예제 #1
0
파일: wav.c 프로젝트: Akuaksh/FFmpeg-alsenc
/* wav input */
static int wav_read_header(AVFormatContext *s,
                           AVFormatParameters *ap)
{
    int64_t size, av_uninit(data_size);
    int rf64;
    unsigned int tag;
    ByteIOContext *pb = s->pb;
    AVStream *st;
    WAVContext *wav = s->priv_data;

    /* check RIFF header */
    tag = get_le32(pb);

    rf64 = tag == MKTAG('R', 'F', '6', '4');
    if (!rf64 && tag != MKTAG('R', 'I', 'F', 'F'))
        return -1;
    get_le32(pb); /* file size */
    tag = get_le32(pb);
    if (tag != MKTAG('W', 'A', 'V', 'E'))
        return -1;

    if (rf64) {
        if (get_le32(pb) != MKTAG('d', 's', '6', '4'))
            return -1;
        size = get_le32(pb);
        if (size < 16)
            return -1;
        get_le64(pb); /* RIFF size */
        data_size = get_le64(pb);
        url_fskip(pb, size - 16); /* skip rest of ds64 chunk */
    }

    /* parse fmt header */
    size = find_tag(pb, MKTAG('f', 'm', 't', ' '));
    if (size < 0)
        return -1;
    st = av_new_stream(s, 0);
    if (!st)
        return AVERROR(ENOMEM);

    ff_get_wav_header(pb, st->codec, size);
    st->need_parsing = AVSTREAM_PARSE_FULL;

    av_set_pts_info(st, 64, 1, st->codec->sample_rate);

    size = find_tag(pb, MKTAG('d', 'a', 't', 'a'));
    if (rf64)
        size = data_size;
    if (size < 0)
        return -1;
    if (!size) {
        wav->data_end = INT64_MAX;
    } else
        wav->data_end= url_ftell(pb) + size;
    return 0;
}
예제 #2
0
파일: wavdec.c 프로젝트: Rodeo314/tim-libav
static int w64_read_header(AVFormatContext *s)
{
    int64_t size;
    AVIOContext *pb      = s->pb;
    WAVDemuxContext *wav = s->priv_data;
    AVStream *st;
    uint8_t guid[16];
    int ret;

    avio_read(pb, guid, 16);
    if (memcmp(guid, guid_riff, 16))
        return AVERROR_INVALIDDATA;

    /* riff + wave + fmt + sizes */
    if (avio_rl64(pb) < 16 + 8 + 16 + 8 + 16 + 8)
        return AVERROR_INVALIDDATA;

    avio_read(pb, guid, 16);
    if (memcmp(guid, guid_wave, 16)) {
        av_log(s, AV_LOG_ERROR, "could not find wave guid\n");
        return AVERROR_INVALIDDATA;
    }

    size = find_guid(pb, guid_fmt);
    if (size < 0) {
        av_log(s, AV_LOG_ERROR, "could not find fmt guid\n");
        return AVERROR_INVALIDDATA;
    }

    st = avformat_new_stream(s, NULL);
    if (!st)
        return AVERROR(ENOMEM);

    /* subtract chunk header size - normal wav file doesn't count it */
    ret = ff_get_wav_header(s, pb, st->codecpar, size - 24);
    if (ret < 0)
        return ret;
    avio_skip(pb, FFALIGN(size, INT64_C(8)) - size);

    st->need_parsing = AVSTREAM_PARSE_FULL;

    avpriv_set_pts_info(st, 64, 1, st->codecpar->sample_rate);

    size = find_guid(pb, guid_data);
    if (size < 0) {
        av_log(s, AV_LOG_ERROR, "could not find data guid\n");
        return AVERROR_INVALIDDATA;
    }
    wav->data_end = avio_tell(pb) + size - 24;
    wav->w64      = 1;

    return 0;
}
예제 #3
0
파일: wav.c 프로젝트: Akuaksh/FFmpeg-alsenc
static int w64_read_header(AVFormatContext *s, AVFormatParameters *ap)
{
    int64_t size;
    ByteIOContext *pb  = s->pb;
    WAVContext    *wav = s->priv_data;
    AVStream *st;
    uint8_t guid[16];

    get_buffer(pb, guid, 16);
    if (memcmp(guid, guid_riff, 16))
        return -1;

    if (get_le64(pb) < 16 + 8 + 16 + 8 + 16 + 8) /* riff + wave + fmt + sizes */
        return -1;

    get_buffer(pb, guid, 16);
    if (memcmp(guid, guid_wave, 16)) {
        av_log(s, AV_LOG_ERROR, "could not find wave guid\n");
        return -1;
    }

    size = find_guid(pb, guid_fmt);
    if (size < 0) {
        av_log(s, AV_LOG_ERROR, "could not find fmt guid\n");
        return -1;
    }

    st = av_new_stream(s, 0);
    if (!st)
        return AVERROR(ENOMEM);

    /* subtract chunk header size - normal wav file doesn't count it */
    ff_get_wav_header(pb, st->codec, size - 24);
    url_fskip(pb, FFALIGN(size, INT64_C(8)) - size);

    st->need_parsing = AVSTREAM_PARSE_FULL;

    av_set_pts_info(st, 64, 1, st->codec->sample_rate);

    size = find_guid(pb, guid_data);
    if (size < 0) {
        av_log(s, AV_LOG_ERROR, "could not find data guid\n");
        return -1;
    }
    wav->data_end = url_ftell(pb) + size - 24;
    wav->w64      = 1;

    return 0;
}
static int read_header(AVFormatContext *s,
                           AVFormatParameters *ap)
{
    ACTContext* ctx = s->priv_data;
    AVIOContext *pb = s->pb;
    int size;
    AVStream* st;

    int min,sec,msec;

    st = avformat_new_stream(s, NULL);
    if (!st)
        return AVERROR(ENOMEM);

    avio_skip(pb, 16);
    size=avio_rl32(pb);
    ff_get_wav_header(pb, st->codec, size);

    /*
      8000Hz (Fine-rec) file format has 10 bytes long
      packets with 10ms of sound data in them
    */
    if (st->codec->sample_rate != 8000) {
        av_log(s, AV_LOG_ERROR, "Sample rate %d is not supported.\n", st->codec->sample_rate);
        return AVERROR_INVALIDDATA;
    }

    st->codec->frame_size=80;
    st->codec->channels=1;
    av_set_pts_info(st, 64, 1, 100);

    st->codec->codec_id=CODEC_ID_G729;

    avio_seek(pb, 257, SEEK_SET);
    msec=avio_rl16(pb);
    sec=avio_r8(pb);
    min=avio_rl32(pb);

    st->duration = av_rescale(1000*(min*60+sec)+msec, st->codec->sample_rate, 1000 * st->codec->frame_size);

    ctx->bytes_left_in_chunk=CHUNK_SIZE;

    avio_seek(pb, 512, SEEK_SET);

    return 0;
}
예제 #5
0
static int wav_parse_fmt_tag(AVFormatContext *s, int64_t size, AVStream **st)
{
    AVIOContext *pb = s->pb;
    int ret;

    /* parse fmt header */
    *st = av_new_stream(s, 0);
    if (!*st)
        return AVERROR(ENOMEM);

    ret = ff_get_wav_header(pb, (*st)->codec, size);
    if (ret < 0)
        return ret;
    (*st)->need_parsing = AVSTREAM_PARSE_FULL;

    av_set_pts_info(*st, 64, 1, (*st)->codec->sample_rate);

    return 0;
}
예제 #6
0
파일: wav.c 프로젝트: mrtos/Logitech-Revue
/* wav input */
static int wav_read_header(AVFormatContext *s,
                           AVFormatParameters *ap)
{
    int64_t size;
    unsigned int tag;
    ByteIOContext *pb = s->pb;
    AVStream *st;
    WAVContext *wav = s->priv_data;

    /* check RIFF header */
    tag = get_le32(pb);

    if (tag != MKTAG('R', 'I', 'F', 'F'))
        return -1;
    get_le32(pb); /* file size */
    tag = get_le32(pb);
    if (tag != MKTAG('W', 'A', 'V', 'E'))
        return -1;

    /* parse fmt header */
    size = find_tag(pb, MKTAG('f', 'm', 't', ' '));
    if (size < 0)
        return -1;
    st = av_new_stream(s, 0);
    if (!st)
        return AVERROR(ENOMEM);

    ff_get_wav_header(pb, st->codec, size);
    st->need_parsing = AVSTREAM_PARSE_FULL;

    av_set_pts_info(st, 64, 1, st->codec->sample_rate);

    size = find_tag(pb, MKTAG('d', 'a', 't', 'a'));
    if (size < 0)
        return -1;
    wav->data_end= url_ftell(pb) + size;
    return 0;
}
예제 #7
0
static int dxa_read_header(AVFormatContext *s, AVFormatParameters *ap)
{
    AVIOContext *pb = s->pb;
    DXAContext *c = s->priv_data;
    AVStream *st, *ast;
    uint32_t tag;
    int32_t fps;
    int w, h;
    int num, den;
    int flags;

    tag = avio_rl32(pb);
    if (tag != MKTAG('D', 'E', 'X', 'A'))
        return -1;
    flags = avio_r8(pb);
    c->frames = avio_rb16(pb);
    if(!c->frames){
        av_log(s, AV_LOG_ERROR, "File contains no frames ???\n");
        return -1;
    }

    fps = avio_rb32(pb);
    if(fps > 0){
        den = 1000;
        num = fps;
    }else if (fps < 0){
        den = 100000;
        num = -fps;
    }else{
        den = 10;
        num = 1;
    }
    w = avio_rb16(pb);
    h = avio_rb16(pb);
    c->has_sound = 0;

    st = av_new_stream(s, 0);
    if (!st)
        return -1;

    // Parse WAV data header
    if(avio_rl32(pb) == MKTAG('W', 'A', 'V', 'E')){
        uint32_t size, fsize;
        c->has_sound = 1;
        size = avio_rb32(pb);
        c->vidpos = avio_tell(pb) + size;
        avio_seek(pb, 16, SEEK_CUR);
        fsize = avio_rl32(pb);

        ast = av_new_stream(s, 0);
        if (!ast)
            return -1;
        ff_get_wav_header(pb, ast->codec, fsize);
        // find 'data' chunk
        while(avio_tell(pb) < c->vidpos && !url_feof(pb)){
            tag = avio_rl32(pb);
            fsize = avio_rl32(pb);
            if(tag == MKTAG('d', 'a', 't', 'a')) break;
            avio_seek(pb, fsize, SEEK_CUR);
        }
        c->bpc = (fsize + c->frames - 1) / c->frames;
        if(ast->codec->block_align)
            c->bpc = ((c->bpc + ast->codec->block_align - 1) / ast->codec->block_align) * ast->codec->block_align;
        c->bytes_left = fsize;
        c->wavpos = avio_tell(pb);
        avio_seek(pb, c->vidpos, SEEK_SET);
    }

    /* now we are ready: build format streams */
    st->codec->codec_type = AVMEDIA_TYPE_VIDEO;
    st->codec->codec_id   = CODEC_ID_DXA;
    st->codec->width      = w;
    st->codec->height     = h;
    av_reduce(&den, &num, den, num, (1UL<<31)-1);
    av_set_pts_info(st, 33, num, den);
    /* flags & 0x80 means that image is interlaced,
     * flags & 0x40 means that image has double height
     * either way set true height
     */
    if(flags & 0xC0){
        st->codec->height >>= 1;
    }
예제 #8
0
파일: xwma.c 프로젝트: markjreed/vice-emu
static int xwma_read_header(AVFormatContext *s)
{
    int64_t size;
    int ret;
    uint32_t dpds_table_size = 0;
    uint32_t *dpds_table = NULL;
    unsigned int tag;
    AVIOContext *pb = s->pb;
    AVStream *st;
    XWMAContext *xwma = s->priv_data;
    int i;

    /* The following code is mostly copied from wav.c, with some
     * minor alterations.
     */

    /* check RIFF header */
    tag = avio_rl32(pb);
    if (tag != MKTAG('R', 'I', 'F', 'F'))
        return -1;
    avio_rl32(pb); /* file size */
    tag = avio_rl32(pb);
    if (tag != MKTAG('X', 'W', 'M', 'A'))
        return -1;

    /* parse fmt header */
    tag = avio_rl32(pb);
    if (tag != MKTAG('f', 'm', 't', ' '))
        return -1;
    size = avio_rl32(pb);
    st = avformat_new_stream(s, NULL);
    if (!st)
        return AVERROR(ENOMEM);

    ret = ff_get_wav_header(pb, st->codec, size);
    if (ret < 0)
        return ret;
    st->need_parsing = AVSTREAM_PARSE_NONE;

    /* All xWMA files I have seen contained WMAv2 data. If there are files
     * using WMA Pro or some other codec, then we need to figure out the right
     * extradata for that. Thus, ask the user for feedback, but try to go on
     * anyway.
     */
    if (st->codec->codec_id != AV_CODEC_ID_WMAV2) {
        avpriv_request_sample(s, "Unexpected codec (tag 0x04%x; id %d)",
                              st->codec->codec_tag, st->codec->codec_id);
    } else {
        /* In all xWMA files I have seen, there is no extradata. But the WMA
         * codecs require extradata, so we provide our own fake extradata.
         *
         * First, check that there really was no extradata in the header. If
         * there was, then try to use it, after asking the user to provide a
         * sample of this unusual file.
         */
        if (st->codec->extradata_size != 0) {
            /* Surprise, surprise: We *did* get some extradata. No idea
             * if it will work, but just go on and try it, after asking
             * the user for a sample.
             */
            avpriv_request_sample(s, "Unexpected extradata (%d bytes)",
                                  st->codec->extradata_size);
        } else {
            st->codec->extradata_size = 6;
            st->codec->extradata      = av_mallocz(6 + FF_INPUT_BUFFER_PADDING_SIZE);
            if (!st->codec->extradata)
                return AVERROR(ENOMEM);

            /* setup extradata with our experimentally obtained value */
            st->codec->extradata[4] = 31;
        }
    }

    if (!st->codec->channels) {
        av_log(s, AV_LOG_WARNING, "Invalid channel count: %d\n",
               st->codec->channels);
        return AVERROR_INVALIDDATA;
    }
    if (!st->codec->bits_per_coded_sample) {
        av_log(s, AV_LOG_WARNING, "Invalid bits_per_coded_sample: %d\n",
               st->codec->bits_per_coded_sample);
        return AVERROR_INVALIDDATA;
    }

    /* set the sample rate */
    avpriv_set_pts_info(st, 64, 1, st->codec->sample_rate);

    /* parse the remaining RIFF chunks */
    for (;;) {
        if (pb->eof_reached) {
            ret = AVERROR_EOF;
            goto end;
        }
        /* read next chunk tag */
        tag = avio_rl32(pb);
        size = avio_rl32(pb);
        if (tag == MKTAG('d', 'a', 't', 'a')) {
            /* We assume that the data chunk comes last. */
            break;
        } else if (tag == MKTAG('d','p','d','s')) {
            /* Quoting the MSDN xWMA docs on the dpds chunk: "Contains the
             * decoded packet cumulative data size array, each element is the
             * number of bytes accumulated after the corresponding xWMA packet
             * is decoded in order."
             *
             * Each packet has size equal to st->codec->block_align, which in
             * all cases I saw so far was always 2230. Thus, we can use the
             * dpds data to compute a seeking index.
             */

            /* Error out if there is more than one dpds chunk. */
            if (dpds_table) {
                av_log(s, AV_LOG_ERROR, "two dpds chunks present\n");
                ret = AVERROR_INVALIDDATA;
                goto end;
            }

            /* Compute the number of entries in the dpds chunk. */
            if (size & 3) {  /* Size should be divisible by four */
                av_log(s, AV_LOG_WARNING,
                       "dpds chunk size %"PRId64" not divisible by 4\n", size);
            }
            dpds_table_size = size / 4;
            if (dpds_table_size == 0 || dpds_table_size >= INT_MAX / 4) {
                av_log(s, AV_LOG_ERROR,
                       "dpds chunk size %"PRId64" invalid\n", size);
                return AVERROR_INVALIDDATA;
            }

            /* Allocate some temporary storage to keep the dpds data around.
             * for processing later on.
             */
            dpds_table = av_malloc(dpds_table_size * sizeof(uint32_t));
            if (!dpds_table) {
                return AVERROR(ENOMEM);
            }

            for (i = 0; i < dpds_table_size; ++i) {
                dpds_table[i] = avio_rl32(pb);
                size -= 4;
            }
        }
        avio_skip(pb, size);
    }

    /* Determine overall data length */
    if (size < 0) {
        ret = AVERROR_INVALIDDATA;
        goto end;
    }
    if (!size) {
        xwma->data_end = INT64_MAX;
    } else
        xwma->data_end = avio_tell(pb) + size;


    if (dpds_table && dpds_table_size) {
        int64_t cur_pos;
        const uint32_t bytes_per_sample
                = (st->codec->channels * st->codec->bits_per_coded_sample) >> 3;

        /* Estimate the duration from the total number of output bytes. */
        const uint64_t total_decoded_bytes = dpds_table[dpds_table_size - 1];

        if (!bytes_per_sample) {
            av_log(s, AV_LOG_ERROR,
                   "Invalid bits_per_coded_sample %d for %d channels\n",
                   st->codec->bits_per_coded_sample, st->codec->channels);
            ret = AVERROR_INVALIDDATA;
            goto end;
        }

        st->duration = total_decoded_bytes / bytes_per_sample;

        /* Use the dpds data to build a seek table.  We can only do this after
         * we know the offset to the data chunk, as we need that to determine
         * the actual offset to each input block.
         * Note: If we allowed ourselves to assume that the data chunk always
         * follows immediately after the dpds block, we could of course guess
         * the data block's start offset already while reading the dpds chunk.
         * I decided against that, just in case other chunks ever are
         * discovered.
         */
        cur_pos = avio_tell(pb);
        for (i = 0; i < dpds_table_size; ++i) {
            /* From the number of output bytes that would accumulate in the
             * output buffer after decoding the first (i+1) packets, we compute
             * an offset / timestamp pair.
             */
            av_add_index_entry(st,
                               cur_pos + (i+1) * st->codec->block_align, /* pos */
                               dpds_table[i] / bytes_per_sample,         /* timestamp */
                               st->codec->block_align,                   /* size */
                               0,                                        /* duration */
                               AVINDEX_KEYFRAME);
        }
    } else if (st->codec->bit_rate) {
예제 #9
0
파일: wavdec.c 프로젝트: AronVietti/FFmpeg
static int w64_read_header(AVFormatContext *s)
{
    int64_t size, data_ofs = 0;
    AVIOContext *pb      = s->pb;
    WAVDemuxContext *wav = s->priv_data;
    AVStream *st;
    uint8_t guid[16];
    int ret;

    avio_read(pb, guid, 16);
    if (memcmp(guid, ff_w64_guid_riff, 16))
        return AVERROR_INVALIDDATA;

    /* riff + wave + fmt + sizes */
    if (avio_rl64(pb) < 16 + 8 + 16 + 8 + 16 + 8)
        return AVERROR_INVALIDDATA;

    avio_read(pb, guid, 16);
    if (memcmp(guid, ff_w64_guid_wave, 16)) {
        av_log(s, AV_LOG_ERROR, "could not find wave guid\n");
        return AVERROR_INVALIDDATA;
    }

    wav->w64 = 1;

    st = avformat_new_stream(s, NULL);
    if (!st)
        return AVERROR(ENOMEM);

    while (!url_feof(pb)) {
        if (avio_read(pb, guid, 16) != 16)
            break;
        size = avio_rl64(pb);
        if (size <= 24 || INT64_MAX - size < avio_tell(pb))
            return AVERROR_INVALIDDATA;

        if (!memcmp(guid, ff_w64_guid_fmt, 16)) {
            /* subtract chunk header size - normal wav file doesn't count it */
            ret = ff_get_wav_header(pb, st->codec, size - 24);
            if (ret < 0)
                return ret;
            avio_skip(pb, FFALIGN(size, INT64_C(8)) - size);

            avpriv_set_pts_info(st, 64, 1, st->codec->sample_rate);
        } else if (!memcmp(guid, ff_w64_guid_fact, 16)) {
            int64_t samples;

            samples = avio_rl64(pb);
            if (samples > 0)
                st->duration = samples;
        } else if (!memcmp(guid, ff_w64_guid_data, 16)) {
            wav->data_end = avio_tell(pb) + size - 24;

            data_ofs = avio_tell(pb);
            if (!pb->seekable)
                break;

            avio_skip(pb, size - 24);
        } else if (!memcmp(guid, ff_w64_guid_summarylist, 16)) {
            int64_t start, end, cur;
            uint32_t count, chunk_size, i;

            start = avio_tell(pb);
            end = start + size;
            count = avio_rl32(pb);

            for (i = 0; i < count; i++) {
                char chunk_key[5], *value;

                if (url_feof(pb) || (cur = avio_tell(pb)) < 0 || cur > end - 8 /* = tag + size */)
                    break;

                chunk_key[4] = 0;
                avio_read(pb, chunk_key, 4);
                chunk_size = avio_rl32(pb);

                value = av_mallocz(chunk_size + 1);
                if (!value)
                    return AVERROR(ENOMEM);

                ret = avio_get_str16le(pb, chunk_size, value, chunk_size);
                avio_skip(pb, chunk_size - ret);

                av_dict_set(&s->metadata, chunk_key, value, AV_DICT_DONT_STRDUP_VAL);
            }

            avio_skip(pb, end - avio_tell(pb));
        } else {
            av_log(s, AV_LOG_DEBUG, "unknown guid: "FF_PRI_GUID"\n", FF_ARG_GUID(guid));
            avio_skip(pb, size - 24);
        }
    }

    if (!data_ofs)
        return AVERROR_EOF;

    ff_metadata_conv_ctx(s, NULL, wav_metadata_conv);
    ff_metadata_conv_ctx(s, NULL, ff_riff_info_conv);

    handle_stream_probing(st);
    st->need_parsing = AVSTREAM_PARSE_FULL_RAW;

    avio_seek(pb, data_ofs, SEEK_SET);

    return 0;
}
예제 #10
0
/* wav input */
static int wav_read_header(AVFormatContext *s,
                           AVFormatParameters *ap)
{
    int64_t size, av_uninit(data_size);
    int64_t sample_count=0;
    int rf64;
    unsigned int tag;
    AVIOContext *pb = s->pb;
    AVStream *st;
    WAVContext *wav = s->priv_data;

    /* check RIFF header */
    tag = avio_rl32(pb);

    rf64 = tag == MKTAG('R', 'F', '6', '4');
    if (!rf64 && tag != MKTAG('R', 'I', 'F', 'F'))
        return -1;
    avio_rl32(pb); /* file size */
    tag = avio_rl32(pb);
    if (tag != MKTAG('W', 'A', 'V', 'E'))
        return -1;

    if (rf64) {
        if (avio_rl32(pb) != MKTAG('d', 's', '6', '4'))
            return -1;
        size = avio_rl32(pb);
        if (size < 16)
            return -1;
        avio_rl64(pb); /* RIFF size */
        data_size = avio_rl64(pb);
        sample_count = avio_rl64(pb);
        avio_seek(pb, size - 16, SEEK_CUR); /* skip rest of ds64 chunk */
    }

    /* parse fmt header */
    size = find_tag(pb, MKTAG('f', 'm', 't', ' '));
    if (size < 0)
        return -1;
    st = av_new_stream(s, 0);
    if (!st)
        return AVERROR(ENOMEM);

    ff_get_wav_header(pb, st->codec, size);
    st->need_parsing = AVSTREAM_PARSE_FULL;

    av_set_pts_info(st, 64, 1, st->codec->sample_rate);

    for (;;) {
        if (url_feof(pb))
            return -1;
        size = next_tag(pb, &tag);
        if (tag == MKTAG('d', 'a', 't', 'a')){
            break;
        }else if (tag == MKTAG('f','a','c','t') && !sample_count){
            sample_count = avio_rl32(pb);
            size -= 4;
        }
        avio_seek(pb, size, SEEK_CUR);
    }
    if (rf64)
        size = data_size;
    if (size < 0)
        return -1;
    if (!size) {
        wav->data_end = INT64_MAX;
    } else
        wav->data_end= avio_tell(pb) + size;

    if (!sample_count && st->codec->channels && av_get_bits_per_sample(st->codec->codec_id))
        sample_count = (size<<3) / (st->codec->channels * (uint64_t)av_get_bits_per_sample(st->codec->codec_id));
    if (sample_count)
        st->duration = sample_count;
    return 0;
}