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ffmpeg.wasm-core
Commit 65cff814 authored by Rodger Combs's avatar Rodger Combs
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lavc: add AudioToolbox encoders 

Fixes trac #4828
parent d5d32805
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Showing with 493 additions and 6 deletions
Changelog
View file @ 65cff814
......@@ -14,6 +14,7 @@ version <next>:
- MediaCodec H264 decoding
- VC-2 HQ RTP payload format (draft v1) depacketizer
- AudioToolbox audio decoders
- AudioToolbox audio encoders
version 3.0:
......
configure
View file @ 65cff814
......@@ -2661,6 +2661,16 @@ pcm_alaw_at_decoder_deps="audiotoolbox"
pcm_mulaw_at_decoder_deps="audiotoolbox"
qdmc_at_decoder_deps="audiotoolbox"
qdm2_at_decoder_deps="audiotoolbox"
aac_at_encoder_deps="audiotoolbox"
aac_at_encoder_select="audio_frame_queue"
alac_at_encoder_deps="audiotoolbox"
alac_at_encoder_select="audio_frame_queue"
ilbc_at_encoder_deps="audiotoolbox"
ilbc_at_encoder_select="audio_frame_queue"
pcm_alaw_at_encoder_deps="audiotoolbox"
pcm_alaw_at_encoder_select="audio_frame_queue"
pcm_mulaw_at_encoder_deps="audiotoolbox"
pcm_mulaw_at_encoder_select="audio_frame_queue"
chromaprint_muxer_deps="chromaprint"
h264_videotoolbox_encoder_deps="videotoolbox_encoder pthreads"
libcelt_decoder_deps="libcelt"
......
libavcodec/Makefile
View file @ 65cff814
......@@ -815,6 +815,11 @@ OBJS-$(CONFIG_PCM_MULAW_AT_DECODER) += audiotoolboxdec.o
OBJS-$(CONFIG_PCM_ALAW_AT_DECODER) += audiotoolboxdec.o
OBJS-$(CONFIG_QDMC_AT_DECODER) += audiotoolboxdec.o
OBJS-$(CONFIG_QDM2_AT_DECODER) += audiotoolboxdec.o
OBJS-$(CONFIG_AAC_AT_ENCODER) += audiotoolboxenc.o
OBJS-$(CONFIG_ALAC_AT_ENCODER) += audiotoolboxenc.o
OBJS-$(CONFIG_ILBC_AT_ENCODER) += audiotoolboxenc.o
OBJS-$(CONFIG_PCM_ALAW_AT_ENCODER) += audiotoolboxenc.o
OBJS-$(CONFIG_PCM_MULAW_AT_ENCODER) += audiotoolboxenc.o
OBJS-$(CONFIG_LIBCELT_DECODER) += libcelt_dec.o
OBJS-$(CONFIG_LIBDCADEC_DECODER) += libdcadec.o dca.o
OBJS-$(CONFIG_LIBFAAC_ENCODER) += libfaac.o
......
libavcodec/allcodecs.c
View file @ 65cff814
......@@ -563,18 +563,18 @@ void avcodec_register_all(void)
REGISTER_ENCDEC (XSUB, xsub);
/* external libraries */
REGISTER_DECODER(AAC_AT, aac_at);
REGISTER_ENCDEC (AAC_AT, aac_at);
REGISTER_DECODER(AC3_AT, ac3_at);
REGISTER_DECODER(ADPCM_IMA_QT_AT, adpcm_ima_qt_at);
REGISTER_DECODER(ALAC_AT, alac_at);
REGISTER_ENCDEC (ALAC_AT, alac_at);
REGISTER_DECODER(AMR_NB_AT, amr_nb_at);
REGISTER_DECODER(GSM_MS_AT, gsm_ms_at);
REGISTER_DECODER(ILBC_AT, ilbc_at);
REGISTER_ENCDEC (ILBC_AT, ilbc_at);
REGISTER_DECODER(MP1_AT, mp1_at);
REGISTER_DECODER(MP2_AT, mp2_at);
REGISTER_DECODER(MP3_AT, mp3_at);
REGISTER_DECODER(PCM_ALAW_AT, pcm_alaw_at);
REGISTER_DECODER(PCM_MULAW_AT, pcm_mulaw_at);
REGISTER_ENCDEC (PCM_ALAW_AT, pcm_alaw_at);
REGISTER_ENCDEC (PCM_MULAW_AT, pcm_mulaw_at);
REGISTER_DECODER(QDMC_AT, qdmc_at);
REGISTER_DECODER(QDM2_AT, qdm2_at);
REGISTER_DECODER(LIBCELT, libcelt);
......
libavcodec/audiotoolboxenc.c 0 → 100644
View file @ 65cff814
/*
* 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>
#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;
AudioStreamPacketDescription pkt_desc;
AVFrame in_frame;
AVFrame new_in_frame;
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 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,
};
AudioChannelLayout channel_layout = {
.mChannelLayoutTag = kAudioChannelLayoutTag_UseChannelBitmap,
.mChannelBitmap = avctx->channel_layout,
};
UInt32 size = sizeof(channel_layout);
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);
return AVERROR_UNKNOWN;
}
size = sizeof(UInt32);
AudioConverterSetProperty(at->converter, kAudioConverterInputChannelLayout,
size, &channel_layout);
AudioConverterSetProperty(at->converter, kAudioConverterOutputChannelLayout,
size, &channel_layout);
if (avctx->bits_per_raw_sample) {
size = sizeof(avctx->bits_per_raw_sample);
AudioConverterSetProperty(at->converter,
kAudioConverterPropertyBitDepthHint,
size, &avctx->bits_per_raw_sample);
}
if (at->mode == -1)
at->mode = (avctx->flags & AV_CODEC_FLAG_QSCALE) ?
kAudioCodecBitRateControlMode_Variable :
kAudioCodecBitRateControlMode_Constant;
AudioConverterSetProperty(at->converter, kAudioCodecPropertyBitRateControlMode,
size, &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,
size, &q);
} else if (avctx->bit_rate > 0) {
UInt32 rate = avctx->bit_rate;
AudioConverterSetProperty(at->converter, kAudioConverterEncodeBitRate,
size, &rate);
}
at->quality = 96 - at->quality * 32;
AudioConverterSetProperty(at->converter, kAudioConverterCodecQuality,
size, &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 (at->mode == kAudioCodecBitRateControlMode_Variable && avctx->rc_max_rate) {
int max_size = avctx->rc_max_rate * avctx->frame_size / avctx->sample_rate;
if (max_size)
AudioConverterSetProperty(at->converter, kAudioCodecPropertyPacketSizeLimitForVBR,
size, &max_size);
}
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;
if (at->eof) {
*nb_packets = 0;
if (packets) {
*packets = &at->pkt_desc;
at->pkt_desc.mDataByteSize = 0;
}
return 0;
}
av_frame_unref(&at->in_frame);
av_frame_move_ref(&at->in_frame, &at->new_in_frame);
if (!at->in_frame.data[0]) {
*nb_packets = 0;
return 1;
}
data->mNumberBuffers = 1;
data->mBuffers[0].mNumberChannels = 0;
data->mBuffers[0].mDataByteSize = at->in_frame.nb_samples *
av_get_bytes_per_sample(avctx->sample_fmt) *
avctx->channels;
data->mBuffers[0].mData = at->in_frame.data[0];
*nb_packets = (at->in_frame.nb_samples + (at->frame_size - 1)) / at->frame_size;
if (packets) {
*packets = &at->pkt_desc;
at->pkt_desc.mDataByteSize = data->mBuffers[0].mDataByteSize;
at->pkt_desc.mVariableFramesInPacket = at->in_frame.nb_samples;
}
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 ((ret = ff_alloc_packet2(avctx, avpkt, at->pkt_size, 0)) < 0)
return ret;
av_frame_unref(&at->new_in_frame);
if (frame) {
if ((ret = ff_af_queue_add(&at->afq, frame)) < 0)
return ret;
if ((ret = av_frame_ref(&at->new_in_frame, frame)) < 0)
return ret;
} else {
at->eof = 1;
}
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);
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);
av_frame_unref(&at->new_in_frame);
av_frame_unref(&at->in_frame);
}
static av_cold int ffat_close_encoder(AVCodecContext *avctx)
{
ATDecodeContext *at = avctx->priv_data;
AudioConverterDispose(at->converter);
av_frame_unref(&at->new_in_frame);
av_frame_unref(&at->in_frame);
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[] = {
{"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"},
{"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, \
};
FFAT_ENC(aac, AV_CODEC_ID_AAC, aac_profiles)
//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)
libavcodec/version.h
View file @ 65cff814
......@@ -28,7 +28,7 @@
#include "libavutil/version.h"
#define LIBAVCODEC_VERSION_MAJOR 57
#define LIBAVCODEC_VERSION_MINOR 29
#define LIBAVCODEC_VERSION_MINOR 30
#define LIBAVCODEC_VERSION_MICRO 100
#define LIBAVCODEC_VERSION_INT AV_VERSION_INT(LIBAVCODEC_VERSION_MAJOR, \
......
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