mirror of
https://github.com/pineappleEA/pineapple-src.git
synced 2024-12-10 13:48:25 -05:00
663 lines
23 KiB
C
Executable File
663 lines
23 KiB
C
Executable File
/*
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* Audio Toolbox system codecs
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*
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* copyright (c) 2016 Rodger Combs
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*
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* This file is part of FFmpeg.
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*
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* FFmpeg is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Lesser General Public
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* License as published by the Free Software Foundation; either
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* version 2.1 of the License, or (at your option) any later version.
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*
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* FFmpeg is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Lesser General Public License for more details.
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*
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* You should have received a copy of the GNU Lesser General Public
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* License along with FFmpeg; if not, write to the Free Software
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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*/
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#include <AudioToolbox/AudioToolbox.h>
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#define FF_BUFQUEUE_SIZE 256
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#include "libavfilter/bufferqueue.h"
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#include "config.h"
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#include "audio_frame_queue.h"
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#include "avcodec.h"
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#include "bytestream.h"
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#include "internal.h"
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#include "libavformat/isom.h"
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#include "libavutil/avassert.h"
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#include "libavutil/opt.h"
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#include "libavutil/log.h"
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typedef struct ATDecodeContext {
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AVClass *av_class;
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int mode;
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int quality;
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AudioConverterRef converter;
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struct FFBufQueue frame_queue;
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struct FFBufQueue used_frame_queue;
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unsigned pkt_size;
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AudioFrameQueue afq;
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int eof;
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int frame_size;
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AVFrame* encoding_frame;
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} ATDecodeContext;
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static UInt32 ffat_get_format_id(enum AVCodecID codec, int profile)
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{
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switch (codec) {
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case AV_CODEC_ID_AAC:
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switch (profile) {
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case FF_PROFILE_AAC_LOW:
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default:
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return kAudioFormatMPEG4AAC;
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case FF_PROFILE_AAC_HE:
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return kAudioFormatMPEG4AAC_HE;
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case FF_PROFILE_AAC_HE_V2:
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return kAudioFormatMPEG4AAC_HE_V2;
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case FF_PROFILE_AAC_LD:
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return kAudioFormatMPEG4AAC_LD;
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case FF_PROFILE_AAC_ELD:
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return kAudioFormatMPEG4AAC_ELD;
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}
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case AV_CODEC_ID_ADPCM_IMA_QT:
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return kAudioFormatAppleIMA4;
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case AV_CODEC_ID_ALAC:
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return kAudioFormatAppleLossless;
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case AV_CODEC_ID_ILBC:
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return kAudioFormatiLBC;
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case AV_CODEC_ID_PCM_ALAW:
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return kAudioFormatALaw;
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case AV_CODEC_ID_PCM_MULAW:
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return kAudioFormatULaw;
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default:
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av_assert0(!"Invalid codec ID!");
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return 0;
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}
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}
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static void ffat_update_ctx(AVCodecContext *avctx)
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{
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ATDecodeContext *at = avctx->priv_data;
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UInt32 size = sizeof(unsigned);
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AudioConverterPrimeInfo prime_info;
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AudioStreamBasicDescription out_format;
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AudioConverterGetProperty(at->converter,
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kAudioConverterPropertyMaximumOutputPacketSize,
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&size, &at->pkt_size);
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if (at->pkt_size <= 0)
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at->pkt_size = 1024 * 50;
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size = sizeof(prime_info);
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if (!AudioConverterGetProperty(at->converter,
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kAudioConverterPrimeInfo,
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&size, &prime_info)) {
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avctx->initial_padding = prime_info.leadingFrames;
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}
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size = sizeof(out_format);
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if (!AudioConverterGetProperty(at->converter,
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kAudioConverterCurrentOutputStreamDescription,
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&size, &out_format)) {
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if (out_format.mFramesPerPacket)
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avctx->frame_size = out_format.mFramesPerPacket;
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if (out_format.mBytesPerPacket && avctx->codec_id == AV_CODEC_ID_ILBC)
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avctx->block_align = out_format.mBytesPerPacket;
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}
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at->frame_size = avctx->frame_size;
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if (avctx->codec_id == AV_CODEC_ID_PCM_MULAW ||
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avctx->codec_id == AV_CODEC_ID_PCM_ALAW) {
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at->pkt_size *= 1024;
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avctx->frame_size *= 1024;
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}
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}
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static int read_descr(GetByteContext *gb, int *tag)
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{
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int len = 0;
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int count = 4;
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*tag = bytestream2_get_byte(gb);
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while (count--) {
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int c = bytestream2_get_byte(gb);
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len = (len << 7) | (c & 0x7f);
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if (!(c & 0x80))
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break;
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}
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return len;
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}
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static int get_ilbc_mode(AVCodecContext *avctx)
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{
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if (avctx->block_align == 38)
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return 20;
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else if (avctx->block_align == 50)
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return 30;
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else if (avctx->bit_rate > 0)
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return avctx->bit_rate <= 14000 ? 30 : 20;
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else
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return 30;
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}
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static av_cold int get_channel_label(int channel)
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{
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uint64_t map = 1 << channel;
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if (map <= AV_CH_LOW_FREQUENCY)
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return channel + 1;
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else if (map <= AV_CH_BACK_RIGHT)
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return channel + 29;
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else if (map <= AV_CH_BACK_CENTER)
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return channel - 1;
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else if (map <= AV_CH_SIDE_RIGHT)
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return channel - 4;
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else if (map <= AV_CH_TOP_BACK_RIGHT)
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return channel + 1;
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else if (map <= AV_CH_STEREO_RIGHT)
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return -1;
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else if (map <= AV_CH_WIDE_RIGHT)
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return channel + 4;
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else if (map <= AV_CH_SURROUND_DIRECT_RIGHT)
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return channel - 23;
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else if (map == AV_CH_LOW_FREQUENCY_2)
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return kAudioChannelLabel_LFE2;
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else
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return -1;
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}
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static int remap_layout(AudioChannelLayout *layout, uint64_t in_layout, int count)
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{
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int i;
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int c = 0;
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layout->mChannelLayoutTag = kAudioChannelLayoutTag_UseChannelDescriptions;
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layout->mNumberChannelDescriptions = count;
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for (i = 0; i < count; i++) {
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int label;
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while (!(in_layout & (1 << c)) && c < 64)
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c++;
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if (c == 64)
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return AVERROR(EINVAL); // This should never happen
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label = get_channel_label(c);
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layout->mChannelDescriptions[i].mChannelLabel = label;
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if (label < 0)
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return AVERROR(EINVAL);
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c++;
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}
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return 0;
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}
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static int get_aac_tag(uint64_t in_layout)
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{
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switch (in_layout) {
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case AV_CH_LAYOUT_MONO:
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return kAudioChannelLayoutTag_Mono;
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case AV_CH_LAYOUT_STEREO:
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return kAudioChannelLayoutTag_Stereo;
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case AV_CH_LAYOUT_QUAD:
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return kAudioChannelLayoutTag_AAC_Quadraphonic;
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case AV_CH_LAYOUT_OCTAGONAL:
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return kAudioChannelLayoutTag_AAC_Octagonal;
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case AV_CH_LAYOUT_SURROUND:
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return kAudioChannelLayoutTag_AAC_3_0;
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case AV_CH_LAYOUT_4POINT0:
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return kAudioChannelLayoutTag_AAC_4_0;
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case AV_CH_LAYOUT_5POINT0:
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return kAudioChannelLayoutTag_AAC_5_0;
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case AV_CH_LAYOUT_5POINT1:
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return kAudioChannelLayoutTag_AAC_5_1;
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case AV_CH_LAYOUT_6POINT0:
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return kAudioChannelLayoutTag_AAC_6_0;
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case AV_CH_LAYOUT_6POINT1:
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return kAudioChannelLayoutTag_AAC_6_1;
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case AV_CH_LAYOUT_7POINT0:
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return kAudioChannelLayoutTag_AAC_7_0;
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case AV_CH_LAYOUT_7POINT1_WIDE_BACK:
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return kAudioChannelLayoutTag_AAC_7_1;
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case AV_CH_LAYOUT_7POINT1:
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return kAudioChannelLayoutTag_MPEG_7_1_C;
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default:
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return 0;
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}
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}
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static av_cold int ffat_init_encoder(AVCodecContext *avctx)
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{
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ATDecodeContext *at = avctx->priv_data;
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OSStatus status;
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AudioStreamBasicDescription in_format = {
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.mSampleRate = avctx->sample_rate,
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.mFormatID = kAudioFormatLinearPCM,
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.mFormatFlags = ((avctx->sample_fmt == AV_SAMPLE_FMT_FLT ||
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avctx->sample_fmt == AV_SAMPLE_FMT_DBL) ? kAudioFormatFlagIsFloat
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: avctx->sample_fmt == AV_SAMPLE_FMT_U8 ? 0
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: kAudioFormatFlagIsSignedInteger)
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| kAudioFormatFlagIsPacked,
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.mBytesPerPacket = av_get_bytes_per_sample(avctx->sample_fmt) * avctx->channels,
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.mFramesPerPacket = 1,
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.mBytesPerFrame = av_get_bytes_per_sample(avctx->sample_fmt) * avctx->channels,
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.mChannelsPerFrame = avctx->channels,
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.mBitsPerChannel = av_get_bytes_per_sample(avctx->sample_fmt) * 8,
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};
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AudioStreamBasicDescription out_format = {
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.mSampleRate = avctx->sample_rate,
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.mFormatID = ffat_get_format_id(avctx->codec_id, avctx->profile),
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.mChannelsPerFrame = in_format.mChannelsPerFrame,
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};
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UInt32 layout_size = sizeof(AudioChannelLayout) +
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sizeof(AudioChannelDescription) * avctx->channels;
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AudioChannelLayout *channel_layout = av_malloc(layout_size);
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if (!channel_layout)
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return AVERROR(ENOMEM);
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if (avctx->codec_id == AV_CODEC_ID_ILBC) {
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int mode = get_ilbc_mode(avctx);
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out_format.mFramesPerPacket = 8000 * mode / 1000;
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out_format.mBytesPerPacket = (mode == 20 ? 38 : 50);
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}
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status = AudioConverterNew(&in_format, &out_format, &at->converter);
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if (status != 0) {
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av_log(avctx, AV_LOG_ERROR, "AudioToolbox init error: %i\n", (int)status);
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av_free(channel_layout);
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return AVERROR_UNKNOWN;
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}
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if (!avctx->channel_layout)
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avctx->channel_layout = av_get_default_channel_layout(avctx->channels);
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if ((status = remap_layout(channel_layout, avctx->channel_layout, avctx->channels)) < 0) {
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av_log(avctx, AV_LOG_ERROR, "Invalid channel layout\n");
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av_free(channel_layout);
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return status;
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}
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if (AudioConverterSetProperty(at->converter, kAudioConverterInputChannelLayout,
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layout_size, channel_layout)) {
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av_log(avctx, AV_LOG_ERROR, "Unsupported input channel layout\n");
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av_free(channel_layout);
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return AVERROR(EINVAL);
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}
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if (avctx->codec_id == AV_CODEC_ID_AAC) {
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int tag = get_aac_tag(avctx->channel_layout);
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if (tag) {
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channel_layout->mChannelLayoutTag = tag;
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channel_layout->mNumberChannelDescriptions = 0;
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}
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}
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if (AudioConverterSetProperty(at->converter, kAudioConverterOutputChannelLayout,
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layout_size, channel_layout)) {
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av_log(avctx, AV_LOG_ERROR, "Unsupported output channel layout\n");
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av_free(channel_layout);
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return AVERROR(EINVAL);
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}
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av_free(channel_layout);
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if (avctx->bits_per_raw_sample)
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AudioConverterSetProperty(at->converter,
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kAudioConverterPropertyBitDepthHint,
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sizeof(avctx->bits_per_raw_sample),
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&avctx->bits_per_raw_sample);
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#if !TARGET_OS_IPHONE
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if (at->mode == -1)
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at->mode = (avctx->flags & AV_CODEC_FLAG_QSCALE) ?
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kAudioCodecBitRateControlMode_Variable :
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kAudioCodecBitRateControlMode_Constant;
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AudioConverterSetProperty(at->converter, kAudioCodecPropertyBitRateControlMode,
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sizeof(at->mode), &at->mode);
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if (at->mode == kAudioCodecBitRateControlMode_Variable) {
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int q = avctx->global_quality / FF_QP2LAMBDA;
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if (q < 0 || q > 14) {
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av_log(avctx, AV_LOG_WARNING,
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"VBR quality %d out of range, should be 0-14\n", q);
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q = av_clip(q, 0, 14);
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}
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q = 127 - q * 9;
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AudioConverterSetProperty(at->converter, kAudioCodecPropertySoundQualityForVBR,
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sizeof(q), &q);
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} else
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#endif
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if (avctx->bit_rate > 0) {
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UInt32 rate = avctx->bit_rate;
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UInt32 size;
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status = AudioConverterGetPropertyInfo(at->converter,
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kAudioConverterApplicableEncodeBitRates,
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&size, NULL);
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if (!status && size) {
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UInt32 new_rate = rate;
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int count;
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int i;
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AudioValueRange *ranges = av_malloc(size);
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if (!ranges)
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return AVERROR(ENOMEM);
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AudioConverterGetProperty(at->converter,
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kAudioConverterApplicableEncodeBitRates,
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&size, ranges);
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count = size / sizeof(AudioValueRange);
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for (i = 0; i < count; i++) {
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AudioValueRange *range = &ranges[i];
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if (rate >= range->mMinimum && rate <= range->mMaximum) {
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new_rate = rate;
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break;
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} else if (rate > range->mMaximum) {
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new_rate = range->mMaximum;
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} else {
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new_rate = range->mMinimum;
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break;
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}
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}
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if (new_rate != rate) {
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av_log(avctx, AV_LOG_WARNING,
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"Bitrate %u not allowed; changing to %u\n", rate, new_rate);
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rate = new_rate;
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}
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av_free(ranges);
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}
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AudioConverterSetProperty(at->converter, kAudioConverterEncodeBitRate,
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sizeof(rate), &rate);
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}
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at->quality = 96 - at->quality * 32;
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AudioConverterSetProperty(at->converter, kAudioConverterCodecQuality,
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sizeof(at->quality), &at->quality);
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if (!AudioConverterGetPropertyInfo(at->converter, kAudioConverterCompressionMagicCookie,
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&avctx->extradata_size, NULL) &&
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avctx->extradata_size) {
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int extradata_size = avctx->extradata_size;
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uint8_t *extradata;
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if (!(avctx->extradata = av_mallocz(avctx->extradata_size + AV_INPUT_BUFFER_PADDING_SIZE)))
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return AVERROR(ENOMEM);
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if (avctx->codec_id == AV_CODEC_ID_ALAC) {
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avctx->extradata_size = 0x24;
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AV_WB32(avctx->extradata, 0x24);
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AV_WB32(avctx->extradata + 4, MKBETAG('a','l','a','c'));
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extradata = avctx->extradata + 12;
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avctx->extradata_size = 0x24;
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} else {
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extradata = avctx->extradata;
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}
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status = AudioConverterGetProperty(at->converter,
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kAudioConverterCompressionMagicCookie,
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&extradata_size, extradata);
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if (status != 0) {
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av_log(avctx, AV_LOG_ERROR, "AudioToolbox cookie error: %i\n", (int)status);
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return AVERROR_UNKNOWN;
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} else if (avctx->codec_id == AV_CODEC_ID_AAC) {
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GetByteContext gb;
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int tag, len;
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bytestream2_init(&gb, extradata, extradata_size);
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do {
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len = read_descr(&gb, &tag);
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if (tag == MP4DecConfigDescrTag) {
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bytestream2_skip(&gb, 13);
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len = read_descr(&gb, &tag);
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if (tag == MP4DecSpecificDescrTag) {
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len = FFMIN(gb.buffer_end - gb.buffer, len);
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memmove(extradata, gb.buffer, len);
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avctx->extradata_size = len;
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break;
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}
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} else if (tag == MP4ESDescrTag) {
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int flags;
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bytestream2_skip(&gb, 2);
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flags = bytestream2_get_byte(&gb);
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if (flags & 0x80) //streamDependenceFlag
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bytestream2_skip(&gb, 2);
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if (flags & 0x40) //URL_Flag
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bytestream2_skip(&gb, bytestream2_get_byte(&gb));
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if (flags & 0x20) //OCRstreamFlag
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bytestream2_skip(&gb, 2);
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}
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} while (bytestream2_get_bytes_left(&gb));
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} else if (avctx->codec_id != AV_CODEC_ID_ALAC) {
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avctx->extradata_size = extradata_size;
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}
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}
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ffat_update_ctx(avctx);
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#if !TARGET_OS_IPHONE && defined(__MAC_10_9)
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if (at->mode == kAudioCodecBitRateControlMode_Variable && avctx->rc_max_rate) {
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UInt32 max_size = avctx->rc_max_rate * avctx->frame_size / avctx->sample_rate;
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if (max_size)
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AudioConverterSetProperty(at->converter, kAudioCodecPropertyPacketSizeLimitForVBR,
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sizeof(max_size), &max_size);
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}
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#endif
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ff_af_queue_init(avctx, &at->afq);
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at->encoding_frame = av_frame_alloc();
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if (!at->encoding_frame)
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return AVERROR(ENOMEM);
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return 0;
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}
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static OSStatus ffat_encode_callback(AudioConverterRef converter, UInt32 *nb_packets,
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AudioBufferList *data,
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AudioStreamPacketDescription **packets,
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void *inctx)
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{
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AVCodecContext *avctx = inctx;
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ATDecodeContext *at = avctx->priv_data;
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AVFrame *frame;
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int ret;
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if (!at->frame_queue.available) {
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if (at->eof) {
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*nb_packets = 0;
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return 0;
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} else {
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*nb_packets = 0;
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return 1;
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}
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}
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frame = ff_bufqueue_get(&at->frame_queue);
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data->mNumberBuffers = 1;
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data->mBuffers[0].mNumberChannels = avctx->channels;
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data->mBuffers[0].mDataByteSize = frame->nb_samples *
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|
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_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);
|
|
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) \
|
|
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 | \
|
|
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)
|