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49bf94536f
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>
164 lines
8.1 KiB
C
164 lines
8.1 KiB
C
/*
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* MPEG-4 Audio common header
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* Copyright (c) 2008 Baptiste Coudurier <baptiste.coudurier@free.fr>
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*
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* This file is part of FFmpeg.
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*
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* FFmpeg is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Lesser General Public
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* License as published by the Free Software Foundation; either
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* version 2.1 of the License, or (at your option) any later version.
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*
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* FFmpeg is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Lesser General Public License for more details.
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*
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* You should have received a copy of the GNU Lesser General Public
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* License along with FFmpeg; if not, write to the Free Software
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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*/
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#ifndef AVCODEC_MPEG4AUDIO_H
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#define AVCODEC_MPEG4AUDIO_H
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#include <stdint.h>
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#include "libavutil/attributes.h"
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#include "get_bits.h"
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#include "put_bits.h"
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typedef struct MPEG4AudioConfig {
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int object_type;
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int sampling_index;
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int sample_rate;
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int chan_config;
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int sbr; ///< -1 implicit, 1 presence
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int ext_object_type;
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int ext_sampling_index;
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int ext_sample_rate;
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int ext_chan_config;
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int channels;
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int ps; ///< -1 implicit, 1 presence
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int frame_length_short;
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} MPEG4AudioConfig;
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extern const int ff_mpeg4audio_sample_rates[16];
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extern const uint8_t ff_mpeg4audio_channels[14];
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/**
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* Parse MPEG-4 systems extradata from a potentially unaligned GetBitContext to retrieve audio configuration.
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* @param[in] c MPEG4AudioConfig structure to fill.
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* @param[in] gb Extradata from container.
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* @param[in] sync_extension look for a sync extension after config if true.
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* @param[in] logctx opaque struct starting with an AVClass element, used for logging.
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* @return negative AVERROR code on error, on success AudioSpecificConfig bit index in extradata.
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*/
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int ff_mpeg4audio_get_config_gb(MPEG4AudioConfig *c, GetBitContext *gb,
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int sync_extension, void *logctx);
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/**
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* Parse MPEG-4 systems extradata from a raw buffer to retrieve audio configuration.
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* @param[in] c MPEG4AudioConfig structure to fill.
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* @param[in] buf Extradata from container.
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* @param[in] size Extradata size in bytes.
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* @param[in] sync_extension look for a sync extension after config if true.
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* @param[in] logctx opaque struct starting with an AVClass element, used for logging.
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* @return negative AVERROR code on error, AudioSpecificConfig bit index in extradata on success.
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*/
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int avpriv_mpeg4audio_get_config2(MPEG4AudioConfig *c, const uint8_t *buf,
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int size, int sync_extension, void *logctx);
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enum AudioObjectType {
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AOT_NULL,
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// Support? Name
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AOT_AAC_MAIN, ///< Y Main
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AOT_AAC_LC, ///< Y Low Complexity
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AOT_AAC_SSR, ///< N (code in SoC repo) Scalable Sample Rate
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AOT_AAC_LTP, ///< Y Long Term Prediction
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AOT_SBR, ///< Y Spectral Band Replication
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AOT_AAC_SCALABLE, ///< N Scalable
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AOT_TWINVQ, ///< N Twin Vector Quantizer
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AOT_CELP, ///< N Code Excited Linear Prediction
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AOT_HVXC, ///< N Harmonic Vector eXcitation Coding
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AOT_TTSI = 12, ///< N Text-To-Speech Interface
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AOT_MAINSYNTH, ///< N Main Synthesis
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AOT_WAVESYNTH, ///< N Wavetable Synthesis
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AOT_MIDI, ///< N General MIDI
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AOT_SAFX, ///< N Algorithmic Synthesis and Audio Effects
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AOT_ER_AAC_LC, ///< N Error Resilient Low Complexity
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AOT_ER_AAC_LTP = 19, ///< N Error Resilient Long Term Prediction
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AOT_ER_AAC_SCALABLE, ///< N Error Resilient Scalable
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AOT_ER_TWINVQ, ///< N Error Resilient Twin Vector Quantizer
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AOT_ER_BSAC, ///< N Error Resilient Bit-Sliced Arithmetic Coding
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AOT_ER_AAC_LD, ///< N Error Resilient Low Delay
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AOT_ER_CELP, ///< N Error Resilient Code Excited Linear Prediction
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AOT_ER_HVXC, ///< N Error Resilient Harmonic Vector eXcitation Coding
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AOT_ER_HILN, ///< N Error Resilient Harmonic and Individual Lines plus Noise
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AOT_ER_PARAM, ///< N Error Resilient Parametric
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AOT_SSC, ///< N SinuSoidal Coding
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AOT_PS, ///< N Parametric Stereo
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AOT_SURROUND, ///< N MPEG Surround
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AOT_ESCAPE, ///< Y Escape Value
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AOT_L1, ///< Y Layer 1
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AOT_L2, ///< Y Layer 2
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AOT_L3, ///< Y Layer 3
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AOT_DST, ///< N Direct Stream Transfer
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AOT_ALS, ///< Y Audio LosslesS
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AOT_SLS, ///< N Scalable LosslesS
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AOT_SLS_NON_CORE, ///< N Scalable LosslesS (non core)
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AOT_ER_AAC_ELD, ///< N Error Resilient Enhanced Low Delay
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AOT_SMR_SIMPLE, ///< N Symbolic Music Representation Simple
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AOT_SMR_MAIN, ///< N Symbolic Music Representation Main
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AOT_USAC_NOSBR, ///< N Unified Speech and Audio Coding (no SBR)
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AOT_SAOC, ///< N Spatial Audio Object Coding
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AOT_LD_SURROUND, ///< N Low Delay MPEG Surround
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AOT_USAC, ///< N Unified Speech and Audio Coding
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};
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#define MAX_PCE_SIZE 320 ///<Maximum size of a PCE including the 3-bit ID_PCE
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///<marker and the comment
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static av_always_inline unsigned int ff_pce_copy_bits(PutBitContext *pb,
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GetBitContext *gb,
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int bits)
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{
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unsigned int el = get_bits(gb, bits);
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put_bits(pb, bits, el);
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return el;
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}
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static inline int ff_copy_pce_data(PutBitContext *pb, GetBitContext *gb)
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{
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int five_bit_ch, four_bit_ch, comment_size, bits;
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int offset = put_bits_count(pb);
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ff_pce_copy_bits(pb, gb, 10); // Tag, Object Type, Frequency
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five_bit_ch = ff_pce_copy_bits(pb, gb, 4); // Front
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five_bit_ch += ff_pce_copy_bits(pb, gb, 4); // Side
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five_bit_ch += ff_pce_copy_bits(pb, gb, 4); // Back
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four_bit_ch = ff_pce_copy_bits(pb, gb, 2); // LFE
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four_bit_ch += ff_pce_copy_bits(pb, gb, 3); // Data
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five_bit_ch += ff_pce_copy_bits(pb, gb, 4); // Coupling
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if (ff_pce_copy_bits(pb, gb, 1)) // Mono Mixdown
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ff_pce_copy_bits(pb, gb, 4);
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if (ff_pce_copy_bits(pb, gb, 1)) // Stereo Mixdown
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ff_pce_copy_bits(pb, gb, 4);
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if (ff_pce_copy_bits(pb, gb, 1)) // Matrix Mixdown
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ff_pce_copy_bits(pb, gb, 3);
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for (bits = five_bit_ch*5+four_bit_ch*4; bits > 16; bits -= 16)
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ff_pce_copy_bits(pb, gb, 16);
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if (bits)
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ff_pce_copy_bits(pb, gb, bits);
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align_put_bits(pb);
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align_get_bits(gb);
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comment_size = ff_pce_copy_bits(pb, gb, 8);
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for (; comment_size > 0; comment_size--)
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ff_pce_copy_bits(pb, gb, 8);
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return put_bits_count(pb) - offset;
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}
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#endif /* AVCODEC_MPEG4AUDIO_H */
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