get_audio.c

音频编码

read_samples_mp3(lame_global_flags * const gfp,
                 FILE * const musicin, short int mpg123pcm[2][1152], int stereo)
{
    int     out;
#if defined(AMIGA_MPEGA)  ||  defined(HAVE_MPGLIB)
    static const char type_name[] = "MP3 file";

    out =
        lame_decode_fromfile(musicin, mpg123pcm[0], mpg123pcm[1],
                             &mp3input_data);
    /*
     * out < 0:  error, probably EOF
     * out = 0:  not possible with lame_decode_fromfile() ???
     * out > 0:  number of output samples
     */
    if (out < 0) {
        memset(mpg123pcm, 0, sizeof(**mpg123pcm) * 2 * 1152);
        return 0;
    }

    if ( lame_get_num_channels( gfp ) != mp3input_data.stereo )
	if( silent < 10 ) {
	    fprintf(stderr,
		    "Error: number of channels has changed in %s - not supported\n",
		    type_name);
	}
    if ( lame_get_in_samplerate( gfp ) != mp3input_data.samplerate )
	if( silent < 10 ) {
	    fprintf(stderr,
		    "Error: sample frequency has changed in %s - not supported\n",
		    type_name);
	}

#else
    out = -1;
#endif
    return out;
}


int
WriteWaveHeader(FILE * const fp, const int pcmbytes,
                const int freq, const int channels, const int bits)
{
    int     bytes = (bits + 7) / 8;

    /* quick and dirty, but documented */
    fwrite("RIFF", 1, 4, fp); /* label */
    Write32BitsLowHigh(fp, pcmbytes + 44 - 8); /* length in bytes without header */
    fwrite("WAVEfmt ", 2, 4, fp); /* 2 labels */
    Write32BitsLowHigh(fp, 2 + 2 + 4 + 4 + 2 + 2); /* length of PCM format declaration area */
    Write16BitsLowHigh(fp, 1); /* is PCM? */
    Write16BitsLowHigh(fp, channels); /* number of channels */
    Write32BitsLowHigh(fp, freq); /* sample frequency in [Hz] */
    Write32BitsLowHigh(fp, freq * channels * bytes); /* bytes per second */
    Write16BitsLowHigh(fp, channels * bytes); /* bytes per sample time */
    Write16BitsLowHigh(fp, bits); /* bits per sample */
    fwrite("data", 1, 4, fp); /* label */
    Write32BitsLowHigh(fp, pcmbytes); /* length in bytes of raw PCM data */

    return ferror(fp) ? -1 : 0;
}




#if defined(LIBSNDFILE)

/*
** Copyright (C) 1999 Albert Faber
**
 * This library is free software; you can redistribute it and/or
 * modify it under the terms of the GNU Library General Public
 * License as published by the Free Software Foundation; either
 * version 2 of the License, or (at your option) any later version.
 *
 * This library 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
 * Library General Public License for more details.
 *
 * You should have received a copy of the GNU Library General Public
 * License along with this library; if not, write to the
 * Free Software Foundation, Inc., 59 Temple Place - Suite 330,
 * Boston, MA 02111-1307, USA.
 */






void
CloseSndFile(sound_file_format input, FILE * musicin)
{
    SNDFILE *gs_pSndFileIn = (SNDFILE *) musicin;
    if (input == sf_mp1 || input == sf_mp2 || input == sf_mp3) {
#ifndef AMIGA_MPEGA
        if (fclose(musicin) != 0) {
	    if( silent < 10 ) {
		fprintf(stderr, "Could not close audio input file\n");
	    }
            exit(2);
        }
#endif
    }
    else {
        if (gs_pSndFileIn) {
            if (sf_close(gs_pSndFileIn) != 0) {
		if( silent < 10 ) {
		    fprintf(stderr, "Could not close sound file \n");
		}
                exit(2);
            }
        }
    }
}



FILE   *
OpenSndFile(lame_global_flags * gfp, char *inPath)
{
    char   *lpszFileName = inPath;
    FILE   *musicin;
    SNDFILE *gs_pSndFileIn;
    SF_INFO gs_wfInfo;

    if (input_format == sf_mp1 ||
        input_format == sf_mp2 || input_format == sf_mp3) {
#ifdef AMIGA_MPEGA
        if (-1 == lame_decode_initfile(lpszFileName, &mp3input_data)) {
	    if( silent < 10 ) {
		fprintf(stderr, "Error reading headers in mp3 input file %s.\n",
			lpszFileName);
	    }
            exit(1);
        }
#endif
#ifdef HAVE_MPGLIB
        if ((musicin = fopen(lpszFileName, "rb")) == NULL) {
	    if( silent < 10 ) {
		fprintf(stderr, "Could not find \"%s\".\n", lpszFileName);
	    }
            exit(1);
        }
        if (-1 == lame_decode_initfile(musicin, &mp3input_data)) {
	    if( silent < 10 ) {
		fprintf(stderr, "Error reading headers in mp3 input file %s.\n",
			lpszFileName);
	    }
            exit(1);
        }
#endif

        if( -1 == lame_set_num_channels( gfp, mp3input_data.stereo ) ) {
	    if( silent < 10 ) {
		fprintf( stderr,
			 "Unsupported number of channels: %ud\n",
			 mp3input_data.stereo );
	    }
            exit( 1 );
        }
        (void) lame_set_in_samplerate( gfp, mp3input_data.samplerate );
        (void) lame_set_num_samples( gfp, mp3input_data.nsamp );
    }
    else if (input_format == sf_ogg) {
	if( silent < 10 ) {
	    fprintf(stderr, "sorry, vorbis support in LAME is deprecated.\n");
	}
        exit(1);
    }
    else {

        /* Try to open the sound file */
        /* set some defaults incase input is raw PCM */
        gs_wfInfo.seekable = (input_format != sf_raw); /* if user specified -r, set to not seekable */
        gs_wfInfo.samplerate = lame_get_in_samplerate( gfp );
        gs_wfInfo.pcmbitwidth = in_bitwidth;
        gs_wfInfo.channels = lame_get_num_channels( gfp );

	if (in_bitwidth == 8) {
	    if (in_signed)
		gs_wfInfo.format = SF_FORMAT_RAW_S8;
	    else
		gs_wfInfo.format = SF_FORMAT_RAW_U8;

	} else {
	    if (!in_signed) {
		fputs("Unsigned input only supported with bitwidth 8\n", stderr);
		exit(1);
	    }
	    if (in_endian != order_unknown) {
		if (in_endian == order_littleEndian)
		    gs_wfInfo.format = SF_FORMAT_RAW_LE;
		else
		    gs_wfInfo.format = SF_FORMAT_RAW_BE;
	    } else {
#ifndef WORDS_BIGENDIAN
		/* little endian */
		if (swapbytes)
		    gs_wfInfo.format = SF_FORMAT_RAW_BE;
		else
		    gs_wfInfo.format = SF_FORMAT_RAW_LE;
#else
		if (swapbytes)
		    gs_wfInfo.format = SF_FORMAT_RAW_LE;
		else
		    gs_wfInfo.format = SF_FORMAT_RAW_BE;
#endif
	    }
	}

        gs_pSndFileIn = sf_open_read(lpszFileName, &gs_wfInfo);
        musicin = (SNDFILE *) gs_pSndFileIn;

        /* Check result */
        if (gs_pSndFileIn == NULL) {
            sf_perror(gs_pSndFileIn);
	    if( silent < 10 ) {
		fprintf(stderr, "Could not open sound file \"%s\".\n",
			lpszFileName);
	    }
            exit(1);
        }

        if ((gs_wfInfo.format == SF_FORMAT_RAW_LE) ||
            (gs_wfInfo.format == SF_FORMAT_RAW_BE) ||
	    (gs_wfInfo.format == SF_FORMAT_RAW_S8) ||
	    (gs_wfInfo.format == SF_FORMAT_RAW_U8))
	    input_format = sf_raw;

#ifdef _DEBUG_SND_FILE
        DEBUGF("\n\nSF_INFO structure\n");
        DEBUGF("samplerate        :%d\n", gs_wfInfo.samplerate);
        DEBUGF("samples           :%d\n", gs_wfInfo.samples);
        DEBUGF("channels          :%d\n", gs_wfInfo.channels);
        DEBUGF("pcmbitwidth       :%d\n", gs_wfInfo.pcmbitwidth);
        DEBUGF("format            :");

        /* new formats from [e-mail]  1/2000 */

        switch (gs_wfInfo.format & SF_FORMAT_TYPEMASK) {
        case SF_FORMAT_WAV:
            DEBUGF("Microsoft WAV format (big endian). ");
            break;
        case SF_FORMAT_AIFF:
            DEBUGF("Apple/SGI AIFF format (little endian). ");
            break;
        case SF_FORMAT_AU:
            DEBUGF("Sun/NeXT AU format (big endian). ");
            break;
        case SF_FORMAT_AULE:
            DEBUGF("DEC AU format (little endian). ");
            break;
        case SF_FORMAT_RAW:
            DEBUGF("RAW PCM data. ");
            break;
        case SF_FORMAT_PAF:
            DEBUGF("Ensoniq PARIS file format. ");
            break;
        case SF_FORMAT_SVX:
            DEBUGF("Amiga IFF / SVX8 / SV16 format. ");
            break;
        case SF_FORMAT_NIST:
            DEBUGF("Sphere NIST format. ");
            break;
        default:
            assert(0);
            break;
        }

        switch (gs_wfInfo.format & SF_FORMAT_SUBMASK) {
        case SF_FORMAT_PCM:
            DEBUGF("PCM data in 8, 16, 24 or 32 bits.");
            break;
        case SF_FORMAT_FLOAT:
            DEBUGF("32 bit Intel x86 floats.");
            break;
        case SF_FORMAT_ULAW:
            DEBUGF("U-Law encoded.");
            break;
        case SF_FORMAT_ALAW:
            DEBUGF("A-Law encoded.");
            break;
        case SF_FORMAT_IMA_ADPCM:
            DEBUGF("IMA ADPCM.");
            break;
        case SF_FORMAT_MS_ADPCM:
            DEBUGF("Microsoft ADPCM.");
            break;
        case SF_FORMAT_PCM_BE:
            DEBUGF("Big endian PCM data.");
            break;
        case SF_FORMAT_PCM_LE:
            DEBUGF("Little endian PCM data.");
            break;
        case SF_FORMAT_PCM_S8:
            DEBUGF("Signed 8 bit PCM.");
            break;
        case SF_FORMAT_PCM_U8:
            DEBUGF("Unsigned 8 bit PCM.");
            break;
        case SF_FORMAT_SVX_FIB:
            DEBUGF("SVX Fibonacci Delta encoding.");
            break;
        case SF_FORMAT_SVX_EXP:
            DEBUGF("SVX Exponential Delta encoding.");
            break;
        default:
            assert(0);
            break;
        }

        DEBUGF("\n");
        DEBUGF("pcmbitwidth       :%d\n", gs_wfInfo.pcmbitwidth);
        DEBUGF("sections          :%d\n", gs_wfInfo.sections);
        DEBUGF("seekable          :\n", gs_wfInfo.seekable);
#endif

        (void) lame_set_num_samples( gfp, gs_wfInfo.samples );
        if( -1 == lame_set_num_channels( gfp, gs_wfInfo.channels ) ) {
	    if( silent < 10 ) {
		fprintf( stderr,
			 "Unsupported number of channels: %ud\n",
			 gs_wfInfo.channels );
	    }
            exit( 1 );
        }
        (void) lame_set_in_samplerate( gfp, gs_wfInfo.samplerate );
        pcmbitwidth = gs_wfInfo.pcmbitwidth;
    }

    if (lame_get_num_samples( gfp ) == MAX_U_32_NUM) {
        /* try to figure out num_samples */
        double  flen = lame_get_file_size( lpszFileName );

        if (flen >= 0) {
            /* try file size, assume 2 bytes per sample */
            if (input_format == sf_mp1 ||
                input_format == sf_mp2 || input_format == sf_mp3) {
		if (mp3input_data.bitrate>0) {
		    double  totalseconds =
			(flen * 8.0 / (1000.0 * mp3input_data.bitrate));
		    unsigned long tmp_num_samples =
			totalseconds * lame_get_in_samplerate( gfp );
		    
		    (void) lame_set_num_samples( gfp, tmp_num_samples );
		    mp3input_data.nsamp = tmp_num_samples;
		}
            }
            else {
                lame_set_num_samples( gfp,
                    flen / (2 * lame_get_num_channels( gfp )) );
            }
        }
    }


    return musicin;
}


/************************************************************************
*
* read_samples()
*
* PURPOSE:  reads the PCM samples from a file to the buffer
*
*  SEMANTICS:
* Reads #samples_read# number of shorts from #musicin# filepointer
* into #sample_buffer[]#.  Returns the number of samples read.
*
************************************************************************/

static int
read_samples_pcm(FILE * const musicin, int sample_buffer[2304],
                 int frame_size /* unused */ , int samples_to_read)
{
    int     i;
    int     samples_read;

    samples_read =
        sf_read_int((SNDFILE *) musicin, sample_buffer, samples_to_read);

    switch (pcmbitwidth) {
    case 8:
        for (i = 0; i < samples_read; i++)
            sample_buffer[i] <<= (8 * sizeof(int) - 8);
        break;
    case 16:
        for (i = 0; i < samples_read; i++)
            sample_buffer[i] <<= (8 * sizeof(int) - 16);
        break;
    case 24:
        for (i = 0; i < samples_read; i++)
            sample_buffer[i] <<= (8 * sizeof(int) - 24);
	break;
    case 32:
	break;
    default:
	if( silent < 10 ) {
	    fprintf(stderr, "Only 8, 16, 24 and 32 bit input files supported \n");