コード例 #1
0
ファイル: http.c プロジェクト: JDsolution/ipnc
/* used only when posting data */
static int http_write(URLContext *h, uint8_t *buf, int size)
{
    char temp[11];  /* 32-bit hex + CRLF + nul */
    int ret;
    char crlf[] = "\r\n";
    HTTPContext *s = h->priv_data;

    if (s->chunksize == -1) {
        /* headers are sent without any special encoding */
        return url_write(s->hd, buf, size);
    }

    /* silently ignore zero-size data since chunk encoding that would
     * signal EOF */
    if (size > 0) {
        /* upload data using chunked encoding */
        snprintf(temp, sizeof(temp), "%x\r\n", size);

        if ((ret = url_write(s->hd, temp, strlen(temp))) < 0 ||
            (ret = url_write(s->hd, buf, size)) < 0 ||
            (ret = url_write(s->hd, crlf, sizeof(crlf) - 1)) < 0)
            return ret;
    }
    return size;
}
コード例 #2
0
ファイル: rtmppkt.c プロジェクト: mrtos/Logitech-Revue
int ff_rtmp_packet_write(URLContext *h, RTMPPacket *pkt,
                         int chunk_size, RTMPPacket *prev_pkt)
{
    uint8_t pkt_hdr[16], *p = pkt_hdr;
    int mode = RTMP_PS_TWELVEBYTES;
    int off = 0;

    //TODO: header compression
    bytestream_put_byte(&p, pkt->channel_id | (mode << 6));
    if (mode != RTMP_PS_ONEBYTE) {
        bytestream_put_be24(&p, pkt->timestamp);
        if (mode != RTMP_PS_FOURBYTES) {
            bytestream_put_be24(&p, pkt->data_size);
            bytestream_put_byte(&p, pkt->type);
            if (mode == RTMP_PS_TWELVEBYTES)
                bytestream_put_le32(&p, pkt->extra);
        }
    }
    url_write(h, pkt_hdr, p-pkt_hdr);
    while (off < pkt->data_size) {
        int towrite = FFMIN(chunk_size, pkt->data_size - off);
        url_write(h, pkt->data + off, towrite);
        off += towrite;
        if (off < pkt->data_size) {
            uint8_t marker = 0xC0 | pkt->channel_id;
            url_write(h, &marker, 1);
        }
    }
    return 0;
}
コード例 #3
0
ファイル: rtspenc.c プロジェクト: allweax/ffmpeg-msvc
static int tcp_write_packet(AVFormatContext *s, RTSPStream *rtsp_st)
{
    RTSPState *rt = s->priv_data;
    AVFormatContext *rtpctx = rtsp_st->transport_priv;
    uint8_t *buf, *ptr;
    int size;
    uint8_t interleave_header[4];

    size = url_close_dyn_buf(rtpctx->pb, &buf);
    ptr = buf;
    while (size > 4) {
        uint32_t packet_len = AV_RB32(ptr);
        int id;
        ptr += 4;
        size -= 4;
        if (packet_len > size || packet_len < 2)
            break;
        if (ptr[1] >= 200 && ptr[1] <= 204)
            id = rtsp_st->interleaved_max; /* RTCP */
        else
            id = rtsp_st->interleaved_min; /* RTP */
        interleave_header[0] = '$';
        interleave_header[1] = id;
        AV_WB16(interleave_header + 2, packet_len);
        url_write(rt->rtsp_hd, interleave_header, 4);
        url_write(rt->rtsp_hd, ptr, packet_len);
        ptr += packet_len;
        size -= packet_len;
    }
    av_free(buf);
    url_open_dyn_packet_buf(&rtpctx->pb, RTSP_TCP_MAX_PACKET_SIZE);
    return 0;
}
コード例 #4
0
static ssize_t exhaustive_send(URLContext *urlContext, uint8_t *packetized_data,
    int packetized_data_len) {
    uint8_t *ptr = packetized_data;
    uint8_t *end = packetized_data + packetized_data_len;
    ssize_t numBytes = 0;

    /* XXX: Do we need packet clock synchronization?  I don't think so... */
    while (ptr < end) {
        int remaining = end - ptr;
        int len;

        /* The size for each packet is actually encoded in the first 4 bytes of
         * each of a series of packets contained in `rtp_data'. */
        assert(remaining >= 4);

        len = ptr[0] << 24 | ptr[1] << 16 | ptr[2] << 8 | ptr[3];
        assert(len <= remaining - 4);

        ptr += 4;
        int n = url_write(urlContext, ptr, len);
        ptr += len;

        /* XXX: Our poor error handling comes courtesy of ffserver.c */
        if (n > 0) {
            numBytes += n;
        }
    }

    return numBytes;
}
コード例 #5
0
static int rtp_write(URLContext *h, uint8_t *buf, int size)
{
    RTPContext *s = h->priv_data;
    int ret;
    URLContext *hd;

    if (buf[1] >= 200 && buf[1] <= 204) {
        /* RTCP payload type */
        hd = s->rtcp_hd;
    } else {
        /* RTP payload type */
        hd = s->rtp_hd;
    }

    ret = url_write(hd, buf, size);
#if 0
    {
        struct timespec ts;
        ts.tv_sec = 0;
        ts.tv_nsec = 10 * 1000000;
        nanosleep(&ts, NULL);
    }
#endif
    return ret;
}
コード例 #6
0
ファイル: mmst.c プロジェクト: AirDev/linphone-android
/** Send a prepared MMST command packet. */
static int send_command_packet(MMSTContext *mmst)
{
    MMSContext *mms  = &mmst->mms;
    int len= mms->write_out_ptr - mms->out_buffer;
    int exact_length = FFALIGN(len, 8);
    int first_length= exact_length - 16;
    int len8= first_length/8;
    int write_result;

    // update packet length fields.
    AV_WL32(mms->out_buffer + 8, first_length);
    AV_WL32(mms->out_buffer + 16, len8);
    AV_WL32(mms->out_buffer + 32, len8-2);
    memset(mms->write_out_ptr, 0, exact_length - len);

    // write it out.
    write_result= url_write(mms->mms_hd, mms->out_buffer, exact_length);
    if(write_result != exact_length) {
        av_log(NULL, AV_LOG_ERROR,
               "Failed to write data of length %d: %d (%s)\n",
               exact_length, write_result,
               write_result < 0 ? strerror(write_result) :
                   "The server closed the connection");
        return AVERROR_IO;
    }

    return 0;
}
コード例 #7
0
ファイル: md5proto.c プロジェクト: AWilco/xbmc
static int md5_close(URLContext *h)
{
    const char *filename = h->filename;
    uint8_t md5[16], buf[64];
    URLContext *out;
    int i, err = 0;

    av_md5_final(h->priv_data, md5);
    for (i = 0; i < sizeof(md5); i++)
        snprintf(buf + i*2, 3, "%02x", md5[i]);
    buf[i*2] = '\n';

    av_strstart(filename, "md5:", &filename);

    if (*filename) {
        err = url_open(&out, filename, URL_WRONLY);
        if (err)
            return err;
        err = url_write(out, buf, i*2+1);
        url_close(out);
    } else {
        if (fwrite(buf, 1, i*2+1, stdout) < i*2+1)
            err = AVERROR(errno);
    }

    return err;
}
コード例 #8
0
ファイル: rtpdec.c プロジェクト: AbdunNurTomal/AndZop
void rtp_send_punch_packets(URLContext* rtp_handle)
{
    AVIOContext *pb;
    uint8_t *buf;
    int len;

    /* Send a small RTP packet */
    if (url_open_dyn_buf(&pb) < 0)
        return;

    avio_w8(pb, (RTP_VERSION << 6));
    avio_w8(pb, 0); /* Payload type */
    avio_wb16(pb, 0); /* Seq */
    avio_wb32(pb, 0); /* Timestamp */
    avio_wb32(pb, 0); /* SSRC */

    avio_flush(pb);
    len = url_close_dyn_buf(pb, &buf);
    if ((len > 0) && buf)
        url_write(rtp_handle, buf, len);
    av_free(buf);

    /* Send a minimal RTCP RR */
    if (url_open_dyn_buf(&pb) < 0)
        return;

    avio_w8(pb, (RTP_VERSION << 6));
    avio_w8(pb, RTCP_RR); /* receiver report */
    avio_wb16(pb, 1); /* length in words - 1 */
    avio_wb32(pb, 0); /* our own SSRC */

    avio_flush(pb);
    len = url_close_dyn_buf(pb, &buf);
    if ((len > 0) && buf)
        url_write(rtp_handle, buf, len);
    av_free(buf);
}
コード例 #9
0
ファイル: http.c プロジェクト: JDsolution/ipnc
static int http_close(URLContext *h)
{
    int ret = 0;
    char footer[] = "0\r\n\r\n";
    HTTPContext *s = h->priv_data;

    /* signal end of chunked encoding if used */
    if ((h->flags & URL_WRONLY) && s->chunksize != -1) {
        ret = url_write(s->hd, footer, sizeof(footer) - 1);
        ret = ret > 0 ? 0 : ret;
    }

    url_close(s->hd);
    av_free(s);
    return ret;
}
コード例 #10
0
ファイル: mmst.c プロジェクト: 119/dropcam_for_iphone
/** Send a prepared MMST command packet. */
static int send_command_packet(MMSContext *mms)
{
    int exact_length= mms->write_out_ptr - mms->out_buffer;
    int first_length= exact_length - 16;
    int len8= first_length/8;
    int write_result;

    // update packet length fields.
    AV_WL32(mms->out_buffer + 8, first_length);
    AV_WL32(mms->out_buffer + 16, len8);
    AV_WL32(mms->out_buffer + 32, len8-2);

    // write it out.
    write_result= url_write(mms->mms_hd, mms->out_buffer, exact_length);
    if(write_result != exact_length) {
        dprintf(NULL, "url_write returned: %d != %d\n",
                write_result, exact_length);
        return AVERROR_IO;
    }

    return 0;
}
コード例 #11
0
ファイル: rtspenc.c プロジェクト: EaSonic/ffmpeg
static int tcp_write_packet(AVFormatContext *s, RTSPStream *rtsp_st)
{
    RTSPState *rt = s->priv_data;
    AVFormatContext *rtpctx = rtsp_st->transport_priv;
    uint8_t *buf, *ptr;
    int size;
    uint8_t *interleave_header, *interleaved_packet;

    size = url_close_dyn_buf(rtpctx->pb, &buf);
    ptr = buf;
    while (size > 4) {
        uint32_t packet_len = AV_RB32(ptr);
        int id;
        /* The interleaving header is exactly 4 bytes, which happens to be
         * the same size as the packet length header from
         * url_open_dyn_packet_buf. So by writing the interleaving header
         * over these bytes, we get a consecutive interleaved packet
         * that can be written in one call. */
        interleaved_packet = interleave_header = ptr;
        ptr += 4;
        size -= 4;
        if (packet_len > size || packet_len < 2)
            break;
        if (ptr[1] >= RTCP_SR && ptr[1] <= RTCP_APP)
            id = rtsp_st->interleaved_max; /* RTCP */
        else
            id = rtsp_st->interleaved_min; /* RTP */
        interleave_header[0] = '$';
        interleave_header[1] = id;
        AV_WB16(interleave_header + 2, packet_len);
        url_write(rt->rtsp_hd_out, interleaved_packet, 4 + packet_len);
        ptr += packet_len;
        size -= packet_len;
    }
    av_free(buf);
    url_open_dyn_packet_buf(&rtpctx->pb, RTSP_TCP_MAX_PACKET_SIZE);
    return 0;
}
コード例 #12
0
ファイル: avio.c プロジェクト: chucolin/vlc
/*****************************************************************************
 * Write:
 *****************************************************************************/
static ssize_t Write(sout_access_out_t *p_access, block_t *p_buffer)
{
    access_sys_t *p_sys = (access_sys_t*)p_access->p_sys;
    size_t i_write = 0;

    while (p_buffer != NULL) {
        block_t *p_next = p_buffer->p_next;

#if LIBAVFORMAT_VERSION_MAJOR < 54
        int written = url_write(p_sys->context, p_buffer->p_buffer, p_buffer->i_buffer);
        if (written < 0) {
            errno = AVUNERROR(written);
            goto error;
        }
        i_write += written;
#else
        avio_write(p_sys->context, p_buffer->p_buffer, p_buffer->i_buffer);
        avio_flush(p_sys->context);
        if (p_sys->context->error) {
            errno = AVUNERROR(p_sys->context->error);
            p_sys->context->error = 0; /* FIXME? */
            goto error;
        }
        i_write += p_buffer->i_buffer;
#endif

        block_Release(p_buffer);

        p_buffer = p_next;
    }

    return i_write;

error:
    msg_Err(p_access, "Wrote only %zu bytes (%m)", i_write);
    block_ChainRelease( p_buffer );
    return i_write;
}
コード例 #13
0
ファイル: avio.c プロジェクト: sun-friderick/vlc-1
/*****************************************************************************
 * Write:
 *****************************************************************************/
static ssize_t Write(sout_access_out_t *p_access, block_t *p_buffer)
{
    sout_access_out_sys_t *p_sys = (sout_access_out_sys_t*)p_access->p_sys;
    size_t i_write = 0;
    int val;

    while (p_buffer != NULL) {
        block_t *p_next = p_buffer->p_next;

#if LIBAVFORMAT_VERSION_MAJOR < 54
        val = url_write(p_sys->context, p_buffer->p_buffer, p_buffer->i_buffer);
        if (val < 0)
            goto error;
        i_write += val;
#else
        avio_write(p_sys->context, p_buffer->p_buffer, p_buffer->i_buffer);
        avio_flush(p_sys->context);
        if ((val = p_sys->context->error) != 0) {
            p_sys->context->error = 0; /* FIXME? */
            goto error;
        }
        i_write += p_buffer->i_buffer;
#endif

        block_Release(p_buffer);

        p_buffer = p_next;
    }

    return i_write;

error:
    msg_Err(p_access, "Wrote only %zu bytes: %s", i_write,
            vlc_strerror_c(AVUNERROR(val)));
    block_ChainRelease( p_buffer );
    return i_write;
}
コード例 #14
0
static int write_buffer(stream_t *s, char *buffer, int len)
{
    int r = url_write(s->priv, buffer, len);
    return (r <= 0) ? -1 : r;
}
コード例 #15
0
ファイル: rtpdec.c プロジェクト: AbdunNurTomal/AndZop
int rtp_check_and_send_back_rr(RTPDemuxContext *s, int count)
{
    AVIOContext *pb;
    uint8_t *buf;
    int len;
    int rtcp_bytes;
    RTPStatistics *stats= &s->statistics;
    uint32_t lost;
    uint32_t extended_max;
    uint32_t expected_interval;
    uint32_t received_interval;
    uint32_t lost_interval;
    uint32_t expected;
    uint32_t fraction;
    uint64_t ntp_time= s->last_rtcp_ntp_time; // TODO: Get local ntp time?

    if (!s->rtp_ctx || (count < 1))
        return -1;

    /* TODO: I think this is way too often; RFC 1889 has algorithm for this */
    /* XXX: mpeg pts hardcoded. RTCP send every 0.5 seconds */
    s->octet_count += count;
    rtcp_bytes = ((s->octet_count - s->last_octet_count) * RTCP_TX_RATIO_NUM) /
        RTCP_TX_RATIO_DEN;
    rtcp_bytes /= 50; // mmu_man: that's enough for me... VLC sends much less btw !?
    if (rtcp_bytes < 28)
        return -1;
    s->last_octet_count = s->octet_count;

    if (url_open_dyn_buf(&pb) < 0)
        return -1;

    // Receiver Report
    avio_w8(pb, (RTP_VERSION << 6) + 1); /* 1 report block */
    avio_w8(pb, RTCP_RR);
    avio_wb16(pb, 7); /* length in words - 1 */
    // our own SSRC: we use the server's SSRC + 1 to avoid conflicts
    avio_wb32(pb, s->ssrc + 1);
    avio_wb32(pb, s->ssrc); // server SSRC
    // some placeholders we should really fill...
    // RFC 1889/p64
    extended_max= stats->cycles + stats->max_seq;
    expected= extended_max - stats->base_seq + 1;
    lost= expected - stats->received;
    lost= FFMIN(lost, 0xffffff); // clamp it since it's only 24 bits...
    expected_interval= expected - stats->expected_prior;
    stats->expected_prior= expected;
    received_interval= stats->received - stats->received_prior;
    stats->received_prior= stats->received;
    lost_interval= expected_interval - received_interval;
    if (expected_interval==0 || lost_interval<=0) fraction= 0;
    else fraction = (lost_interval<<8)/expected_interval;

    fraction= (fraction<<24) | lost;

    avio_wb32(pb, fraction); /* 8 bits of fraction, 24 bits of total packets lost */
    avio_wb32(pb, extended_max); /* max sequence received */
    avio_wb32(pb, stats->jitter>>4); /* jitter */

    if(s->last_rtcp_ntp_time==AV_NOPTS_VALUE)
    {
        avio_wb32(pb, 0); /* last SR timestamp */
        avio_wb32(pb, 0); /* delay since last SR */
    } else {
        uint32_t middle_32_bits= s->last_rtcp_ntp_time>>16; // this is valid, right? do we need to handle 64 bit values special?
        uint32_t delay_since_last= ntp_time - s->last_rtcp_ntp_time;

        avio_wb32(pb, middle_32_bits); /* last SR timestamp */
        avio_wb32(pb, delay_since_last); /* delay since last SR */
    }

    // CNAME
    avio_w8(pb, (RTP_VERSION << 6) + 1); /* 1 report block */
    avio_w8(pb, RTCP_SDES);
    len = strlen(s->hostname);
    avio_wb16(pb, (6 + len + 3) / 4); /* length in words - 1 */
    avio_wb32(pb, s->ssrc);
    avio_w8(pb, 0x01);
    avio_w8(pb, len);
    avio_write(pb, s->hostname, len);
    // padding
    for (len = (6 + len) % 4; len % 4; len++) {
        avio_w8(pb, 0);
    }

    avio_flush(pb);
    len = url_close_dyn_buf(pb, &buf);
    if ((len > 0) && buf) {
        int result;
        av_dlog(s->ic, "sending %d bytes of RR\n", len);
        result= url_write(s->rtp_ctx, buf, len);
        av_dlog(s->ic, "result from url_write: %d\n", result);
        av_free(buf);
    }
    return 0;
}
コード例 #16
0
ファイル: http.c プロジェクト: paranojik/multitv
/* used only when posting data */
static int http_write(URLContext *h, uint8_t *buf, int size)
{
    HTTPContext *s = h->priv_data;
    return url_write(s->hd, buf, size);
}
コード例 #17
0
ファイル: rtsp.c プロジェクト: Androtos/toolchain_benchmark
static void rtsp_send_cmd(AVFormatContext *s,
                          const char *cmd, RTSPHeader *reply,
                          unsigned char **content_ptr)
{
    RTSPState *rt = s->priv_data;
    char buf[4096], buf1[1024], *q;
    unsigned char ch;
    const char *p;
    int content_length, line_count;
    unsigned char *content = NULL;

    memset(reply, 0, sizeof(RTSPHeader));

    rt->seq++;
    pstrcpy(buf, sizeof(buf), cmd);
    snprintf(buf1, sizeof(buf1), "CSeq: %d\r\n", rt->seq);
    pstrcat(buf, sizeof(buf), buf1);
    if (rt->session_id[0] != '\0' && !strstr(cmd, "\nIf-Match:")) {
        snprintf(buf1, sizeof(buf1), "Session: %s\r\n", rt->session_id);
        pstrcat(buf, sizeof(buf), buf1);
    }
    pstrcat(buf, sizeof(buf), "\r\n");
#ifdef DEBUG
    printf("Sending:\n%s--\n", buf);
#endif
    url_write(rt->rtsp_hd, buf, strlen(buf));

    /* parse reply (XXX: use buffers) */
    line_count = 0;
    rt->last_reply[0] = '\0';
    for(;;) {
        q = buf;
        for(;;) {
            if (url_readbuf(rt->rtsp_hd, &ch, 1) != 1)
                break;
            if (ch == '\n')
                break;
            if (ch == '$') {
                /* XXX: only parse it if first char on line ? */
                rtsp_skip_packet(s);
            } else if (ch != '\r') {
                if ((q - buf) < sizeof(buf) - 1)
                    *q++ = ch;
            }
        }
        *q = '\0';
#ifdef DEBUG
        printf("line='%s'\n", buf);
#endif
        /* test if last line */
        if (buf[0] == '\0')
            break;
        p = buf;
        if (line_count == 0) {
            /* get reply code */
            get_word(buf1, sizeof(buf1), &p);
            get_word(buf1, sizeof(buf1), &p);
            reply->status_code = atoi(buf1);
        } else {
            rtsp_parse_line(reply, p);
            pstrcat(rt->last_reply, sizeof(rt->last_reply), p);
            pstrcat(rt->last_reply, sizeof(rt->last_reply), "\n");
        }
        line_count++;
    }

    if (rt->session_id[0] == '\0' && reply->session_id[0] != '\0')
        pstrcpy(rt->session_id, sizeof(rt->session_id), reply->session_id);

    content_length = reply->content_length;
    if (content_length > 0) {
        /* leave some room for a trailing '\0' (useful for simple parsing) */
        content = av_malloc(content_length + 1);
        (void)url_readbuf(rt->rtsp_hd, content, content_length);
        content[content_length] = '\0';
    }
    if (content_ptr)
        *content_ptr = content;
}
コード例 #18
0
ファイル: http.c プロジェクト: 1Server/OneServer-Android
static int http_connect(URLContext *h, const char *path, const char *hoststr,
                        const char *auth, int *new_location)
{
    HTTPContext *s = h->priv_data;
    int post, err;
    char line[1024];
    char headers[1024] = "";
    char *authstr = NULL;
    int64_t off = s->off;
    int len = 0;


    /* send http header */
    post = h->flags & URL_WRONLY;
    authstr = ff_http_auth_create_response(&s->auth_state, auth, path,
                                        post ? "POST" : "GET");

    /* set default headers if needed */
    if (!has_header(s->headers, "\r\nUser-Agent: "))
       len += av_strlcatf(headers + len, sizeof(headers) - len,
                          "User-Agent: %s\r\n", LIBAVFORMAT_IDENT);
    if (!has_header(s->headers, "\r\nAccept: "))
        len += av_strlcpy(headers + len, "Accept: */*\r\n",
                          sizeof(headers) - len);
    if (!has_header(s->headers, "\r\nRange: "))
        len += av_strlcatf(headers + len, sizeof(headers) - len,
                           "Range: bytes=%"PRId64"-\r\n", s->off);
    if (!has_header(s->headers, "\r\nConnection: "))
        len += av_strlcpy(headers + len, "Connection: close\r\n",
                          sizeof(headers)-len);
    if (!has_header(s->headers, "\r\nHost: "))
        len += av_strlcatf(headers + len, sizeof(headers) - len,
                           "Host: %s\r\n", hoststr);

    /* now add in custom headers */
    av_strlcpy(headers+len, s->headers, sizeof(headers)-len);

    snprintf(s->buffer, sizeof(s->buffer),
             "%s %s HTTP/1.1\r\n"
             "%s"
             "%s"
             "%s"
             "\r\n",
             post ? "POST" : "GET",
             path,
             post && s->is_chunked ? "Transfer-Encoding: chunked\r\n" : "",
             headers,
             authstr ? authstr : "");

    av_freep(&authstr);
    if (url_write(s->hd, s->buffer, strlen(s->buffer)) < 0)
        return AVERROR(EIO);

    /* init input buffer */
    s->buf_ptr = s->buffer;
    s->buf_end = s->buffer;
    s->line_count = 0;
    s->off = 0;
    s->filesize = -1;
    s->chunksize = -1;
    if (post) {
        /* always use chunked encoding for upload data */
        s->chunksize = 0;
        /* Pretend that it did work. We didn't read any header yet, since
         * we've still to send the POST data, but the code calling this
         * function will check http_code after we return. */
        s->http_code = 200;
        return 0;
    }

    /* wait for header */
    for(;;) {
        if (http_get_line(s, line, sizeof(line)) < 0)
            return AVERROR(EIO);

        dprintf(NULL, "header='%s'\n", line);

        err = process_line(h, line, s->line_count, new_location);
        if (err < 0)
            return err;
        if (err == 0)
            break;
        s->line_count++;
    }

    return (off == s->off) ? 0 : -1;
}
コード例 #19
0
ファイル: rtmppkt.c プロジェクト: LazyZhu/rt-n56u-1
int ff_rtmp_packet_write(URLContext *h, RTMPPacket *pkt,
                         int chunk_size, RTMPPacket *prev_pkt)
{
    uint8_t pkt_hdr[16], *p = pkt_hdr;
    int mode = RTMP_PS_TWELVEBYTES;
    int off = 0;
    int size = 0;

    pkt->ts_delta = pkt->timestamp - prev_pkt[pkt->channel_id].timestamp;

    //if channel_id = 0, this is first presentation of prev_pkt, send full hdr.
    if (prev_pkt[pkt->channel_id].channel_id &&
        pkt->extra == prev_pkt[pkt->channel_id].extra) {
        if (pkt->type == prev_pkt[pkt->channel_id].type &&
            pkt->data_size == prev_pkt[pkt->channel_id].data_size) {
            mode = RTMP_PS_FOURBYTES;
            if (pkt->ts_delta == prev_pkt[pkt->channel_id].ts_delta)
                mode = RTMP_PS_ONEBYTE;
        } else {
            mode = RTMP_PS_EIGHTBYTES;
        }
    }

    if (pkt->channel_id < 64) {
        bytestream_put_byte(&p, pkt->channel_id | (mode << 6));
    } else if (pkt->channel_id < 64 + 256) {
        bytestream_put_byte(&p, 0               | (mode << 6));
        bytestream_put_byte(&p, pkt->channel_id - 64);
    } else {
        bytestream_put_byte(&p, 1               | (mode << 6));
        bytestream_put_le16(&p, pkt->channel_id - 64);
    }
    if (mode != RTMP_PS_ONEBYTE) {
        uint32_t timestamp = pkt->timestamp;
        if (mode != RTMP_PS_TWELVEBYTES)
            timestamp = pkt->ts_delta;
        bytestream_put_be24(&p, timestamp >= 0xFFFFFF ? 0xFFFFFF : timestamp);
        if (mode != RTMP_PS_FOURBYTES) {
            bytestream_put_be24(&p, pkt->data_size);
            bytestream_put_byte(&p, pkt->type);
            if (mode == RTMP_PS_TWELVEBYTES)
                bytestream_put_le32(&p, pkt->extra);
        }
        if (timestamp >= 0xFFFFFF)
            bytestream_put_be32(&p, timestamp);
    }
    // save history
    prev_pkt[pkt->channel_id].channel_id = pkt->channel_id;
    prev_pkt[pkt->channel_id].type       = pkt->type;
    prev_pkt[pkt->channel_id].data_size  = pkt->data_size;
    prev_pkt[pkt->channel_id].timestamp  = pkt->timestamp;
    if (mode != RTMP_PS_TWELVEBYTES) {
        prev_pkt[pkt->channel_id].ts_delta   = pkt->ts_delta;
    } else {
        prev_pkt[pkt->channel_id].ts_delta   = pkt->timestamp;
    }
    prev_pkt[pkt->channel_id].extra      = pkt->extra;

    url_write(h, pkt_hdr, p-pkt_hdr);
    size = p - pkt_hdr + pkt->data_size;
    while (off < pkt->data_size) {
        int towrite = FFMIN(chunk_size, pkt->data_size - off);
        url_write(h, pkt->data + off, towrite);
        off += towrite;
        if (off < pkt->data_size) {
            uint8_t marker = 0xC0 | pkt->channel_id;
            url_write(h, &marker, 1);
            size++;
        }
    }
    return size;
}
コード例 #20
0
ファイル: gopher.c プロジェクト: WangCrystal/FFplayer
static int gopher_write(URLContext *h, const uint8_t *buf, int size)
{
    GopherContext *s = h->priv_data;
    return url_write(s->hd, buf, size);
}