annotate audioconvert.h @ 10874:bcfe2acbf190 libavcodec

AAC: Compress codebook tables and optimise sign bit handling The codebooks each consist of small number of values repeated in groups of 2 or 4. Storing the codebooks as a packed list of 2- or 4-bit indexes into a table reduces their size substantially (from 7.5k to 1.5k), resulting in less cache pressure. For the band types with sign bits in the bitstream, storing the number and position of non-zero codebook values using a few bits avoids multiple get_bits() calls and floating-point comparisons which gcc handles miserably. Some float/int type punning also avoids gcc brain damage. Overall speedup 20-35% on Cortex-A8, 20% on Core i7.

/*
 * audio conversion
 * Copyright (c) 2006 Michael Niedermayer <[e-mail]>
 * Copyright (c) 2008 Peter Ross
 *
 * 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_AUDIOCONVERT_H
#define AVCODEC_AUDIOCONVERT_H

/**
 * @file libavcodec/audioconvert.h
 * Audio format conversion routines
 */


#include "avcodec.h"


/**
 * Generate string corresponding to the sample format with
 * number sample_fmt, or a header if sample_fmt is negative.
 *
 * @param[in] buf the buffer where to write the string
 * @param[in] buf_size the size of buf
 * @param[in] sample_fmt the number of the sample format to print the corresponding info string, or
 * a negative value to print the corresponding header.
 * Meaningful values for obtaining a sample format info vary from 0 to SAMPLE_FMT_NB -1.
 */
void avcodec_sample_fmt_string(char *buf, int buf_size, int sample_fmt);

/**
 * @return NULL on error
 */
const char *avcodec_get_sample_fmt_name(int sample_fmt);

/**
 * @return SAMPLE_FMT_NONE on error
 */
enum SampleFormat avcodec_get_sample_fmt(const char* name);

/**
 * @return NULL on error
 */
const char *avcodec_get_channel_name(int channel_id);

/**
 * Return description of channel layout
 */
void avcodec_get_channel_layout_string(char *buf, int buf_size, int nb_channels, int64_t channel_layout);

/**
 * Guess the channel layout
 * @param nb_channels
 * @param codec_id Codec identifier, or CODEC_ID_NONE if unknown
 * @param fmt_name Format name, or NULL if unknown
 * @return Channel layout mask
 */
int64_t avcodec_guess_channel_layout(int nb_channels, enum CodecID codec_id, const char *fmt_name);

/**
 * @return the number of channels in the channel layout.
 */
int avcodec_channel_layout_num_channels(int64_t channel_layout);

struct AVAudioConvert;
typedef struct AVAudioConvert AVAudioConvert;

/**
 * Create an audio sample format converter context
 * @param out_fmt Output sample format
 * @param out_channels Number of output channels
 * @param in_fmt Input sample format
 * @param in_channels Number of input channels
 * @param[in] matrix Channel mixing matrix (of dimension in_channel*out_channels). Set to NULL to ignore.
 * @param flags See FF_MM_xx
 * @return NULL on error
 */
AVAudioConvert *av_audio_convert_alloc(enum SampleFormat out_fmt, int out_channels,
                                       enum SampleFormat in_fmt, int in_channels,
                                       const float *matrix, int flags);

/**
 * Free audio sample format converter context
 */
void av_audio_convert_free(AVAudioConvert *ctx);

/**
 * Convert between audio sample formats
 * @param[in] out array of output buffers for each channel. set to NULL to ignore processing of the given channel.
 * @param[in] out_stride distance between consecutive input samples (measured in bytes)
 * @param[in] in array of input buffers for each channel
 * @param[in] in_stride distance between consecutive output samples (measured in bytes)
 * @param len length of audio frame size (measured in samples)
 */
int av_audio_convert(AVAudioConvert *ctx,
                           void * const out[6], const int out_stride[6],
                     const void * const  in[6], const int  in_stride[6], int len);

#endif /* AVCODEC_AUDIOCONVERT_H */