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ffmpeg/libavcodec/mpeg4audio.h
Andreas Rheinhardt 49bf94536f avcodec/mpeg4audio: Unavpriv and deduplicate mpeg4audio_sample_rates 
avpriv_mpeg4audio_sample_rates has a size of 64B and it is currently
avpriv; a clone of it exists in aacenctab.h and from there it is inlined
in aacenc.c (which also uses the avpriv version) and in the FLV muxer.
This means that despite it being avpriv both libavformat as well as
libavcodec have copies already.

This situation is clearly suboptimal. Given the overhead of exporting
symbols (for x64 Elf/Linux/GNU: 2x2B version, 2x24B .dynsym, 24B .rela.dyn,
8B .got, 4B hash + twice the size of the name (here 31B)) the object is
unavprived, i.e. duplicated into libavformat when creating a shared
build; but the duplicates in the AAC encoder and FLV muxer are removed.

This involves splitting of the sample rate table into a file of its own;
this allowed to break some spurious dependencies (e.g. both the AAC
encoder as well as the Matroska demuxer actually don't need the
mpeg4audio_get_config stuff).

Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
2022-01-04 13:16:50 +01:00

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/*
* MPEG-4 Audio common header
* Copyright (c) 2008 Baptiste Coudurier <baptiste.coudurier@free.fr>
*
* 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
*/
#ifndef AVCODEC_MPEG4AUDIO_H
#define AVCODEC_MPEG4AUDIO_H
#include <stdint.h>
#include "libavutil/attributes.h"
#include "get_bits.h"
#include "put_bits.h"
typedef struct MPEG4AudioConfig {
int object_type;
int sampling_index;
int sample_rate;
int chan_config;
int sbr; ///< -1 implicit, 1 presence
int ext_object_type;
int ext_sampling_index;
int ext_sample_rate;
int ext_chan_config;
int channels;
int ps; ///< -1 implicit, 1 presence
int frame_length_short;
} MPEG4AudioConfig;
extern const int ff_mpeg4audio_sample_rates[16];
extern const uint8_t ff_mpeg4audio_channels[14];
/**
* Parse MPEG-4 systems extradata from a potentially unaligned GetBitContext to retrieve audio configuration.
* @param[in] c MPEG4AudioConfig structure to fill.
* @param[in] gb Extradata from container.
* @param[in] sync_extension look for a sync extension after config if true.
* @param[in] logctx opaque struct starting with an AVClass element, used for logging.
* @return negative AVERROR code on error, on success AudioSpecificConfig bit index in extradata.
*/
int ff_mpeg4audio_get_config_gb(MPEG4AudioConfig *c, GetBitContext *gb,
int sync_extension, void *logctx);
/**
* Parse MPEG-4 systems extradata from a raw buffer to retrieve audio configuration.
* @param[in] c MPEG4AudioConfig structure to fill.
* @param[in] buf Extradata from container.
* @param[in] size Extradata size in bytes.
* @param[in] sync_extension look for a sync extension after config if true.
* @param[in] logctx opaque struct starting with an AVClass element, used for logging.
* @return negative AVERROR code on error, AudioSpecificConfig bit index in extradata on success.
*/
int avpriv_mpeg4audio_get_config2(MPEG4AudioConfig *c, const uint8_t *buf,
int size, int sync_extension, void *logctx);
enum AudioObjectType {
AOT_NULL,
// Support? Name
AOT_AAC_MAIN, ///< Y Main
AOT_AAC_LC, ///< Y Low Complexity
AOT_AAC_SSR, ///< N (code in SoC repo) Scalable Sample Rate
AOT_AAC_LTP, ///< Y Long Term Prediction
AOT_SBR, ///< Y Spectral Band Replication
AOT_AAC_SCALABLE, ///< N Scalable
AOT_TWINVQ, ///< N Twin Vector Quantizer
AOT_CELP, ///< N Code Excited Linear Prediction
AOT_HVXC, ///< N Harmonic Vector eXcitation Coding
AOT_TTSI = 12, ///< N Text-To-Speech Interface
AOT_MAINSYNTH, ///< N Main Synthesis
AOT_WAVESYNTH, ///< N Wavetable Synthesis
AOT_MIDI, ///< N General MIDI
AOT_SAFX, ///< N Algorithmic Synthesis and Audio Effects
AOT_ER_AAC_LC, ///< N Error Resilient Low Complexity
AOT_ER_AAC_LTP = 19, ///< N Error Resilient Long Term Prediction
AOT_ER_AAC_SCALABLE, ///< N Error Resilient Scalable
AOT_ER_TWINVQ, ///< N Error Resilient Twin Vector Quantizer
AOT_ER_BSAC, ///< N Error Resilient Bit-Sliced Arithmetic Coding
AOT_ER_AAC_LD, ///< N Error Resilient Low Delay
AOT_ER_CELP, ///< N Error Resilient Code Excited Linear Prediction
AOT_ER_HVXC, ///< N Error Resilient Harmonic Vector eXcitation Coding
AOT_ER_HILN, ///< N Error Resilient Harmonic and Individual Lines plus Noise
AOT_ER_PARAM, ///< N Error Resilient Parametric
AOT_SSC, ///< N SinuSoidal Coding
AOT_PS, ///< N Parametric Stereo
AOT_SURROUND, ///< N MPEG Surround
AOT_ESCAPE, ///< Y Escape Value
AOT_L1, ///< Y Layer 1
AOT_L2, ///< Y Layer 2
AOT_L3, ///< Y Layer 3
AOT_DST, ///< N Direct Stream Transfer
AOT_ALS, ///< Y Audio LosslesS
AOT_SLS, ///< N Scalable LosslesS
AOT_SLS_NON_CORE, ///< N Scalable LosslesS (non core)
AOT_ER_AAC_ELD, ///< N Error Resilient Enhanced Low Delay
AOT_SMR_SIMPLE, ///< N Symbolic Music Representation Simple
AOT_SMR_MAIN, ///< N Symbolic Music Representation Main
AOT_USAC_NOSBR, ///< N Unified Speech and Audio Coding (no SBR)
AOT_SAOC, ///< N Spatial Audio Object Coding
AOT_LD_SURROUND, ///< N Low Delay MPEG Surround
AOT_USAC, ///< N Unified Speech and Audio Coding
};
#define MAX_PCE_SIZE 320 ///<Maximum size of a PCE including the 3-bit ID_PCE
///<marker and the comment
static av_always_inline unsigned int ff_pce_copy_bits(PutBitContext *pb,
GetBitContext *gb,
int bits)
{
unsigned int el = get_bits(gb, bits);
put_bits(pb, bits, el);
return el;
}
static inline int ff_copy_pce_data(PutBitContext *pb, GetBitContext *gb)
{
int five_bit_ch, four_bit_ch, comment_size, bits;
int offset = put_bits_count(pb);
ff_pce_copy_bits(pb, gb, 10); // Tag, Object Type, Frequency
five_bit_ch = ff_pce_copy_bits(pb, gb, 4); // Front
five_bit_ch += ff_pce_copy_bits(pb, gb, 4); // Side
five_bit_ch += ff_pce_copy_bits(pb, gb, 4); // Back
four_bit_ch = ff_pce_copy_bits(pb, gb, 2); // LFE
four_bit_ch += ff_pce_copy_bits(pb, gb, 3); // Data
five_bit_ch += ff_pce_copy_bits(pb, gb, 4); // Coupling
if (ff_pce_copy_bits(pb, gb, 1)) // Mono Mixdown
ff_pce_copy_bits(pb, gb, 4);
if (ff_pce_copy_bits(pb, gb, 1)) // Stereo Mixdown
ff_pce_copy_bits(pb, gb, 4);
if (ff_pce_copy_bits(pb, gb, 1)) // Matrix Mixdown
ff_pce_copy_bits(pb, gb, 3);
for (bits = five_bit_ch*5+four_bit_ch*4; bits > 16; bits -= 16)
ff_pce_copy_bits(pb, gb, 16);
if (bits)
ff_pce_copy_bits(pb, gb, bits);
align_put_bits(pb);
align_get_bits(gb);
comment_size = ff_pce_copy_bits(pb, gb, 8);
for (; comment_size > 0; comment_size--)
ff_pce_copy_bits(pb, gb, 8);
return put_bits_count(pb) - offset;
}
#endif /* AVCODEC_MPEG4AUDIO_H */
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