/*
  * Audio Toolbox system codecs
  *
  * copyright (c) 2016 Rodger Combs
  *
  * 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 "internal.h"
 #include "libavformat/isom.h"
 #include "libavutil/avassert.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;
 } 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);
 
     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;

     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;
 
     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_packet2(avctx, avpkt, at->pkt_size, 0)) < 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);
     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) \
     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_DR1 | AV_CODEC_CAP_DELAY __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, \
     };
 

 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 | AV_CODEC_CAP_LOSSLESS)
 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)