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FFmpeg/libavcodec/audiotoolboxenc.c
Philip Langdale 22b25b3ea5 avcodec: Add explicit capability flag for encoder flushing 
Previously, there was no way to flush an encoder such that after
draining, the encoder could be used again. We generally suggested
that clients teardown and replace the encoder instance in these
situations. However, for at least some hardware encoders, the cost of
this tear down/replace cycle is very high, which can get in the way of
some use-cases - for example: segmented encoding with nvenc.

To help address that use case, we added support for calling
avcodec_flush_buffers() to nvenc and things worked in practice,
although it was not clearly documented as to whether this should work
or not. There was only one previous example of an encoder implementing
the flush callback (audiotoolboxenc) and it's unclear if that was
intentional or not. However, it was clear that calling
avocdec_flush_buffers() on any other encoder would leave the encoder in
an undefined state, and that's not great.

As part of cleaning this up, this change introduces a formal capability
flag for encoders that support flushing and ensures a flush call is a
no-op for any other encoder. This allows client code to check if it is
meaningful to call flush on an encoder before actually doing it.

I have not attempted to separate the steps taken inside
avcodec_flush_buffers() because it's not doing anything that's wrong
for an encoder. But I did add a sanity check to reject attempts to
flush a frame threaded encoder because I couldn't wrap my head around
whether that code path was actually safe or not. As this combination
doesn't exist today, we'll deal with it if it ever comes up.
2020-04-15 14:54:42 -07:00

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/*
* 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;
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_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 | 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)
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