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FFmpeg/libavcodec/audiotoolboxdec.c
wm4 b945fed629 avcodec: add metadata to identify wrappers and hardware decoders 
Explicitly identify decoder/encoder wrappers with a common name. This
saves API users from guessing by the name suffix. For example, they
don't have to guess that "h264_qsv" is the h264 QSV implementation, and
instead they can just check the AVCodec .codec and .wrapper_name fields.

Explicitly mark AVCodec entries that are hardware decoders or most
likely hardware decoders with new AV_CODEC_CAPs. The purpose is allowing
API users listing hardware decoders in a more generic way. The proposed
AVCodecHWConfig does not provide this information fully, because it's
concerned with decoder configuration, not information about the fact
whether the hardware is used or not.

AV_CODEC_CAP_HYBRID exists specifically for QSV, which can have software
implementations in case the hardware is not capable.

Based on a patch by Philip Langdale <philipl@overt.org>.

Merges Libav commit 47687a2f8a.
2017-12-14 19:37:56 +01:00

618 lines
21 KiB
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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_internal.h"
#include "bytestream.h"
#include "internal.h"
#include "mpegaudiodecheader.h"
#include "libavutil/avassert.h"
#include "libavutil/opt.h"
#include "libavutil/log.h"
#if __MAC_OS_X_VERSION_MIN_REQUIRED < 101100
#define kAudioFormatEnhancedAC3 'ec-3'
#endif
typedef struct ATDecodeContext {
AVClass *av_class;
AudioConverterRef converter;
AudioStreamPacketDescription pkt_desc;
AVPacket in_pkt;
AVPacket new_in_pkt;
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;
GetBitContext gbc;
init_get_bits(&gbc, pkt->data, pkt->size);
if (ff_ac3_parse_header(&gbc, &hdr) < 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;
if (avctx->extradata_size) {
at->extradata = av_mallocz(avctx->extradata_size + AV_INPUT_BUFFER_PADDING_SIZE);
if (!at->extradata)
return AVERROR(ENOMEM);
at->extradata_size = avctx->extradata_size;
memcpy(at->extradata, avctx->extradata, 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;
OSStatus ret;
AudioBufferList out_buffers;
if (avctx->codec_id == AV_CODEC_ID_AAC) {
if (!at->extradata_size) {
uint8_t *side_data;
int side_data_size = 0;
side_data = av_packet_get_side_data(avpkt, AV_PKT_DATA_NEW_EXTRADATA,
&side_data_size);
if (side_data_size) {
at->extradata = av_mallocz(side_data_size + AV_INPUT_BUFFER_PADDING_SIZE);
if (!at->extradata)
return AVERROR(ENOMEM);
at->extradata_size = side_data_size;
memcpy(at->extradata, side_data, side_data_size);
}
}
}
if (!at->converter) {
if ((ret = ffat_create_decoder(avctx, avpkt)) < 0) {
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 ((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->pts = at->last_pts;
#if FF_API_PKT_PTS
FF_DISABLE_DEPRECATION_WARNINGS
frame->pkt_pts = at->last_pts;
FF_ENABLE_DEPRECATION_WARNINGS
#endif
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;
if (at->converter)
AudioConverterDispose(at->converter);
av_packet_unref(&at->new_in_pkt);
av_packet_unref(&at->in_pkt);
av_freep(&at->decoded_data);
av_freep(&at->extradata);
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, bsf_name) \
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, \
.bsfs = bsf_name, \
.capabilities = AV_CODEC_CAP_DR1 | AV_CODEC_CAP_DELAY, \
.caps_internal = FF_CODEC_CAP_INIT_THREADSAFE | FF_CODEC_CAP_INIT_CLEANUP, \
.wrapper_name = "at", \
};
FFAT_DEC(aac, AV_CODEC_ID_AAC, "aac_adtstoasc")
FFAT_DEC(ac3, AV_CODEC_ID_AC3, NULL)
FFAT_DEC(adpcm_ima_qt, AV_CODEC_ID_ADPCM_IMA_QT, NULL)
FFAT_DEC(alac, AV_CODEC_ID_ALAC, NULL)
FFAT_DEC(amr_nb, AV_CODEC_ID_AMR_NB, NULL)
FFAT_DEC(eac3, AV_CODEC_ID_EAC3, NULL)
FFAT_DEC(gsm_ms, AV_CODEC_ID_GSM_MS, NULL)
FFAT_DEC(ilbc, AV_CODEC_ID_ILBC, NULL)
FFAT_DEC(mp1, AV_CODEC_ID_MP1, NULL)
FFAT_DEC(mp2, AV_CODEC_ID_MP2, NULL)
FFAT_DEC(mp3, AV_CODEC_ID_MP3, NULL)
FFAT_DEC(pcm_alaw, AV_CODEC_ID_PCM_ALAW, NULL)
FFAT_DEC(pcm_mulaw, AV_CODEC_ID_PCM_MULAW, NULL)
FFAT_DEC(qdmc, AV_CODEC_ID_QDMC, NULL)
FFAT_DEC(qdm2, AV_CODEC_ID_QDM2, NULL)
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