riff.c.svn-base

mediastreamer2是开源的网络传输媒体流的库

    put_tag(pb, tag);
    put_le32(pb, 0);
    return url_ftell(pb);
}

void end_tag(ByteIOContext *pb, offset_t start)
{
    offset_t pos;

    pos = url_ftell(pb);
    url_fseek(pb, start - 4, SEEK_SET);
    put_le32(pb, (uint32_t)(pos - start));
    url_fseek(pb, pos, SEEK_SET);
}

/* WAVEFORMATEX header */
/* returns the size or -1 on error */
int put_wav_header(ByteIOContext *pb, AVCodecContext *enc)
{
    int bps, blkalign, bytespersec;
    int hdrsize = 18;

    if(!enc->codec_tag || enc->codec_tag > 0xffff)
        return -1;

    put_le16(pb, enc->codec_tag);
    put_le16(pb, enc->channels);
    put_le32(pb, enc->sample_rate);
    if (enc->codec_id == CODEC_ID_PCM_U8 ||
        enc->codec_id == CODEC_ID_PCM_ALAW ||
        enc->codec_id == CODEC_ID_PCM_MULAW) {
        bps = 8;
    } else if (enc->codec_id == CODEC_ID_MP2 || enc->codec_id == CODEC_ID_MP3 || enc->codec_id == CODEC_ID_GSM_MS) {
        bps = 0;
    } else if (enc->codec_id == CODEC_ID_ADPCM_IMA_WAV || enc->codec_id == CODEC_ID_ADPCM_MS || enc->codec_id == CODEC_ID_ADPCM_G726 || enc->codec_id == CODEC_ID_ADPCM_YAMAHA) { //
        bps = 4;
    } else if (enc->codec_id == CODEC_ID_PCM_S24LE) {
        bps = 24;
    } else if (enc->codec_id == CODEC_ID_PCM_S32LE) {
        bps = 32;
    } else {
        bps = 16;
    }
    if(bps != enc->bits_per_sample && enc->bits_per_sample){
        av_log(enc, AV_LOG_WARNING, "requested bits_per_sample (%d) and actually stored (%d) differ\n", enc->bits_per_sample, bps);
    }

    if (enc->codec_id == CODEC_ID_MP2 || enc->codec_id == CODEC_ID_MP3 || enc->codec_id == CODEC_ID_GSM_MS) {
        blkalign = enc->frame_size; //this is wrong, but it seems many demuxers do not work if this is set correctly
        //blkalign = 144 * enc->bit_rate/enc->sample_rate;
    } else if (enc->codec_id == CODEC_ID_ADPCM_G726) { //
        blkalign = 1;
    } else if (enc->block_align != 0) { /* specified by the codec */
        blkalign = enc->block_align;
    } else
        blkalign = enc->channels*bps >> 3;
    if (enc->codec_id == CODEC_ID_PCM_U8 ||
        enc->codec_id == CODEC_ID_PCM_S24LE ||
        enc->codec_id == CODEC_ID_PCM_S32LE ||
        enc->codec_id == CODEC_ID_PCM_S16LE) {
        bytespersec = enc->sample_rate * blkalign;
    } else {
        bytespersec = enc->bit_rate / 8;
    }
    put_le32(pb, bytespersec); /* bytes per second */
    put_le16(pb, blkalign); /* block align */
    put_le16(pb, bps); /* bits per sample */
    if (enc->codec_id == CODEC_ID_MP3) {
        put_le16(pb, 12); /* wav_extra_size */
        hdrsize += 12;
        put_le16(pb, 1); /* wID */
        put_le32(pb, 2); /* fdwFlags */
        put_le16(pb, 1152); /* nBlockSize */
        put_le16(pb, 1); /* nFramesPerBlock */
        put_le16(pb, 1393); /* nCodecDelay */
    } else if (enc->codec_id == CODEC_ID_MP2) {
        put_le16(pb, 22); /* wav_extra_size */
        hdrsize += 22;
        put_le16(pb, 2);  /* fwHeadLayer */
        put_le32(pb, enc->bit_rate); /* dwHeadBitrate */
        put_le16(pb, enc->channels == 2 ? 1 : 8); /* fwHeadMode */
        put_le16(pb, 0);  /* fwHeadModeExt */
        put_le16(pb, 1);  /* wHeadEmphasis */
        put_le16(pb, 16); /* fwHeadFlags */
        put_le32(pb, 0);  /* dwPTSLow */
        put_le32(pb, 0);  /* dwPTSHigh */
    } else if (enc->codec_id == CODEC_ID_GSM_MS) {
        put_le16(pb, 2); /* wav_extra_size */
        hdrsize += 2;
        put_le16(pb, enc->frame_size); /* wSamplesPerBlock */
    } else if (enc->codec_id == CODEC_ID_ADPCM_IMA_WAV) {
        put_le16(pb, 2); /* wav_extra_size */
        hdrsize += 2;
        put_le16(pb, enc->frame_size); /* wSamplesPerBlock */
    } else if(enc->extradata_size){
        put_le16(pb, enc->extradata_size);
        put_buffer(pb, enc->extradata, enc->extradata_size);
        hdrsize += enc->extradata_size;
        if(hdrsize&1){
            hdrsize++;
            put_byte(pb, 0);
        }
    } else {
        hdrsize -= 2;
    }

    return hdrsize;
}

/* BITMAPINFOHEADER header */
void put_bmp_header(ByteIOContext *pb, AVCodecContext *enc, const AVCodecTag *tags, int for_asf)
{
    put_le32(pb, 40 + enc->extradata_size); /* size */
    put_le32(pb, enc->width);
    put_le32(pb, enc->height);
    put_le16(pb, 1); /* planes */

    put_le16(pb, enc->bits_per_sample ? enc->bits_per_sample : 24); /* depth */
    /* compression type */
    put_le32(pb, enc->codec_tag);
    put_le32(pb, enc->width * enc->height * 3);
    put_le32(pb, 0);
    put_le32(pb, 0);
    put_le32(pb, 0);
    put_le32(pb, 0);

    put_buffer(pb, enc->extradata, enc->extradata_size);

    if (enc->extradata_size & 1)
        put_byte(pb, 0);
}
#endif //CONFIG_MUXERS

#ifdef CONFIG_DEMUXERS
/* We could be given one of the three possible structures here:
 * WAVEFORMAT, PCMWAVEFORMAT or WAVEFORMATEX. Each structure
 * is an expansion of the previous one with the fields added
 * at the bottom. PCMWAVEFORMAT adds 'WORD wBitsPerSample' and
 * WAVEFORMATEX adds 'WORD  cbSize' and basically makes itself
 * an openended structure.
 */
void get_wav_header(ByteIOContext *pb, AVCodecContext *codec, int size)
{
    int id;

    id = get_le16(pb);
    codec->codec_type = CODEC_TYPE_AUDIO;
    codec->codec_tag = id;
    codec->channels = get_le16(pb);
    codec->sample_rate = get_le32(pb);
    codec->bit_rate = get_le32(pb) * 8;
    codec->block_align = get_le16(pb);
    if (size == 14) {  /* We're dealing with plain vanilla WAVEFORMAT */
        codec->bits_per_sample = 8;
    }else
        codec->bits_per_sample = get_le16(pb);
    if (size >= 18) {  /* We're obviously dealing with WAVEFORMATEX */
        int cbSize = get_le16(pb); /* cbSize */
        size -= 18;
        cbSize = FFMIN(size, cbSize);
        if (cbSize >= 22 && id == 0xfffe) { /* WAVEFORMATEXTENSIBLE */
            codec->bits_per_sample = get_le16(pb);
            get_le32(pb); /* dwChannelMask */
            id = get_le32(pb); /* 4 first bytes of GUID */
            url_fskip(pb, 12); /* skip end of GUID */
            cbSize -= 22;
            size -= 22;
        }
        codec->extradata_size = cbSize;
        if (cbSize > 0) {
            codec->extradata = av_mallocz(codec->extradata_size + FF_INPUT_BUFFER_PADDING_SIZE);
            get_buffer(pb, codec->extradata, codec->extradata_size);
            size -= cbSize;
        }

        /* It is possible for the chunk to contain garbage at the end */
        if (size > 0)
            url_fskip(pb, size);
    }
    codec->codec_id = wav_codec_get_id(id, codec->bits_per_sample);
}


int wav_codec_get_id(unsigned int tag, int bps)
{
    int id;
    id = codec_get_id(codec_wav_tags, tag);
    if (id <= 0)
        return id;
    /* handle specific u8 codec */
    if (id == CODEC_ID_PCM_S16LE && bps == 8)
        id = CODEC_ID_PCM_U8;
    if (id == CODEC_ID_PCM_S16LE && bps == 24)
        id = CODEC_ID_PCM_S24LE;
    if (id == CODEC_ID_PCM_S16LE && bps == 32)
        id = CODEC_ID_PCM_S32LE;
    if (id == CODEC_ID_ADPCM_IMA_WAV && bps == 8)
        id = CODEC_ID_PCM_ZORK;
    return id;
}
#endif // CONFIG_DEMUXERS

void ff_parse_specific_params(AVCodecContext *stream, int *au_rate, int *au_ssize, int *au_scale)
{
    int gcd;

    *au_ssize= stream->block_align;
    if(stream->frame_size && stream->sample_rate){
        *au_scale=stream->frame_size;
        *au_rate= stream->sample_rate;
    }else if(stream->codec_type == CODEC_TYPE_VIDEO){
        *au_scale= stream->time_base.num;
        *au_rate = stream->time_base.den;
    }else{
        *au_scale= stream->block_align ? stream->block_align*8 : 8;
        *au_rate = stream->bit_rate;
    }
    gcd= ff_gcd(*au_scale, *au_rate);
    *au_scale /= gcd;
    *au_rate /= gcd;
}