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ffmpeg/libavcodec/audiotoolboxenc.c

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/*
* 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 "codec_internal.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 AV_PROFILE_AAC_LOW:
default:
return kAudioFormatMPEG4AAC;
case AV_PROFILE_AAC_HE:
return kAudioFormatMPEG4AAC_HE;
case AV_PROFILE_AAC_HE_V2:
return kAudioFormatMPEG4AAC_HE_V2;
case AV_PROFILE_AAC_LD:
return kAudioFormatMPEG4AAC_LD;
#if MAC_OS_X_VERSION_MIN_REQUIRED >= 1060
case AV_PROFILE_AAC_ELD:
return kAudioFormatMPEG4AAC_ELD;
#endif
}
case AV_CODEC_ID_ADPCM_IMA_QT:
return kAudioFormatAppleIMA4;
case AV_CODEC_ID_ALAC:
return kAudioFormatAppleLossless;
#if MAC_OS_X_VERSION_MIN_REQUIRED >= 1060
case AV_CODEC_ID_ILBC:
return kAudioFormatiLBC;
#endif
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 int 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;
} else {
/* The doc on mFramesPerPacket says:
* For formats with a variable number of frames per packet, such as
* Ogg Vorbis, set this field to 0.
* Looks like it means for decoding. There is no known case that
* mFramesPerPacket is zero for encoding. Use a default value for safety.
*/
avctx->frame_size = 1024;
av_log(avctx, AV_LOG_WARNING, "Missing mFramesPerPacket\n");
}
if (out_format.mBytesPerPacket && avctx->codec_id == AV_CODEC_ID_ILBC)
avctx->block_align = out_format.mBytesPerPacket;
} else {
av_log(avctx, AV_LOG_ERROR, "Get OutputStreamDescription failed\n");
return AVERROR_EXTERNAL;
}
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;
}
return 0;
}
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, const AVChannelLayout *in_layout)
{
int i;
layout->mChannelLayoutTag = kAudioChannelLayoutTag_UseChannelDescriptions;
layout->mNumberChannelDescriptions = in_layout->nb_channels;
for (i = 0; i < in_layout->nb_channels; i++) {
int c, label;
c = av_channel_layout_channel_from_index(in_layout, i);
if (c < 0 || c >= 64)
return AVERROR(EINVAL);
label = get_channel_label(c);
layout->mChannelDescriptions[i].mChannelLabel = label;
if (label < 0)
return AVERROR(EINVAL);
c++;
}
return 0;
}
static int get_aac_tag(const AVChannelLayout *in_layout)
{
static const struct {
AVChannelLayout chl;
int tag;
} map[] = {
{ AV_CHANNEL_LAYOUT_MONO, kAudioChannelLayoutTag_Mono },
{ AV_CHANNEL_LAYOUT_STEREO, kAudioChannelLayoutTag_Stereo },
{ AV_CHANNEL_LAYOUT_QUAD, kAudioChannelLayoutTag_AAC_Quadraphonic },
{ AV_CHANNEL_LAYOUT_OCTAGONAL, kAudioChannelLayoutTag_AAC_Octagonal },
{ AV_CHANNEL_LAYOUT_SURROUND, kAudioChannelLayoutTag_AAC_3_0 },
{ AV_CHANNEL_LAYOUT_4POINT0, kAudioChannelLayoutTag_AAC_4_0 },
{ AV_CHANNEL_LAYOUT_5POINT0, kAudioChannelLayoutTag_AAC_5_0 },
{ AV_CHANNEL_LAYOUT_5POINT1, kAudioChannelLayoutTag_AAC_5_1 },
{ AV_CHANNEL_LAYOUT_6POINT0, kAudioChannelLayoutTag_AAC_6_0 },
{ AV_CHANNEL_LAYOUT_6POINT1, kAudioChannelLayoutTag_AAC_6_1 },
{ AV_CHANNEL_LAYOUT_7POINT0, kAudioChannelLayoutTag_AAC_7_0 },
{ AV_CHANNEL_LAYOUT_7POINT1_WIDE_BACK, kAudioChannelLayoutTag_AAC_7_1 },
{ AV_CHANNEL_LAYOUT_7POINT1, kAudioChannelLayoutTag_MPEG_7_1_C },
};
int i;
for (i = 0; i < FF_ARRAY_ELEMS(map); i++)
if (!av_channel_layout_compare(in_layout, &map[i].chl))
return map[i].tag;
return 0;
}
static av_cold int ffat_init_encoder(AVCodecContext *avctx)
{
ATDecodeContext *at = avctx->priv_data;
OSStatus status;
int ret;
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->ch_layout.nb_channels,
.mFramesPerPacket = 1,
.mBytesPerFrame = av_get_bytes_per_sample(avctx->sample_fmt) * avctx->ch_layout.nb_channels,
.mChannelsPerFrame = avctx->ch_layout.nb_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->ch_layout.nb_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->ch_layout.order == AV_CHANNEL_ORDER_UNSPEC)
av_channel_layout_default(&avctx->ch_layout, avctx->ch_layout.nb_channels);
if ((status = remap_layout(channel_layout, &avctx->ch_layout)) < 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->ch_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;
}
}
ret = ffat_update_ctx(avctx);
if (ret < 0)
return ret;
#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->ch_layout.nb_channels;
data->mBuffers[0].mDataByteSize = frame->nb_samples *
av_get_bytes_per_sample(avctx->sample_fmt) *
avctx->ch_layout.nb_channels;
data->mBuffers[0].mData = frame->data[0];
if (*nb_packets > frame->nb_samples)
*nb_packets = frame->nb_samples;
ret = av_frame_replace(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->ch_layout.nb_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_ERROR, "Encode error: %i\n", ret);
return AVERROR_EXTERNAL;
}
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[] = {
{ AV_PROFILE_AAC_LOW, "LC" },
{ AV_PROFILE_AAC_HE, "HE-AAC" },
{ AV_PROFILE_AAC_HE_V2, "HE-AACv2" },
{ AV_PROFILE_AAC_LD, "LD" },
{ AV_PROFILE_AAC_ELD, "ELD" },
{ AV_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, .unit = "mode"},
{"auto", "VBR if global quality is given; CBR otherwise", 0, AV_OPT_TYPE_CONST, {.i64 = -1}, INT_MIN, INT_MAX, AE, .unit = "mode"},
{"cbr", "constant bitrate", 0, AV_OPT_TYPE_CONST, {.i64 = kAudioCodecBitRateControlMode_Constant}, INT_MIN, INT_MAX, AE, .unit = "mode"},
{"abr", "long-term average bitrate", 0, AV_OPT_TYPE_CONST, {.i64 = kAudioCodecBitRateControlMode_LongTermAverage}, INT_MIN, INT_MAX, AE, .unit = "mode"},
{"cvbr", "constrained variable bitrate", 0, AV_OPT_TYPE_CONST, {.i64 = kAudioCodecBitRateControlMode_VariableConstrained}, INT_MIN, INT_MAX, AE, .unit = "mode"},
{"vbr" , "variable bitrate", 0, AV_OPT_TYPE_CONST, {.i64 = kAudioCodecBitRateControlMode_Variable}, INT_MIN, INT_MAX, AE, .unit = "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, CAPS, CHANNEL_LAYOUTS, CH_LAYOUTS) \
FFAT_ENC_CLASS(NAME) \
const FFCodec ff_##NAME##_at_encoder = { \
.p.name = #NAME "_at", \
CODEC_LONG_NAME(#NAME " (AudioToolbox)"), \
.p.type = AVMEDIA_TYPE_AUDIO, \
.p.id = ID, \
.priv_data_size = sizeof(ATDecodeContext), \
.init = ffat_init_encoder, \
.close = ffat_close_encoder, \
FF_CODEC_ENCODE_CB(ffat_encode), \
.flush = ffat_encode_flush, \
.p.priv_class = &ffat_##NAME##_enc_class, \
.p.capabilities = AV_CODEC_CAP_DELAY | \
AV_CODEC_CAP_ENCODER_FLUSH CAPS, \
CODEC_OLD_CHANNEL_LAYOUTS_ARRAY(CHANNEL_LAYOUTS) \
.p.ch_layouts = CH_LAYOUTS, \
.p.sample_fmts = (const enum AVSampleFormat[]) { \
AV_SAMPLE_FMT_S16, \
AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_NONE \
}, \
.p.profiles = PROFILES, \
.p.wrapper_name = "at", \
};
static const AVChannelLayout aac_at_ch_layouts[] = {
AV_CHANNEL_LAYOUT_MONO,
AV_CHANNEL_LAYOUT_STEREO,
AV_CHANNEL_LAYOUT_SURROUND,
AV_CHANNEL_LAYOUT_4POINT0,
AV_CHANNEL_LAYOUT_5POINT0,
AV_CHANNEL_LAYOUT_5POINT1,
AV_CHANNEL_LAYOUT_6POINT0,
AV_CHANNEL_LAYOUT_6POINT1,
AV_CHANNEL_LAYOUT_7POINT0,
AV_CHANNEL_LAYOUT_7POINT1_WIDE_BACK,
AV_CHANNEL_LAYOUT_QUAD,
AV_CHANNEL_LAYOUT_OCTAGONAL,
{ 0 },
};
#if FF_API_OLD_CHANNEL_LAYOUT
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,
};
#endif
FFAT_ENC(aac, AV_CODEC_ID_AAC, aac_profiles, , aac_at_channel_layouts, aac_at_ch_layouts)
//FFAT_ENC(adpcm_ima_qt, AV_CODEC_ID_ADPCM_IMA_QT, NULL)
FFAT_ENC(alac, AV_CODEC_ID_ALAC, NULL, , NULL, NULL)
FFAT_ENC(ilbc, AV_CODEC_ID_ILBC, NULL, , NULL, NULL)
FFAT_ENC(pcm_alaw, AV_CODEC_ID_PCM_ALAW, NULL, , NULL, NULL)
FFAT_ENC(pcm_mulaw, AV_CODEC_ID_PCM_MULAW, NULL, , NULL, NULL)
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