV_CH_LAYOUT_6POINT1:
        return kAudioChannelLayoutTag_AAC_6_1;
    case AV_CH_LAYOUT_7POINT0:
        return kAudioChannelLayoutTag_AAC_7_0;
    case AV_CH_LAYOUT_7POINT1_WIDE_BACK:
        return kAudioChannelLayoutTag_AAC_7_1;
    case AV_CH_LAYOUT_7POINT1:
        return kAudioChannelLayoutTag_MPEG_7_1_C;
    default:
        return 0;
    }
}

static av_cold int ffat_init_encoder(AVCodecContext *avctx)
{
    ATDecodeContext *at = avctx->priv_data;
    OSStatus status;

    AudioStreamBasicDescription in_format = {
        .mSampleRate = avctx->sample_rate,
        .mFormatID = kAudioFormatLinearPCM,
        .mFormatFlags = ((avctx->sample_fmt == AV_SAMPLE_FMT_FLT ||
                          avctx->sample_fmt == AV_SAMPLE_FMT_DBL) ? kAudioFormatFlagIsFloat
                        : avctx->sample_fmt == AV_SAMPLE_FMT_U8 ? 0
                        : kAudioFormatFlagIsSignedInteger)
                        | kAudioFormatFlagIsPacked,
        .mBytesPerPacket = av_get_bytes_per_sample(avctx->sample_fmt) * avctx->channels,
        .mFramesPerPacket = 1,
        .mBytesPerFrame = av_get_bytes_per_sample(avctx->sample_fmt) * avctx->channels,
        .mChannelsPerFrame = avctx->channels,
        .mBitsPerChannel = av_get_bytes_per_sample(avctx->sample_fmt) * 8,
    };
    AudioStreamBasicDescription out_format = {
        .mSampleRate = avctx->sample_rate,
        .mFormatID = ffat_get_format_id(avctx->codec_id, avctx->profile),
        .mChannelsPerFrame = in_format.mChannelsPerFrame,
    };
    UInt32 layout_size = sizeof(AudioChannelLayout) +
                         sizeof(AudioChannelDescription) * avctx->channels;
    AudioChannelLayout *channel_layout = av_malloc(layout_size);

    if (!channel_layout)
        return AVERROR(ENOMEM);

    if (avctx->codec_id == AV_CODEC_ID_ILBC) {
        int mode = get_ilbc_mode(avctx);
        out_format.mFramesPerPacket  = 8000 * mode / 1000;
        out_format.mBytesPerPacket   = (mode == 20 ? 38 : 50);
    }

    status = AudioConverterNew(&in_format, &out_format, &at->converter);

    if (status != 0) {
        av_log(avctx, AV_LOG_ERROR, "AudioToolbox init error: %i\n", (int)status);
        av_free(channel_layout);
        return AVERROR_UNKNOWN;
    }

    if (!avctx->channel_layout)
        avctx->channel_layout = av_get_default_channel_layout(avctx->channels);

    if ((status = remap_layout(channel_layout, avctx->channel_layout, avctx->channels)) < 0) {
        av_log(avctx, AV_LOG_ERROR, "Invalid channel layout\n");
        av_free(channel_layout);
        return status;
    }

    if (AudioConverterSetProperty(at->converter, kAudioConverterInputChannelLayout,
                                  layout_size, channel_layout)) {
        av_log(avctx, AV_LOG_ERROR, "Unsupported input channel layout\n");
        av_free(channel_layout);
        return AVERROR(EINVAL);
    }
    if (avctx->codec_id == AV_CODEC_ID_AAC) {
        int tag = get_aac_tag(avctx->channel_layout);
        if (tag) {
            channel_layout->mChannelLayoutTag = tag;
            channel_layout->mNumberChannelDescriptions = 0;
        }
    }
    if (AudioConverterSetProperty(at->converter, kAudioConverterOutputChannelLayout,
                                  layout_size, channel_layout)) {
        av_log(avctx, AV_LOG_ERROR, "Unsupported output channel layout\n");
        av_free(channel_layout);
        return AVERROR(EINVAL);
    }
    av_free(channel_layout);

    if (avctx->bits_per_raw_sample)
        AudioConverterSetProperty(at->converter,
                                  kAudioConverterPropertyBitDepthHint,
                                  sizeof(avctx->bits_per_raw_sample),
                                  &avctx->bits_per_raw_sample);

#if !TARGET_OS_IPHONE
    if (at->mode == -1)
        at->mode = (avctx->flags & AV_CODEC_FLAG_QSCALE) ?
                   kAudioCodecBitRateControlMode_Variable :
                   kAudioCodecBitRateControlMode_Constant;

    AudioConverterSetProperty(at->converter, kAudioCodecPropertyBitRateControlMode,
                              sizeof(at->mode), &at->mode);

    if (at->mode == kAudioCodecBitRateControlMode_Variable) {
        int q = avctx->global_quality / FF_QP2LAMBDA;
        if (q < 0 || q > 14) {
            av_log(avctx, AV_LOG_WARNING,
                   "VBR quality %d out of range, should be 0-14\n", q);
            q = av_clip(q, 0, 14);
        }
        q = 127 - q * 9;
        AudioConverterSetProperty(at->converter, kAudioCodecPropertySoundQualityForVBR,
                                  sizeof(q), &q);
    } else
#endif
    if (avctx->bit_rate > 0) {
        UInt32 rate = avctx->bit_rate;
        UInt32 size;
        status = AudioConverterGetPropertyInfo(at->converter,
                                               kAudioConverterApplicableEncodeBitRates,
                                               &size, NULL);
        if (!status && size) {
            UInt32 new_rate = rate;
            int count;
            int i;
            AudioValueRange *ranges = av_malloc(size);
            if (!ranges)
                return AVERROR(ENOMEM);
            AudioConverterGetProperty(at->converter,
                                      kAudioConverterApplicableEncodeBitRates,
                                      &size, ranges);
            count = size / sizeof(AudioValueRange);
            for (i = 0; i < count; i++) {
                AudioValueRange *range = &ranges[i];
                if (rate >= range->mMinimum && rate <= range->mMaximum) {
                    new_rate = rate;
                    break;
                } else if (rate > range->mMaximum) {
                    new_rate = range->mMaximum;
                } else {
                    new_rate = range->mMinimum;
                    break;
                }
            }
            if (new_rate != rate) {
                av_log(avctx, AV_LOG_WARNING,
                       "Bitrate %u not allowed; changing to %u\n", rate, new_rate);
                rate = new_rate;
            }
            av_free(ranges);
        }
        AudioConverterSetProperty(at->converter, kAudioConverterEncodeBitRate,
                                  sizeof(rate), &rate);
    }

    at->quality = 96 - at->quality * 32;
    AudioConverterSetProperty(at->converter, kAudioConverterCodecQuality,
                              sizeof(at->quality), &at->quality);

    if (!AudioConverterGetPropertyInfo(at->converter, kAudioConverterCompressionMagicCookie,
                                       &avctx->extradata_size, NULL) &&
        avctx->extradata_size) {
        int extradata_size = avctx->extradata_size;
        uint8_t *extradata;
        if (!(avctx->extradata = av_mallocz(avctx->extradata_size + AV_INPUT_BUFFER_PADDING_SIZE)))
            return AVERROR(ENOMEM);
        if (avctx->codec_id == AV_CODEC_ID_ALAC) {
            avctx->extradata_size = 0x24;
            AV_WB32(avctx->extradata,     0x24);
            AV_WB32(avctx->extradata + 4, MKBETAG('a','l','a','c'));
            extradata = avctx->extradata + 12;
            avctx->extradata_size = 0x24;
        } else {
            extradata = avctx->extradata;
        }
        status = AudioConverterGetProperty(at->converter,
                                           kAudioConverterCompressionMagicCookie,
                                           &extradata_size, extradata);
        if (status != 0) {
            av_log(avctx, AV_LOG_ERROR, "AudioToolbox cookie error: %i\n", (int)status);
            return AVERROR_UNKNOWN;
        } else if (avctx->codec_id == AV_CODEC_ID_AAC) {
            GetByteContext gb;
            int tag, len;
            bytestream2_init(&gb, extradata, extradata_size);
            do {
                len = read_descr(&gb, &tag);
                if (tag == MP4DecConfigDescrTag) {
                    bytestream2_skip(&gb, 13);
                    len = read_descr(&gb, &tag);
                    if (tag == MP4DecSpecificDescrTag) {
                        len = FFMIN(gb.buffer_end - gb.buffer, len);
                        memmove(extradata, gb.buffer, len);
                        avctx->extradata_size = len;
                        break;
                    }
                } else if (tag == MP4ESDescrTag) {
                    int flags;
                    bytestream2_skip(&gb, 2);
                    flags = bytestream2_get_byte(&gb);
                    if (flags & 0x80) //streamDependenceFlag
                        bytestream2_skip(&gb, 2);
                    if (flags & 0x40) //URL_Flag
                        bytestream2_skip(&gb, bytestream2_get_byte(&gb));
                    if (flags & 0x20) //OCRstreamFlag
                        bytestream2_skip(&gb, 2);
                }
            } while (bytestream2_get_bytes_left(&gb));
        } else if (avctx->codec_id != AV_CODEC_ID_ALAC) {
            avctx->extradata_size = extradata_size;
        }
    }

    ffat_update_ctx(avctx);

#if !TARGET_OS_IPHONE && defined(__MAC_10_9)
    if (at->mode == kAudioCodecBitRateControlMode_Variable && avctx->rc_max_rate) {
        UInt32 max_size = avctx->rc_max_rate * avctx->frame_size / avctx->sample_rate;
        if (max_size)
            AudioConverterSetProperty(at->converter, kAudioCodecPropertyPacketSizeLimitForVBR,
                                      sizeof(max_size), &max_size);
    }
#endif

    ff_af_queue_init(avctx, &at->afq);

    at->encoding_frame = av_frame_alloc();
    if (!at->encoding_frame)
        return AVERROR(ENOMEM);

    return 0;
}

static OSStatus ffat_encode_callback(AudioConverterRef converter, UInt32 *nb_packets,
                                     AudioBufferList *data,
                                     AudioStreamPacketDescription **packets,
                                     void *inctx)
{
    AVCodecContext *avctx = inctx;
    ATDecodeContext *at = avctx->priv_data;
    AVFrame *frame;
    int ret;

    if (!at->frame_queue.available) {
        if (at->eof) {
            *nb_packets = 0;
            return 0;
        } else {
            *nb_packets = 0;
            return 1;
        }
    }

    frame = ff_bufqueue_get(&at->frame_queue);

    data->mNumberBuffers              = 1;
    data->mBuffers[0].mNumberChannels = avctx->channels;
    data->mBuffers[0].mDataByteSize   = frame->nb_samples *
                                        av_get_bytes_per_sample(avctx->sample_fmt) *
                                        avctx->channels;
    data->mBuffers[0].mData           = frame->data[0];
    if (*nb_packets > frame->nb_samples)
        *nb_packets = frame->nb_samples;

    av_frame_unref(at->encoding_frame);
    ret = av_frame_ref(at->encoding_frame, frame);
    if (ret < 0) {
        *nb_packets = 0;
        return ret;
    }

    ff_bufqueue_add(avctx, &at->used_frame_queue, frame);

    return 0;
}

static int ffat_encode(AVCodecContext *avctx, AVPacket *avpkt,
                       const AVFrame *frame, int *got_packet_ptr)
{
    ATDecodeContext *at = avctx->priv_data;
    OSStatus ret;

    AudioBufferList out_buffers = {
        .mNumberBuffers = 1,
        .mBuffers = {
            {
                .mNumberChannels = avctx->channels,
                .mDataByteSize = at->pkt_size,
            }
        }
    };
    AudioStreamPacketDescription out_pkt_desc = {0};

    if (frame) {
        AVFrame *in_frame;

        if (ff_bufqueue_is_full(&at->frame_queue)) {
            /*
             * The frame queue is significantly larger than needed in practice,
             * but no clear way to determine the minimum number of samples to
             * get output from AudioConverterFillComplexBuffer().
             */
            av_log(avctx, AV_LOG_ERROR, "Bug: frame queue is too small.\n");
            return AVERROR_BUG;
        }

        if ((ret = ff_af_queue_add(&at->afq, frame)) < 0)
            return ret;

        in_frame = av_frame_clone(frame);
        if (!in_frame)
            return AVERROR(ENOMEM);

        ff_bufqueue_add(avctx, &at->frame_queue, in_frame);
    } else {
        at->eof = 1;
    }

    if ((ret = ff_alloc_packet(avctx, avpkt, at->pkt_size)) < 0)
        return ret;


    out_buffers.mBuffers[0].mData = avpkt->data;

    *got_packet_ptr = avctx->frame_size / at->frame_size;

    ret = AudioConverterFillComplexBuffer(at->converter, ffat_encode_callback, avctx,
                                          got_packet_ptr, &out_buffers,
                                          (avctx->frame_size > at->frame_size) ? NULL : &out_pkt_desc);

    ff_bufqueue_discard_all(&at->used_frame_queue);

    if ((!ret || ret == 1) && *got_packet_ptr) {
        avpkt->size = out_buffers.mBuffers[0].mDataByteSize;
        ff_af_queue_remove(&at->afq, out_pkt_desc.mVariableFramesInPacket ?
                                     out_pkt_desc.mVariableFramesInPacket :
                                     avctx->frame_size,
                           &avpkt->pts,
                           &avpkt->duration);
    } else if (ret && ret != 1) {
        av_log(avctx, AV_LOG_WARNING, "Encode error: %i\n", ret);
    }

    return 0;
}

static av_cold void ffat_encode_flush(AVCodecContext *avctx)
{
    ATDecodeContext *at = avctx->priv_data;
    AudioConverterReset(at->converter);
    ff_bufqueue_discard_all(&at->frame_queue);
    ff_bufqueue_discard_all(&at->used_frame_queue);
}

static av_cold int ffat_close_encoder(AVCodecContext *avctx)
{
    ATDecodeContext *at = avctx->priv_data;
    AudioConverterDispose(at->converter);
    ff_bufqueue_discard_all(&at->frame_queue);
    ff_bufqueue_discard_all(&at->used_frame_queue);
    ff_af_queue_close(&at->afq);
    av_frame_free(&at->encoding_frame);
    return 0;
}

static const AVProfile aac_profiles[] = {
    { FF_PROFILE_AAC_LOW,   "LC"       },
    { FF_PROFILE_AAC_HE,    "HE-AAC"   },
    { FF_PROFILE_AAC_HE_V2, "HE-AACv2" },
    { FF_PROFILE_AAC_LD,    "LD"       },
    { FF_PROFILE_AAC_ELD,   "ELD"      },
    { FF_PROFILE_UNKNOWN },
};

#define AE AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_ENCODING_PARAM
static const AVOption options[] = {
#if !TARGET_OS_IPHONE
    {"aac_at_mode", "ratecontrol mode", offsetof(ATDecodeContext, mode), AV_OPT_TYPE_INT, {.i64 = -1}, -1, kAudioCodecBitRateControlMode_Variable, AE, "mode"},
        {"auto", "VBR if global quality is given; CBR otherwise", 0, AV_OPT_TYPE_CONST, {.i64 = -1}, INT_MIN, INT_MAX, AE, "mode"},
        {"cbr",  "constant bitrate", 0, AV_OPT_TYPE_CONST, {.i64 = kAudioCodecBitRateControlMode_Constant}, INT_MIN, INT_MAX, AE, "mode"},
        {"abr",  "long-term average bitrate", 0, AV_OPT_TYPE_CONST, {.i64 = kAudioCodecBitRateControlMode_LongTermAverage}, INT_MIN, INT_MAX, AE, "mode"},
        {"cvbr", "constrained variable bitrate", 0, AV_OPT_TYPE_CONST, {.i64 = kAudioCodecBitRateControlMode_VariableConstrained}, INT_MIN, INT_MAX, AE, "mode"},
        {"vbr" , "variable bitrate", 0, AV_OPT_TYPE_CONST, {.i64 = kAudioCodecBitRateControlMode_Variable}, INT_MIN, INT_MAX, AE, "mode"},
#endif
    {"aac_at_quality", "quality vs speed control", offsetof(ATDecodeContext, quality), AV_OPT_TYPE_INT, {.i64 = 0}, 0, 2, AE},
    { NULL },
};

#define FFAT_ENC_CLASS(NAME) \
    static const AVClass ffat_##NAME##_enc_class = { \
        .class_name = "at_" #NAME "_enc", \
        .item_name  = av_default_item_name, \
        .option     = options, \
        .version    = LIBAVUTIL_VERSION_INT, \
    };

#define FFAT_ENC(NAME, ID, PROFILES, ...) \
    FFAT_ENC_CLASS(NAME) \
    const AVCodec ff_##NAME##_at_encoder = { \
        .name           = #NAME "_at", \
        .long_name      = NULL_IF_CONFIG_SMALL(#NAME " (AudioToolbox)"), \
        .type           = AVMEDIA_TYPE_AUDIO, \
        .id             = ID, \
        .priv_data_size = sizeof(ATDecodeContext), \
        .init           = ffat_init_encoder, \
        .close          = ffat_close_encoder, \
        .encode2        = ffat_encode, \
        .flush          = ffat_encode_flush, \
        .priv_class     = &ffat_##NAME##_enc_class, \
        .capabilities   = AV_CODEC_CAP_DELAY | \
                          AV_CODEC_CAP_ENCODER_FLUSH __VA_ARGS__, \
        .sample_fmts    = (const enum AVSampleFormat[]) { \
            AV_SAMPLE_FMT_S16, \
            AV_SAMPLE_FMT_U8,  AV_SAMPLE_FMT_NONE \
        }, \
        .caps_internal  = FF_CODEC_CAP_INIT_THREADSAFE, \
        .profiles       = PROFILES, \
        .wrapper_name   = "at", \
    };

static const uint64_t aac_at_channel_layouts[] = {
    AV_CH_LAYOUT_MONO,
    AV_CH_LAYOUT_STEREO,
    AV_CH_LAYOUT_SURROUND,
    AV_CH_LAYOUT_4POINT0,
    AV_CH_LAYOUT_5POINT0,
    AV_CH_LAYOUT_5POINT1,
    AV_CH_LAYOUT_6POINT0,
    AV_CH_LAYOUT_6POINT1,
    AV_CH_LAYOUT_7POINT0,
    AV_CH_LAYOUT_7POINT1_WIDE_BACK,
    AV_CH_LAYOUT_QUAD,
    AV_CH_LAYOUT_OCTAGONAL,
    0,
};

FFAT_ENC(aac,          AV_CODEC_ID_AAC,          aac_profiles, , .channel_layouts = aac_at_channel_layouts)
//FFAT_ENC(adpcm_ima_qt, AV_CODEC_ID_ADPCM_IMA_QT, NULL)
FFAT_ENC(alac,         AV_CODEC_ID_ALAC,         NULL, | AV_CODEC_CAP_VARIABLE_FRAME_SIZE)
FFAT_ENC(ilbc,         AV_CODEC_ID_ILBC,         NULL)
FFAT_ENC(pcm_alaw,     AV_CODEC_ID_PCM_ALAW,     NULL)
FFAT_ENC(pcm_mulaw,    AV_CODEC_ID_PCM_MULAW,    NULL)