annotate audioconvert.c @ 12406:b25537518e40 libavcodec

aacenc: Rename Psy3gpp* structs to AacPsy* This allows cleaner implementation of other psymodels using the existing structs. It also will make it easier to interchange individual parts of the psymodel to create hybrid models. Patch by: Nathan Caldwell <[e-mail]>

/*
 * audio conversion
 * Copyright (c) 2006 Michael Niedermayer <[e-mail]>
 *
 * 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
 */

/**
 * @file
 * audio conversion
 * @author Michael Niedermayer <[e-mail]>
 */

#include "libavutil/avstring.h"
#include "libavutil/libm.h"
#include "avcodec.h"
#include "audioconvert.h"

typedef struct SampleFmtInfo {
    const char *name;
    int bits;
} SampleFmtInfo;

/** this table gives more information about formats */
static const SampleFmtInfo sample_fmt_info[SAMPLE_FMT_NB] = {
    [SAMPLE_FMT_U8]  = { .name = "u8",  .bits = 8 },
    [SAMPLE_FMT_S16] = { .name = "s16", .bits = 16 },
    [SAMPLE_FMT_S32] = { .name = "s32", .bits = 32 },
    [SAMPLE_FMT_FLT] = { .name = "flt", .bits = 32 },
    [SAMPLE_FMT_DBL] = { .name = "dbl", .bits = 64 },
};

const char *avcodec_get_sample_fmt_name(int sample_fmt)
{
    if (sample_fmt < 0 || sample_fmt >= SAMPLE_FMT_NB)
        return NULL;
    return sample_fmt_info[sample_fmt].name;
}

enum SampleFormat avcodec_get_sample_fmt(const char* name)
{
    int i;

    for (i=0; i < SAMPLE_FMT_NB; i++)
        if (!strcmp(sample_fmt_info[i].name, name))
            return i;
    return SAMPLE_FMT_NONE;
}

void avcodec_sample_fmt_string (char *buf, int buf_size, int sample_fmt)
{
    /* print header */
    if (sample_fmt < 0)
        snprintf (buf, buf_size, "name  " " depth");
    else if (sample_fmt < SAMPLE_FMT_NB) {
        SampleFmtInfo info= sample_fmt_info[sample_fmt];
        snprintf (buf, buf_size, "%-6s" "   %2d ", info.name, info.bits);
    }
}

static const char* const channel_names[]={
    "FL", "FR", "FC", "LFE", "BL",  "BR",  "FLC", "FRC",
    "BC", "SL", "SR", "TC",  "TFL", "TFC", "TFR", "TBL",
    "TBC", "TBR",
    [29] = "DL",
    [30] = "DR",
};

static const char *get_channel_name(int channel_id)
{
    if (channel_id<0 || channel_id>=FF_ARRAY_ELEMS(channel_names))
        return NULL;
    return channel_names[channel_id];
}

int64_t avcodec_guess_channel_layout(int nb_channels, enum CodecID codec_id, const char *fmt_name)
{
    switch(nb_channels) {
    case 1: return CH_LAYOUT_MONO;
    case 2: return CH_LAYOUT_STEREO;
    case 3: return CH_LAYOUT_SURROUND;
    case 4: return CH_LAYOUT_QUAD;
    case 5: return CH_LAYOUT_5POINT0;
    case 6: return CH_LAYOUT_5POINT1;
    case 8: return CH_LAYOUT_7POINT1;
    default: return 0;
    }
}

static const struct {
    const char *name;
    int         nb_channels;
    int64_t     layout;
} channel_layout_map[] = {
    { "mono",        1,  CH_LAYOUT_MONO },
    { "stereo",      2,  CH_LAYOUT_STEREO },
    { "4.0",         4,  CH_LAYOUT_4POINT0 },
    { "quad",        4,  CH_LAYOUT_QUAD },
    { "5.0",         5,  CH_LAYOUT_5POINT0 },
    { "5.0",         5,  CH_LAYOUT_5POINT0_BACK },
    { "5.1",         6,  CH_LAYOUT_5POINT1 },
    { "5.1",         6,  CH_LAYOUT_5POINT1_BACK },
    { "5.1+downmix", 8,  CH_LAYOUT_5POINT1|CH_LAYOUT_STEREO_DOWNMIX, },
    { "7.1",         8,  CH_LAYOUT_7POINT1 },
    { "7.1(wide)",   8,  CH_LAYOUT_7POINT1_WIDE },
    { "7.1+downmix", 10, CH_LAYOUT_7POINT1|CH_LAYOUT_STEREO_DOWNMIX, },
    { 0 }
};

void avcodec_get_channel_layout_string(char *buf, int buf_size, int nb_channels, int64_t channel_layout)
{
    int i;

    for (i=0; channel_layout_map[i].name; i++)
        if (nb_channels    == channel_layout_map[i].nb_channels &&
            channel_layout == channel_layout_map[i].layout) {
            av_strlcpy(buf, channel_layout_map[i].name, buf_size);
            return;
        }

    snprintf(buf, buf_size, "%d channels", nb_channels);
    if (channel_layout) {
        int i,ch;
        av_strlcat(buf, " (", buf_size);
        for(i=0,ch=0; i<64; i++) {
            if ((channel_layout & (1L<<i))) {
                const char *name = get_channel_name(i);
                if (name) {
                    if (ch>0) av_strlcat(buf, "|", buf_size);
                    av_strlcat(buf, name, buf_size);
                }
                ch++;
            }
        }
        av_strlcat(buf, ")", buf_size);
    }
}

int avcodec_channel_layout_num_channels(int64_t channel_layout)
{
    int count;
    uint64_t x = channel_layout;
    for (count = 0; x; count++)
        x &= x-1; // unset lowest set bit
    return count;
}

struct AVAudioConvert {
    int in_channels, out_channels;
    int fmt_pair;
};

AVAudioConvert *av_audio_convert_alloc(enum SampleFormat out_fmt, int out_channels,
                                       enum SampleFormat in_fmt, int in_channels,
                                       const float *matrix, int flags)
{
    AVAudioConvert *ctx;
    if (in_channels!=out_channels)
        return NULL;  /* FIXME: not supported */
    ctx = av_malloc(sizeof(AVAudioConvert));
    if (!ctx)
        return NULL;
    ctx->in_channels = in_channels;
    ctx->out_channels = out_channels;
    ctx->fmt_pair = out_fmt + SAMPLE_FMT_NB*in_fmt;
    return ctx;
}

void av_audio_convert_free(AVAudioConvert *ctx)
{
    av_free(ctx);
}

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)
{
    int ch;

    //FIXME optimize common cases

    for(ch=0; ch<ctx->out_channels; ch++){
        const int is=  in_stride[ch];
        const int os= out_stride[ch];
        const uint8_t *pi=  in[ch];
        uint8_t *po= out[ch];
        uint8_t *end= po + os*len;
        if(!out[ch])
            continue;

#define CONV(ofmt, otype, ifmt, expr)\
if(ctx->fmt_pair == ofmt + SAMPLE_FMT_NB*ifmt){\
    do{\
        *(otype*)po = expr; pi += is; po += os;\
    }while(po < end);\
}

//FIXME put things below under ifdefs so we do not waste space for cases no codec will need
//FIXME rounding ?

             CONV(SAMPLE_FMT_U8 , uint8_t, SAMPLE_FMT_U8 ,  *(const uint8_t*)pi)
        else CONV(SAMPLE_FMT_S16, int16_t, SAMPLE_FMT_U8 , (*(const uint8_t*)pi - 0x80)<<8)
        else CONV(SAMPLE_FMT_S32, int32_t, SAMPLE_FMT_U8 , (*(const uint8_t*)pi - 0x80)<<24)
        else CONV(SAMPLE_FMT_FLT, float  , SAMPLE_FMT_U8 , (*(const uint8_t*)pi - 0x80)*(1.0 / (1<<7)))
        else CONV(SAMPLE_FMT_DBL, double , SAMPLE_FMT_U8 , (*(const uint8_t*)pi - 0x80)*(1.0 / (1<<7)))
        else CONV(SAMPLE_FMT_U8 , uint8_t, SAMPLE_FMT_S16, (*(const int16_t*)pi>>8) + 0x80)
        else CONV(SAMPLE_FMT_S16, int16_t, SAMPLE_FMT_S16,  *(const int16_t*)pi)
        else CONV(SAMPLE_FMT_S32, int32_t, SAMPLE_FMT_S16,  *(const int16_t*)pi<<16)
        else CONV(SAMPLE_FMT_FLT, float  , SAMPLE_FMT_S16,  *(const int16_t*)pi*(1.0 / (1<<15)))
        else CONV(SAMPLE_FMT_DBL, double , SAMPLE_FMT_S16,  *(const int16_t*)pi*(1.0 / (1<<15)))
        else CONV(SAMPLE_FMT_U8 , uint8_t, SAMPLE_FMT_S32, (*(const int32_t*)pi>>24) + 0x80)
        else CONV(SAMPLE_FMT_S16, int16_t, SAMPLE_FMT_S32,  *(const int32_t*)pi>>16)
        else CONV(SAMPLE_FMT_S32, int32_t, SAMPLE_FMT_S32,  *(const int32_t*)pi)
        else CONV(SAMPLE_FMT_FLT, float  , SAMPLE_FMT_S32,  *(const int32_t*)pi*(1.0 / (1<<31)))
        else CONV(SAMPLE_FMT_DBL, double , SAMPLE_FMT_S32,  *(const int32_t*)pi*(1.0 / (1<<31)))
        else CONV(SAMPLE_FMT_U8 , uint8_t, SAMPLE_FMT_FLT, av_clip_uint8(  lrintf(*(const float*)pi * (1<<7)) + 0x80))
        else CONV(SAMPLE_FMT_S16, int16_t, SAMPLE_FMT_FLT, av_clip_int16(  lrintf(*(const float*)pi * (1<<15))))
        else CONV(SAMPLE_FMT_S32, int32_t, SAMPLE_FMT_FLT, av_clipl_int32(llrintf(*(const float*)pi * (1U<<31))))
        else CONV(SAMPLE_FMT_FLT, float  , SAMPLE_FMT_FLT, *(const float*)pi)
        else CONV(SAMPLE_FMT_DBL, double , SAMPLE_FMT_FLT, *(const float*)pi)
        else CONV(SAMPLE_FMT_U8 , uint8_t, SAMPLE_FMT_DBL, av_clip_uint8(  lrint(*(const double*)pi * (1<<7)) + 0x80))
        else CONV(SAMPLE_FMT_S16, int16_t, SAMPLE_FMT_DBL, av_clip_int16(  lrint(*(const double*)pi * (1<<15))))
        else CONV(SAMPLE_FMT_S32, int32_t, SAMPLE_FMT_DBL, av_clipl_int32(llrint(*(const double*)pi * (1U<<31))))
        else CONV(SAMPLE_FMT_FLT, float  , SAMPLE_FMT_DBL, *(const double*)pi)
        else CONV(SAMPLE_FMT_DBL, double , SAMPLE_FMT_DBL, *(const double*)pi)
        else return -1;
    }
    return 0;
}