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ffmpeg/libavcodec/audiotoolboxdec.c

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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>
#include "config.h"
#include "avcodec.h"
#include "ac3_parser.h"
#include "bytestream.h"
#include "internal.h"
#include "mpegaudiodecheader.h"
#include "libavutil/avassert.h"
#include "libavutil/opt.h"
#include "libavutil/log.h"
#ifndef __MAC_10_11
#define kAudioFormatEnhancedAC3 'ec-3'
#endif
typedef struct ATDecodeContext {
AVClass *av_class;
AudioConverterRef converter;
AudioStreamPacketDescription pkt_desc;
AVPacket in_pkt;
AVPacket new_in_pkt;
AVBSFContext *bsf;
char *decoded_data;
int channel_map[64];
uint8_t *extradata;
int extradata_size;
int64_t last_pts;
int eof;
} ATDecodeContext;
static UInt32 ffat_get_format_id(enum AVCodecID codec, int profile)
{
switch (codec) {
case AV_CODEC_ID_AAC:
return kAudioFormatMPEG4AAC;
case AV_CODEC_ID_AC3:
return kAudioFormatAC3;
case AV_CODEC_ID_ADPCM_IMA_QT:
return kAudioFormatAppleIMA4;
case AV_CODEC_ID_ALAC:
return kAudioFormatAppleLossless;
case AV_CODEC_ID_AMR_NB:
return kAudioFormatAMR;
case AV_CODEC_ID_EAC3:
return kAudioFormatEnhancedAC3;
case AV_CODEC_ID_GSM_MS:
return kAudioFormatMicrosoftGSM;
case AV_CODEC_ID_ILBC:
return kAudioFormatiLBC;
case AV_CODEC_ID_MP1:
return kAudioFormatMPEGLayer1;
case AV_CODEC_ID_MP2:
return kAudioFormatMPEGLayer2;
case AV_CODEC_ID_MP3:
return kAudioFormatMPEGLayer3;
case AV_CODEC_ID_PCM_ALAW:
return kAudioFormatALaw;
case AV_CODEC_ID_PCM_MULAW:
return kAudioFormatULaw;
case AV_CODEC_ID_QDMC:
return kAudioFormatQDesign;
case AV_CODEC_ID_QDM2:
return kAudioFormatQDesign2;
default:
av_assert0(!"Invalid codec ID!");
return 0;
}
}
static int ffat_get_channel_id(AudioChannelLabel label)
{
if (label == 0)
return -1;
else if (label <= kAudioChannelLabel_LFEScreen)
return label - 1;
else if (label <= kAudioChannelLabel_RightSurround)
return label + 4;
else if (label <= kAudioChannelLabel_CenterSurround)
return label + 1;
else if (label <= kAudioChannelLabel_RightSurroundDirect)
return label + 23;
else if (label <= kAudioChannelLabel_TopBackRight)
return label - 1;
else if (label < kAudioChannelLabel_RearSurroundLeft)
return -1;
else if (label <= kAudioChannelLabel_RearSurroundRight)
return label - 29;
else if (label <= kAudioChannelLabel_RightWide)
return label - 4;
else if (label == kAudioChannelLabel_LFE2)
return ff_ctzll(AV_CH_LOW_FREQUENCY_2);
else if (label == kAudioChannelLabel_Mono)
return ff_ctzll(AV_CH_FRONT_CENTER);
else
return -1;
}
static int ffat_compare_channel_descriptions(const void* a, const void* b)
{
const AudioChannelDescription* da = a;
const AudioChannelDescription* db = b;
return ffat_get_channel_id(da->mChannelLabel) - ffat_get_channel_id(db->mChannelLabel);
}
static AudioChannelLayout *ffat_convert_layout(AudioChannelLayout *layout, UInt32* size)
{
AudioChannelLayoutTag tag = layout->mChannelLayoutTag;
AudioChannelLayout *new_layout;
if (tag == kAudioChannelLayoutTag_UseChannelDescriptions)
return layout;
else if (tag == kAudioChannelLayoutTag_UseChannelBitmap)
AudioFormatGetPropertyInfo(kAudioFormatProperty_ChannelLayoutForBitmap,
sizeof(UInt32), &layout->mChannelBitmap, size);
else
AudioFormatGetPropertyInfo(kAudioFormatProperty_ChannelLayoutForTag,
sizeof(AudioChannelLayoutTag), &tag, size);
new_layout = av_malloc(*size);
if (!new_layout) {
av_free(layout);
return NULL;
}
if (tag == kAudioChannelLayoutTag_UseChannelBitmap)
AudioFormatGetProperty(kAudioFormatProperty_ChannelLayoutForBitmap,
sizeof(UInt32), &layout->mChannelBitmap, size, new_layout);
else
AudioFormatGetProperty(kAudioFormatProperty_ChannelLayoutForTag,
sizeof(AudioChannelLayoutTag), &tag, size, new_layout);
new_layout->mChannelLayoutTag = kAudioChannelLayoutTag_UseChannelDescriptions;
av_free(layout);
return new_layout;
}
static int ffat_update_ctx(AVCodecContext *avctx)
{
ATDecodeContext *at = avctx->priv_data;
AudioStreamBasicDescription format;
UInt32 size = sizeof(format);
if (!AudioConverterGetProperty(at->converter,
kAudioConverterCurrentInputStreamDescription,
&size, &format)) {
if (format.mSampleRate)
avctx->sample_rate = format.mSampleRate;
avctx->channels = format.mChannelsPerFrame;
avctx->channel_layout = av_get_default_channel_layout(avctx->channels);
avctx->frame_size = format.mFramesPerPacket;
}
if (!AudioConverterGetProperty(at->converter,
kAudioConverterCurrentOutputStreamDescription,
&size, &format)) {
format.mSampleRate = avctx->sample_rate;
format.mChannelsPerFrame = avctx->channels;
AudioConverterSetProperty(at->converter,
kAudioConverterCurrentOutputStreamDescription,
size, &format);
}
if (!AudioConverterGetPropertyInfo(at->converter, kAudioConverterOutputChannelLayout,
&size, NULL) && size) {
AudioChannelLayout *layout = av_malloc(size);
uint64_t layout_mask = 0;
int i;
if (!layout)
return AVERROR(ENOMEM);
AudioConverterGetProperty(at->converter, kAudioConverterOutputChannelLayout,
&size, layout);
if (!(layout = ffat_convert_layout(layout, &size)))
return AVERROR(ENOMEM);
for (i = 0; i < layout->mNumberChannelDescriptions; i++) {
int id = ffat_get_channel_id(layout->mChannelDescriptions[i].mChannelLabel);
if (id < 0)
goto done;
if (layout_mask & (1 << id))
goto done;
layout_mask |= 1 << id;
layout->mChannelDescriptions[i].mChannelFlags = i; // Abusing flags as index
}
avctx->channel_layout = layout_mask;
qsort(layout->mChannelDescriptions, layout->mNumberChannelDescriptions,
sizeof(AudioChannelDescription), &ffat_compare_channel_descriptions);
for (i = 0; i < layout->mNumberChannelDescriptions; i++)
at->channel_map[i] = layout->mChannelDescriptions[i].mChannelFlags;
done:
av_free(layout);
}
if (!avctx->frame_size)
avctx->frame_size = 2048;
return 0;
}
static void put_descr(PutByteContext *pb, int tag, unsigned int size)
{
int i = 3;
bytestream2_put_byte(pb, tag);
for (; i > 0; i--)
bytestream2_put_byte(pb, (size >> (7 * i)) | 0x80);
bytestream2_put_byte(pb, size & 0x7F);
}
static uint8_t* ffat_get_magic_cookie(AVCodecContext *avctx, UInt32 *cookie_size)
{
ATDecodeContext *at = avctx->priv_data;
if (avctx->codec_id == AV_CODEC_ID_AAC) {
char *extradata;
PutByteContext pb;
*cookie_size = 5 + 3 + 5+13 + 5+at->extradata_size;
if (!(extradata = av_malloc(*cookie_size)))
return NULL;
bytestream2_init_writer(&pb, extradata, *cookie_size);
// ES descriptor
put_descr(&pb, 0x03, 3 + 5+13 + 5+at->extradata_size);
bytestream2_put_be16(&pb, 0);
bytestream2_put_byte(&pb, 0x00); // flags (= no flags)
// DecoderConfig descriptor
put_descr(&pb, 0x04, 13 + 5+at->extradata_size);
// Object type indication
bytestream2_put_byte(&pb, 0x40);
bytestream2_put_byte(&pb, 0x15); // flags (= Audiostream)
bytestream2_put_be24(&pb, 0); // Buffersize DB
bytestream2_put_be32(&pb, 0); // maxbitrate
bytestream2_put_be32(&pb, 0); // avgbitrate
// DecoderSpecific info descriptor
put_descr(&pb, 0x05, at->extradata_size);
bytestream2_put_buffer(&pb, at->extradata, at->extradata_size);
return extradata;
} else {
*cookie_size = at->extradata_size;
return at->extradata;
}
}
static av_cold int ffat_usable_extradata(AVCodecContext *avctx)
{
ATDecodeContext *at = avctx->priv_data;
return at->extradata_size &&
(avctx->codec_id == AV_CODEC_ID_ALAC ||
avctx->codec_id == AV_CODEC_ID_QDM2 ||
avctx->codec_id == AV_CODEC_ID_QDMC ||
avctx->codec_id == AV_CODEC_ID_AAC);
}
static int ffat_set_extradata(AVCodecContext *avctx)
{
ATDecodeContext *at = avctx->priv_data;
if (ffat_usable_extradata(avctx)) {
OSStatus status;
UInt32 cookie_size;
uint8_t *cookie = ffat_get_magic_cookie(avctx, &cookie_size);
if (!cookie)
return AVERROR(ENOMEM);
status = AudioConverterSetProperty(at->converter,
kAudioConverterDecompressionMagicCookie,
cookie_size, cookie);
if (status != 0)
av_log(avctx, AV_LOG_WARNING, "AudioToolbox cookie error: %i\n", (int)status);
if (cookie != at->extradata)
av_free(cookie);
}
return 0;
}
static av_cold int ffat_create_decoder(AVCodecContext *avctx, AVPacket *pkt)
{
ATDecodeContext *at = avctx->priv_data;
OSStatus status;
int i;
enum AVSampleFormat sample_fmt = (avctx->bits_per_raw_sample == 32) ?
AV_SAMPLE_FMT_S32 : AV_SAMPLE_FMT_S16;
AudioStreamBasicDescription in_format = {
.mFormatID = ffat_get_format_id(avctx->codec_id, avctx->profile),
.mBytesPerPacket = (avctx->codec_id == AV_CODEC_ID_ILBC) ? avctx->block_align : 0,
};
AudioStreamBasicDescription out_format = {
.mFormatID = kAudioFormatLinearPCM,
.mFormatFlags = kAudioFormatFlagIsSignedInteger | kAudioFormatFlagIsPacked,
.mFramesPerPacket = 1,
.mBitsPerChannel = av_get_bytes_per_sample(sample_fmt) * 8,
};
avctx->sample_fmt = sample_fmt;
if (ffat_usable_extradata(avctx)) {
UInt32 format_size = sizeof(in_format);
UInt32 cookie_size;
uint8_t *cookie = ffat_get_magic_cookie(avctx, &cookie_size);
if (!cookie)
return AVERROR(ENOMEM);
status = AudioFormatGetProperty(kAudioFormatProperty_FormatInfo,
cookie_size, cookie, &format_size, &in_format);
if (cookie != at->extradata)
av_free(cookie);
if (status != 0) {
av_log(avctx, AV_LOG_ERROR, "AudioToolbox header-parse error: %i\n", (int)status);
return AVERROR_UNKNOWN;
}
#if CONFIG_MP1_AT_DECODER || CONFIG_MP2_AT_DECODER || CONFIG_MP3_AT_DECODER
} else if (pkt && pkt->size >= 4 &&
(avctx->codec_id == AV_CODEC_ID_MP1 ||
avctx->codec_id == AV_CODEC_ID_MP2 ||
avctx->codec_id == AV_CODEC_ID_MP3)) {
enum AVCodecID codec_id;
int bit_rate;
if (ff_mpa_decode_header(AV_RB32(pkt->data), &avctx->sample_rate,
&in_format.mChannelsPerFrame, &avctx->frame_size,
&bit_rate, &codec_id) < 0)
return AVERROR_INVALIDDATA;
avctx->bit_rate = bit_rate;
in_format.mSampleRate = avctx->sample_rate;
#endif
#if CONFIG_AC3_AT_DECODER || CONFIG_EAC3_AT_DECODER
} else if (pkt && pkt->size >= 7 &&
(avctx->codec_id == AV_CODEC_ID_AC3 ||
avctx->codec_id == AV_CODEC_ID_EAC3)) {
AC3HeaderInfo hdr, *phdr = &hdr;
GetBitContext gbc;
init_get_bits(&gbc, pkt->data, pkt->size);
if (avpriv_ac3_parse_header(&gbc, &phdr) < 0)
return AVERROR_INVALIDDATA;
in_format.mSampleRate = hdr.sample_rate;
in_format.mChannelsPerFrame = hdr.channels;
avctx->frame_size = hdr.num_blocks * 256;
avctx->bit_rate = hdr.bit_rate;
#endif
} else {
in_format.mSampleRate = avctx->sample_rate ? avctx->sample_rate : 44100;
in_format.mChannelsPerFrame = avctx->channels ? avctx->channels : 1;
}
avctx->sample_rate = out_format.mSampleRate = in_format.mSampleRate;
avctx->channels = out_format.mChannelsPerFrame = in_format.mChannelsPerFrame;
if (avctx->codec_id == AV_CODEC_ID_ADPCM_IMA_QT)
in_format.mFramesPerPacket = 64;
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;
}
if ((status = ffat_set_extradata(avctx)) < 0)
return status;
for (i = 0; i < (sizeof(at->channel_map) / sizeof(at->channel_map[0])); i++)
at->channel_map[i] = i;
ffat_update_ctx(avctx);
if(!(at->decoded_data = av_malloc(av_get_bytes_per_sample(avctx->sample_fmt)
* avctx->frame_size * avctx->channels)))
return AVERROR(ENOMEM);
at->last_pts = AV_NOPTS_VALUE;
return 0;
}
static av_cold int ffat_init_decoder(AVCodecContext *avctx)
{
ATDecodeContext *at = avctx->priv_data;
at->extradata = avctx->extradata;
at->extradata_size = avctx->extradata_size;
if ((avctx->channels && avctx->sample_rate) || ffat_usable_extradata(avctx))
return ffat_create_decoder(avctx, NULL);
else
return 0;
}
static OSStatus ffat_decode_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_packet_unref(&at->in_pkt);
av_packet_move_ref(&at->in_pkt, &at->new_in_pkt);
if (!at->in_pkt.data) {
*nb_packets = 0;
return 1;
}
data->mNumberBuffers = 1;
data->mBuffers[0].mNumberChannels = 0;
data->mBuffers[0].mDataByteSize = at->in_pkt.size;
data->mBuffers[0].mData = at->in_pkt.data;
*nb_packets = 1;
if (packets) {
*packets = &at->pkt_desc;
at->pkt_desc.mDataByteSize = at->in_pkt.size;
}
return 0;
}
#define COPY_SAMPLES(type) \
type *in_ptr = (type*)at->decoded_data; \
type *end_ptr = in_ptr + frame->nb_samples * avctx->channels; \
type *out_ptr = (type*)frame->data[0]; \
for (; in_ptr < end_ptr; in_ptr += avctx->channels, out_ptr += avctx->channels) { \
int c; \
for (c = 0; c < avctx->channels; c++) \
out_ptr[c] = in_ptr[at->channel_map[c]]; \
}
static void ffat_copy_samples(AVCodecContext *avctx, AVFrame *frame)
{
ATDecodeContext *at = avctx->priv_data;
if (avctx->sample_fmt == AV_SAMPLE_FMT_S32) {
COPY_SAMPLES(int32_t);
} else {
COPY_SAMPLES(int16_t);
}
}
static int ffat_decode(AVCodecContext *avctx, void *data,
int *got_frame_ptr, AVPacket *avpkt)
{
ATDecodeContext *at = avctx->priv_data;
AVFrame *frame = data;
int pkt_size = avpkt->size;
AVPacket filtered_packet = {0};
OSStatus ret;
AudioBufferList out_buffers;
if (avctx->codec_id == AV_CODEC_ID_AAC && avpkt->size > 2 &&
(AV_RB16(avpkt->data) & 0xfff0) == 0xfff0) {
AVPacket filter_pkt = {0};
if (!at->bsf) {
const AVBitStreamFilter *bsf = av_bsf_get_by_name("aac_adtstoasc");
if(!bsf)
return AVERROR_BSF_NOT_FOUND;
if ((ret = av_bsf_alloc(bsf, &at->bsf)))
return ret;
if (((ret = avcodec_parameters_from_context(at->bsf->par_in, avctx)) < 0) ||
((ret = av_bsf_init(at->bsf)) < 0)) {
av_bsf_free(&at->bsf);
return ret;
}
}
if ((ret = av_packet_ref(&filter_pkt, avpkt)) < 0)
return ret;
if ((ret = av_bsf_send_packet(at->bsf, &filter_pkt)) < 0) {
av_packet_unref(&filter_pkt);
return ret;
}
if ((ret = av_bsf_receive_packet(at->bsf, &filtered_packet)) < 0)
return ret;
at->extradata = at->bsf->par_out->extradata;
at->extradata_size = at->bsf->par_out->extradata_size;
avpkt = &filtered_packet;
}
if (!at->converter) {
if ((ret = ffat_create_decoder(avctx, avpkt)) < 0) {
av_packet_unref(&filtered_packet);
return ret;
}
}
out_buffers = (AudioBufferList){
.mNumberBuffers = 1,
.mBuffers = {
{
.mNumberChannels = avctx->channels,
.mDataByteSize = av_get_bytes_per_sample(avctx->sample_fmt) * avctx->frame_size
* avctx->channels,
}
}
};
av_packet_unref(&at->new_in_pkt);
if (avpkt->size) {
if (filtered_packet.data) {
at->new_in_pkt = filtered_packet;
} else if ((ret = av_packet_ref(&at->new_in_pkt, avpkt)) < 0) {
return ret;
}
} else {
at->eof = 1;
}
frame->sample_rate = avctx->sample_rate;
frame->nb_samples = avctx->frame_size;
out_buffers.mBuffers[0].mData = at->decoded_data;
ret = AudioConverterFillComplexBuffer(at->converter, ffat_decode_callback, avctx,
&frame->nb_samples, &out_buffers, NULL);
if ((!ret || ret == 1) && frame->nb_samples) {
if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
return ret;
ffat_copy_samples(avctx, frame);
*got_frame_ptr = 1;
if (at->last_pts != AV_NOPTS_VALUE) {
frame->pkt_pts = at->last_pts;
at->last_pts = avpkt->pts;
}
} else if (ret && ret != 1) {
av_log(avctx, AV_LOG_WARNING, "Decode error: %i\n", ret);
} else {
at->last_pts = avpkt->pts;
}
return pkt_size;
}
static av_cold void ffat_decode_flush(AVCodecContext *avctx)
{
ATDecodeContext *at = avctx->priv_data;
AudioConverterReset(at->converter);
av_packet_unref(&at->new_in_pkt);
av_packet_unref(&at->in_pkt);
}
static av_cold int ffat_close_decoder(AVCodecContext *avctx)
{
ATDecodeContext *at = avctx->priv_data;
AudioConverterDispose(at->converter);
av_bsf_free(&at->bsf);
av_packet_unref(&at->new_in_pkt);
av_packet_unref(&at->in_pkt);
av_free(at->decoded_data);
return 0;
}
#define FFAT_DEC_CLASS(NAME) \
static const AVClass ffat_##NAME##_dec_class = { \
.class_name = "at_" #NAME "_dec", \
.version = LIBAVUTIL_VERSION_INT, \
};
#define FFAT_DEC(NAME, ID) \
FFAT_DEC_CLASS(NAME) \
AVCodec ff_##NAME##_at_decoder = { \
.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_decoder, \
.close = ffat_close_decoder, \
.decode = ffat_decode, \
.flush = ffat_decode_flush, \
.priv_class = &ffat_##NAME##_dec_class, \
.capabilities = AV_CODEC_CAP_DR1 | AV_CODEC_CAP_DELAY, \
.caps_internal = FF_CODEC_CAP_INIT_THREADSAFE, \
};
FFAT_DEC(aac, AV_CODEC_ID_AAC)
FFAT_DEC(ac3, AV_CODEC_ID_AC3)
FFAT_DEC(adpcm_ima_qt, AV_CODEC_ID_ADPCM_IMA_QT)
FFAT_DEC(alac, AV_CODEC_ID_ALAC)
FFAT_DEC(amr_nb, AV_CODEC_ID_AMR_NB)
FFAT_DEC(eac3, AV_CODEC_ID_EAC3)
FFAT_DEC(gsm_ms, AV_CODEC_ID_GSM_MS)
FFAT_DEC(ilbc, AV_CODEC_ID_ILBC)
FFAT_DEC(mp1, AV_CODEC_ID_MP1)
FFAT_DEC(mp2, AV_CODEC_ID_MP2)
FFAT_DEC(mp3, AV_CODEC_ID_MP3)
FFAT_DEC(pcm_alaw, AV_CODEC_ID_PCM_ALAW)
FFAT_DEC(pcm_mulaw, AV_CODEC_ID_PCM_MULAW)
FFAT_DEC(qdmc, AV_CODEC_ID_QDMC)
FFAT_DEC(qdm2, AV_CODEC_ID_QDM2)
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