PeerStreamer

/*

 * RTMP network protocol

 * Copyright (c) 2009 Kostya Shishkov

 *

 * 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 libavformat/rtmpproto.c

 * RTMP protocol

 */

#include "libavcodec/bytestream.h"

#include "libavutil/avstring.h"

#include "libavutil/lfg.h"

#include "libavutil/sha.h"

#include "avformat.h"

#include "network.h"

#include "flv.h"

#include "rtmp.h"

#include "rtmppkt.h"

/* we can't use av_log() with URLContext yet... */

#if LIBAVFORMAT_VERSION_MAJOR < 53

#define LOG_CONTEXT NULL

#else

#define LOG_CONTEXT s

#endif

/** RTMP protocol handler state */

typedef enum {

    STATE_START,      ///< client has not done anything yet

    STATE_HANDSHAKED, ///< client has performed handshake

    STATE_RELEASING,  ///< client releasing stream before publish it (for output)

    STATE_FCPUBLISH,  ///< client FCPublishing stream (for output)

    STATE_CONNECTING, ///< client connected to server successfully

    STATE_READY,      ///< client has sent all needed commands and waits for server reply

    STATE_PLAYING,    ///< client has started receiving multimedia data from server

    STATE_PUBLISHING, ///< client has started sending multimedia data to server (for output)

} ClientState;

/** protocol handler context */

typedef struct RTMPContext {

    URLContext*   stream;                     ///< TCP stream used in interactions with RTMP server

    RTMPPacket    prev_pkt[2][RTMP_CHANNELS]; ///< packet history used when reading and sending packets

    int           chunk_size;                 ///< size of the chunks RTMP packets are divided into

    int           is_input;                   ///< input/output flag

    char          playpath[256];              ///< path to filename to play (with possible "mp4:" prefix)

    char          app[128];                   ///< application

    ClientState   state;                      ///< current state

    int           main_channel_id;            ///< an additional channel ID which is used for some invocations

    uint8_t*      flv_data;                   ///< buffer with data for demuxer

    int           flv_size;                   ///< current buffer size

    int           flv_off;                    ///< number of bytes read from current buffer

    RTMPPacket    out_pkt;                    ///< rtmp packet, created from flv a/v or metadata (for output)

} RTMPContext;

#define PLAYER_KEY_OPEN_PART_LEN 30   ///< length of partial key used for first client digest signing

/** Client key used for digest signing */

static const uint8_t rtmp_player_key[] = {

    'G', 'e', 'n', 'u', 'i', 'n', 'e', ' ', 'A', 'd', 'o', 'b', 'e', ' ',

    'F', 'l', 'a', 's', 'h', ' ', 'P', 'l', 'a', 'y', 'e', 'r', ' ', '0', '0', '1',

    0xF0, 0xEE, 0xC2, 0x4A, 0x80, 0x68, 0xBE, 0xE8, 0x2E, 0x00, 0xD0, 0xD1, 0x02,

    0x9E, 0x7E, 0x57, 0x6E, 0xEC, 0x5D, 0x2D, 0x29, 0x80, 0x6F, 0xAB, 0x93, 0xB8,

    0xE6, 0x36, 0xCF, 0xEB, 0x31, 0xAE

};

#define SERVER_KEY_OPEN_PART_LEN 36   ///< length of partial key used for first server digest signing

/** Key used for RTMP server digest signing */

static const uint8_t rtmp_server_key[] = {

    'G', 'e', 'n', 'u', 'i', 'n', 'e', ' ', 'A', 'd', 'o', 'b', 'e', ' ',

    'F', 'l', 'a', 's', 'h', ' ', 'M', 'e', 'd', 'i', 'a', ' ',

    'S', 'e', 'r', 'v', 'e', 'r', ' ', '0', '0', '1',

    0xF0, 0xEE, 0xC2, 0x4A, 0x80, 0x68, 0xBE, 0xE8, 0x2E, 0x00, 0xD0, 0xD1, 0x02,

    0x9E, 0x7E, 0x57, 0x6E, 0xEC, 0x5D, 0x2D, 0x29, 0x80, 0x6F, 0xAB, 0x93, 0xB8,

    0xE6, 0x36, 0xCF, 0xEB, 0x31, 0xAE

};

/**

 * Generates 'connect' call and sends it to the server.

 */

static void gen_connect(URLContext *s, RTMPContext *rt, const char *proto,

                        const char *host, int port)

{

    RTMPPacket pkt;

    uint8_t ver[64], *p;

    char tcurl[512];

    ff_rtmp_packet_create(&pkt, RTMP_SYSTEM_CHANNEL, RTMP_PT_INVOKE, 0, 4096);

    p = pkt.data;

    snprintf(tcurl, sizeof(tcurl), "%s://%s:%d/%s", proto, host, port, rt->app);

    ff_amf_write_string(&p, "connect");

    ff_amf_write_number(&p, 1.0);

    ff_amf_write_object_start(&p);

    ff_amf_write_field_name(&p, "app");

    ff_amf_write_string(&p, rt->app);

    if (rt->is_input) {

        snprintf(ver, sizeof(ver), "%s %d,%d,%d,%d", RTMP_CLIENT_PLATFORM, RTMP_CLIENT_VER1,

                 RTMP_CLIENT_VER2, RTMP_CLIENT_VER3, RTMP_CLIENT_VER4);

    } else {

        snprintf(ver, sizeof(ver), "FMLE/3.0 (compatible; %s)", LIBAVFORMAT_IDENT);

        ff_amf_write_field_name(&p, "type");

        ff_amf_write_string(&p, "nonprivate");

    }

    ff_amf_write_field_name(&p, "flashVer");

    ff_amf_write_string(&p, ver);

    ff_amf_write_field_name(&p, "tcUrl");

    ff_amf_write_string(&p, tcurl);

    if (rt->is_input) {

        ff_amf_write_field_name(&p, "fpad");

        ff_amf_write_bool(&p, 0);

        ff_amf_write_field_name(&p, "capabilities");

        ff_amf_write_number(&p, 15.0);

        ff_amf_write_field_name(&p, "audioCodecs");

        ff_amf_write_number(&p, 1639.0);

        ff_amf_write_field_name(&p, "videoCodecs");

        ff_amf_write_number(&p, 252.0);

        ff_amf_write_field_name(&p, "videoFunction");

        ff_amf_write_number(&p, 1.0);

    }

    ff_amf_write_object_end(&p);

    pkt.data_size = p - pkt.data;

    ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size, rt->prev_pkt[1]);

}

/**

 * Generates 'releaseStream' call and sends it to the server. It should make

 * the server release some channel for media streams.

 */

static void gen_release_stream(URLContext *s, RTMPContext *rt)

{

    RTMPPacket pkt;

    uint8_t *p;

    ff_rtmp_packet_create(&pkt, RTMP_SYSTEM_CHANNEL, RTMP_PT_INVOKE, 0,

                          29 + strlen(rt->playpath));

    av_log(LOG_CONTEXT, AV_LOG_DEBUG, "Releasing stream...\n");

    p = pkt.data;

    ff_amf_write_string(&p, "releaseStream");

    ff_amf_write_number(&p, 2.0);

    ff_amf_write_null(&p);

    ff_amf_write_string(&p, rt->playpath);

    ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size, rt->prev_pkt[1]);

    ff_rtmp_packet_destroy(&pkt);

}

/**

 * Generates 'FCPublish' call and sends it to the server. It should make

 * the server preapare for receiving media streams.

 */

static void gen_fcpublish_stream(URLContext *s, RTMPContext *rt)

{

    RTMPPacket pkt;

    uint8_t *p;

    ff_rtmp_packet_create(&pkt, RTMP_SYSTEM_CHANNEL, RTMP_PT_INVOKE, 0,

                          25 + strlen(rt->playpath));

    av_log(LOG_CONTEXT, AV_LOG_DEBUG, "FCPublish stream...\n");

    p = pkt.data;

    ff_amf_write_string(&p, "FCPublish");

    ff_amf_write_number(&p, 3.0);

    ff_amf_write_null(&p);

    ff_amf_write_string(&p, rt->playpath);

    ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size, rt->prev_pkt[1]);

    ff_rtmp_packet_destroy(&pkt);

}

/**

 * Generates 'FCUnpublish' call and sends it to the server. It should make

 * the server destroy stream.

 */

static void gen_fcunpublish_stream(URLContext *s, RTMPContext *rt)

{

    RTMPPacket pkt;

    uint8_t *p;

    ff_rtmp_packet_create(&pkt, RTMP_SYSTEM_CHANNEL, RTMP_PT_INVOKE, 0,

                          27 + strlen(rt->playpath));

    av_log(LOG_CONTEXT, AV_LOG_DEBUG, "UnPublishing stream...\n");

    p = pkt.data;

    ff_amf_write_string(&p, "FCUnpublish");

    ff_amf_write_number(&p, 5.0);

    ff_amf_write_null(&p);

    ff_amf_write_string(&p, rt->playpath);

    ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size, rt->prev_pkt[1]);

    ff_rtmp_packet_destroy(&pkt);

}

/**

 * Generates 'createStream' call and sends it to the server. It should make

 * the server allocate some channel for media streams.

 */

static void gen_create_stream(URLContext *s, RTMPContext *rt)

{

    RTMPPacket pkt;

    uint8_t *p;

    av_log(LOG_CONTEXT, AV_LOG_DEBUG, "Creating stream...\n");

    ff_rtmp_packet_create(&pkt, RTMP_SYSTEM_CHANNEL, RTMP_PT_INVOKE, 0, 25);

    p = pkt.data;

    ff_amf_write_string(&p, "createStream");

    ff_amf_write_number(&p, rt->is_input ? 3.0 : 4.0);

    ff_amf_write_null(&p);

    ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size, rt->prev_pkt[1]);

    ff_rtmp_packet_destroy(&pkt);

}

/**

 * Generates 'deleteStream' call and sends it to the server. It should make

 * the server remove some channel for media streams.

 */

static void gen_delete_stream(URLContext *s, RTMPContext *rt)

{

    RTMPPacket pkt;

    uint8_t *p;

    av_log(LOG_CONTEXT, AV_LOG_DEBUG, "Deleting stream...\n");

    ff_rtmp_packet_create(&pkt, RTMP_SYSTEM_CHANNEL, RTMP_PT_INVOKE, 0, 34);

    p = pkt.data;

    ff_amf_write_string(&p, "deleteStream");

    ff_amf_write_number(&p, 0.0);

    ff_amf_write_null(&p);

    ff_amf_write_number(&p, rt->main_channel_id);

    ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size, rt->prev_pkt[1]);

    ff_rtmp_packet_destroy(&pkt);

}

/**

 * Generates 'play' call and sends it to the server, then pings the server

 * to start actual playing.

 */

static void gen_play(URLContext *s, RTMPContext *rt)

{

    RTMPPacket pkt;

    uint8_t *p;

    av_log(LOG_CONTEXT, AV_LOG_DEBUG, "Sending play command for '%s'\n", rt->playpath);

    ff_rtmp_packet_create(&pkt, RTMP_VIDEO_CHANNEL, RTMP_PT_INVOKE, 0,

                          20 + strlen(rt->playpath));

    pkt.extra = rt->main_channel_id;

    p = pkt.data;

    ff_amf_write_string(&p, "play");

    ff_amf_write_number(&p, 0.0);

    ff_amf_write_null(&p);

    ff_amf_write_string(&p, rt->playpath);

    ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size, rt->prev_pkt[1]);

    ff_rtmp_packet_destroy(&pkt);

    // set client buffer time disguised in ping packet

    ff_rtmp_packet_create(&pkt, RTMP_NETWORK_CHANNEL, RTMP_PT_PING, 1, 10);

    p = pkt.data;

    bytestream_put_be16(&p, 3);

    bytestream_put_be32(&p, 1);

    bytestream_put_be32(&p, 256); //TODO: what is a good value here?

    ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size, rt->prev_pkt[1]);

    ff_rtmp_packet_destroy(&pkt);

}

/**

 * Generates 'publish' call and sends it to the server.

 */

static void gen_publish(URLContext *s, RTMPContext *rt)

{

    RTMPPacket pkt;

    uint8_t *p;

    av_log(LOG_CONTEXT, AV_LOG_DEBUG, "Sending publish command for '%s'\n", rt->playpath);

    ff_rtmp_packet_create(&pkt, RTMP_SOURCE_CHANNEL, RTMP_PT_INVOKE, 0,

                          30 + strlen(rt->playpath));

    pkt.extra = rt->main_channel_id;

    p = pkt.data;

    ff_amf_write_string(&p, "publish");

    ff_amf_write_number(&p, 0.0);

    ff_amf_write_null(&p);

    ff_amf_write_string(&p, rt->playpath);

    ff_amf_write_string(&p, "live");

    ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size, rt->prev_pkt[1]);

    ff_rtmp_packet_destroy(&pkt);

}

/**

 * Generates ping reply and sends it to the server.

 */

static void gen_pong(URLContext *s, RTMPContext *rt, RTMPPacket *ppkt)

{

    RTMPPacket pkt;

    uint8_t *p;

    ff_rtmp_packet_create(&pkt, RTMP_NETWORK_CHANNEL, RTMP_PT_PING, ppkt->timestamp + 1, 6);

    p = pkt.data;

    bytestream_put_be16(&p, 7);

    bytestream_put_be32(&p, AV_RB32(ppkt->data+2) + 1);

    ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size, rt->prev_pkt[1]);

    ff_rtmp_packet_destroy(&pkt);

}

//TODO: Move HMAC code somewhere. Eventually.

#define HMAC_IPAD_VAL 0x36

#define HMAC_OPAD_VAL 0x5C

/**

 * Calculates HMAC-SHA2 digest for RTMP handshake packets.

 *

 * @param src    input buffer

 * @param len    input buffer length (should be 1536)

 * @param gap    offset in buffer where 32 bytes should not be taken into account

 *               when calculating digest (since it will be used to store that digest)

 * @param key    digest key

 * @param keylen digest key length

 * @param dst    buffer where calculated digest will be stored (32 bytes)

 */

static void rtmp_calc_digest(const uint8_t *src, int len, int gap,

                             const uint8_t *key, int keylen, uint8_t *dst)

{

    struct AVSHA *sha;

    uint8_t hmac_buf[64+32] = {0};

    int i;

    sha = av_mallocz(av_sha_size);

    if (keylen < 64) {

        memcpy(hmac_buf, key, keylen);

    } else {

        av_sha_init(sha, 256);

        av_sha_update(sha,key, keylen);

        av_sha_final(sha, hmac_buf);

    }

    for (i = 0; i < 64; i++)

        hmac_buf[i] ^= HMAC_IPAD_VAL;

    av_sha_init(sha, 256);

    av_sha_update(sha, hmac_buf, 64);

    if (gap <= 0) {

        av_sha_update(sha, src, len);

    } else { //skip 32 bytes used for storing digest

        av_sha_update(sha, src, gap);

        av_sha_update(sha, src + gap + 32, len - gap - 32);

    }

    av_sha_final(sha, hmac_buf + 64);

    for (i = 0; i < 64; i++)

        hmac_buf[i] ^= HMAC_IPAD_VAL ^ HMAC_OPAD_VAL; //reuse XORed key for opad

    av_sha_init(sha, 256);

    av_sha_update(sha, hmac_buf, 64+32);

    av_sha_final(sha, dst);

    av_free(sha);

}

/**

 * Puts HMAC-SHA2 digest of packet data (except for the bytes where this digest

 * will be stored) into that packet.

 *

 * @param buf handshake data (1536 bytes)

 * @return offset to the digest inside input data

 */

static int rtmp_handshake_imprint_with_digest(uint8_t *buf)

{

    int i, digest_pos = 0;

    for (i = 8; i < 12; i++)

        digest_pos += buf[i];

    digest_pos = (digest_pos % 728) + 12;

    rtmp_calc_digest(buf, RTMP_HANDSHAKE_PACKET_SIZE, digest_pos,

                     rtmp_player_key, PLAYER_KEY_OPEN_PART_LEN,

                     buf + digest_pos);

    return digest_pos;

}

/**

 * Verifies that the received server response has the expected digest value.

 *

 * @param buf handshake data received from the server (1536 bytes)

 * @param off position to search digest offset from

 * @return 0 if digest is valid, digest position otherwise

 */

static int rtmp_validate_digest(uint8_t *buf, int off)

{

    int i, digest_pos = 0;

    uint8_t digest[32];

    for (i = 0; i < 4; i++)

        digest_pos += buf[i + off];

    digest_pos = (digest_pos % 728) + off + 4;

    rtmp_calc_digest(buf, RTMP_HANDSHAKE_PACKET_SIZE, digest_pos,

                     rtmp_server_key, SERVER_KEY_OPEN_PART_LEN,

                     digest);

    if (!memcmp(digest, buf + digest_pos, 32))

        return digest_pos;

    return 0;

}

/**

 * Performs handshake with the server by means of exchanging pseudorandom data

 * signed with HMAC-SHA2 digest.

 *

 * @return 0 if handshake succeeds, negative value otherwise

 */

static int rtmp_handshake(URLContext *s, RTMPContext *rt)

{

    AVLFG rnd;

    uint8_t tosend    [RTMP_HANDSHAKE_PACKET_SIZE+1] = {

        3,                // unencrypted data

        0, 0, 0, 0,       // client uptime

        RTMP_CLIENT_VER1,

        RTMP_CLIENT_VER2,

        RTMP_CLIENT_VER3,

        RTMP_CLIENT_VER4,

    };

    uint8_t clientdata[RTMP_HANDSHAKE_PACKET_SIZE];

    uint8_t serverdata[RTMP_HANDSHAKE_PACKET_SIZE+1];

    int i;

    int server_pos, client_pos;

    uint8_t digest[32];

    av_log(LOG_CONTEXT, AV_LOG_DEBUG, "Handshaking...\n");

    av_lfg_init(&rnd, 0xDEADC0DE);

    // generate handshake packet - 1536 bytes of pseudorandom data

    for (i = 9; i <= RTMP_HANDSHAKE_PACKET_SIZE; i++)

        tosend[i] = av_lfg_get(&rnd) >> 24;

    client_pos = rtmp_handshake_imprint_with_digest(tosend + 1);

    url_write(rt->stream, tosend, RTMP_HANDSHAKE_PACKET_SIZE + 1);

    i = url_read_complete(rt->stream, serverdata, RTMP_HANDSHAKE_PACKET_SIZE + 1);

    if (i != RTMP_HANDSHAKE_PACKET_SIZE + 1) {

        av_log(LOG_CONTEXT, AV_LOG_ERROR, "Cannot read RTMP handshake response\n");

        return -1;

    }

    i = url_read_complete(rt->stream, clientdata, RTMP_HANDSHAKE_PACKET_SIZE);

    if (i != RTMP_HANDSHAKE_PACKET_SIZE) {

        av_log(LOG_CONTEXT, AV_LOG_ERROR, "Cannot read RTMP handshake response\n");

        return -1;

    }

    av_log(LOG_CONTEXT, AV_LOG_DEBUG, "Server version %d.%d.%d.%d\n",

           serverdata[5], serverdata[6], serverdata[7], serverdata[8]);

    if (rt->is_input) {

        server_pos = rtmp_validate_digest(serverdata + 1, 772);

        if (!server_pos) {

            server_pos = rtmp_validate_digest(serverdata + 1, 8);

            if (!server_pos) {

                av_log(LOG_CONTEXT, AV_LOG_ERROR, "Server response validating failed\n");

                return -1;

            }

        }

        rtmp_calc_digest(tosend + 1 + client_pos, 32, 0,

                         rtmp_server_key, sizeof(rtmp_server_key),

                         digest);

        rtmp_calc_digest(clientdata, RTMP_HANDSHAKE_PACKET_SIZE-32, 0,

                         digest, 32,

                         digest);

        if (memcmp(digest, clientdata + RTMP_HANDSHAKE_PACKET_SIZE - 32, 32)) {

            av_log(LOG_CONTEXT, AV_LOG_ERROR, "Signature mismatch\n");

            return -1;

        }

        for (i = 0; i < RTMP_HANDSHAKE_PACKET_SIZE; i++)

            tosend[i] = av_lfg_get(&rnd) >> 24;

        rtmp_calc_digest(serverdata + 1 + server_pos, 32, 0,

                         rtmp_player_key, sizeof(rtmp_player_key),

                         digest);

        rtmp_calc_digest(tosend,  RTMP_HANDSHAKE_PACKET_SIZE - 32, 0,

                         digest, 32,

                         tosend + RTMP_HANDSHAKE_PACKET_SIZE - 32);

        // write reply back to the server

        url_write(rt->stream, tosend, RTMP_HANDSHAKE_PACKET_SIZE);

    } else {

        url_write(rt->stream, serverdata+1, RTMP_HANDSHAKE_PACKET_SIZE);

    }

    return 0;

}

/**

 * Parses received packet and may perform some action depending on

 * the packet contents.

 * @return 0 for no errors, negative values for serious errors which prevent

 *         further communications, positive values for uncritical errors

 */

static int rtmp_parse_result(URLContext *s, RTMPContext *rt, RTMPPacket *pkt)

{

    int i, t;

    const uint8_t *data_end = pkt->data + pkt->data_size;

    switch (pkt->type) {

    case RTMP_PT_CHUNK_SIZE:

        if (pkt->data_size != 4) {

            av_log(LOG_CONTEXT, AV_LOG_ERROR,

                   "Chunk size change packet is not 4 bytes long (%d)\n", pkt->data_size);

            return -1;

        }

        if (!rt->is_input)

            ff_rtmp_packet_write(rt->stream, pkt, rt->chunk_size, rt->prev_pkt[1]);

        rt->chunk_size = AV_RB32(pkt->data);

        if (rt->chunk_size <= 0) {

            av_log(LOG_CONTEXT, AV_LOG_ERROR, "Incorrect chunk size %d\n", rt->chunk_size);

            return -1;

        }

        av_log(LOG_CONTEXT, AV_LOG_DEBUG, "New chunk size = %d\n", rt->chunk_size);

        break;

    case RTMP_PT_PING:

        t = AV_RB16(pkt->data);

        if (t == 6)

            gen_pong(s, rt, pkt);

        break;

    case RTMP_PT_INVOKE:

        //TODO: check for the messages sent for wrong state?

        if (!memcmp(pkt->data, "\002\000\006_error", 9)) {

            uint8_t tmpstr[256];

            if (!ff_amf_get_field_value(pkt->data + 9, data_end,

                                        "description", tmpstr, sizeof(tmpstr)))

                av_log(LOG_CONTEXT, AV_LOG_ERROR, "Server error: %s\n",tmpstr);

            return -1;

        } else if (!memcmp(pkt->data, "\002\000\007_result", 10)) {

            switch (rt->state) {

            case STATE_HANDSHAKED:

                if (!rt->is_input) {

                    gen_release_stream(s, rt);

                    gen_fcpublish_stream(s, rt);

                    rt->state = STATE_RELEASING;

                } else {

                    rt->state = STATE_CONNECTING;

                }

                gen_create_stream(s, rt);

                break;

            case STATE_FCPUBLISH:

                rt->state = STATE_CONNECTING;

                break;

            case STATE_RELEASING:

                rt->state = STATE_FCPUBLISH;

                /* hack for Wowza Media Server, it does not send result for

                 * releaseStream and FCPublish calls */

                if (!pkt->data[10]) {

                    int pkt_id = (int) av_int2dbl(AV_RB64(pkt->data + 11));

                    if (pkt_id == 4)

                        rt->state = STATE_CONNECTING;

                }

                if (rt->state != STATE_CONNECTING)

                    break;

            case STATE_CONNECTING:

                //extract a number from the result

                if (pkt->data[10] || pkt->data[19] != 5 || pkt->data[20]) {

                    av_log(LOG_CONTEXT, AV_LOG_WARNING, "Unexpected reply on connect()\n");

                } else {

                    rt->main_channel_id = (int) av_int2dbl(AV_RB64(pkt->data + 21));

                }

                if (rt->is_input) {

                    gen_play(s, rt);

                } else {

                    gen_publish(s, rt);

                }

                rt->state = STATE_READY;

                break;

            }

        } else if (!memcmp(pkt->data, "\002\000\010onStatus", 11)) {

            const uint8_t* ptr = pkt->data + 11;

            uint8_t tmpstr[256];

            for (i = 0; i < 2; i++) {

                t = ff_amf_tag_size(ptr, data_end);

                if (t < 0)

                    return 1;

                ptr += t;

            }

            t = ff_amf_get_field_value(ptr, data_end,

                                       "level", tmpstr, sizeof(tmpstr));

            if (!t && !strcmp(tmpstr, "error")) {

                if (!ff_amf_get_field_value(ptr, data_end,

                                            "description", tmpstr, sizeof(tmpstr)))

                    av_log(LOG_CONTEXT, AV_LOG_ERROR, "Server error: %s\n",tmpstr);

                return -1;

            }

            t = ff_amf_get_field_value(ptr, data_end,

                                       "code", tmpstr, sizeof(tmpstr));

            if (!t && !strcmp(tmpstr, "NetStream.Play.Start")) rt->state = STATE_PLAYING;

            if (!t && !strcmp(tmpstr, "NetStream.Publish.Start")) rt->state = STATE_PUBLISHING;

        }

        break;

    }

    return 0;

}

/**

 * Interacts with the server by receiving and sending RTMP packets until

 * there is some significant data (media data or expected status notification).

 *

 * @param s          reading context

 * @param for_header non-zero value tells function to work until it

 * gets notification from the server that playing has been started,

 * otherwise function will work until some media data is received (or

 * an error happens)

 * @return 0 for successful operation, negative value in case of error

 */

static int get_packet(URLContext *s, int for_header)

{

    RTMPContext *rt = s->priv_data;

    int ret;

    for (;;) {

        RTMPPacket rpkt;

        if ((ret = ff_rtmp_packet_read(rt->stream, &rpkt,

                                       rt->chunk_size, rt->prev_pkt[0])) != 0) {

            if (ret > 0) {

                return AVERROR(EAGAIN);

            } else {

                return AVERROR(EIO);

            }

        }

        ret = rtmp_parse_result(s, rt, &rpkt);

        if (ret < 0) {//serious error in current packet

            ff_rtmp_packet_destroy(&rpkt);

            return -1;

        }

        if (for_header && (rt->state == STATE_PLAYING || rt->state == STATE_PUBLISHING)) {

            ff_rtmp_packet_destroy(&rpkt);

            return 0;

        }

        if (!rpkt.data_size || !rt->is_input) {

            ff_rtmp_packet_destroy(&rpkt);

            continue;

        }

        if (rpkt.type == RTMP_PT_VIDEO || rpkt.type == RTMP_PT_AUDIO ||

           (rpkt.type == RTMP_PT_NOTIFY && !memcmp("\002\000\012onMetaData", rpkt.data, 13))) {

            uint8_t *p;

            uint32_t ts = rpkt.timestamp;

            // generate packet header and put data into buffer for FLV demuxer

            rt->flv_off  = 0;

            rt->flv_size = rpkt.data_size + 15;

            rt->flv_data = p = av_realloc(rt->flv_data, rt->flv_size);

            bytestream_put_byte(&p, rpkt.type);

            bytestream_put_be24(&p, rpkt.data_size);

            bytestream_put_be24(&p, ts);

            bytestream_put_byte(&p, ts >> 24);

            bytestream_put_be24(&p, 0);

            bytestream_put_buffer(&p, rpkt.data, rpkt.data_size);

            bytestream_put_be32(&p, 0);

            ff_rtmp_packet_destroy(&rpkt);

            return 0;

        } else if (rpkt.type == RTMP_PT_METADATA) {

            // we got raw FLV data, make it available for FLV demuxer

            rt->flv_off  = 0;

            rt->flv_size = rpkt.data_size;

            rt->flv_data = av_realloc(rt->flv_data, rt->flv_size);

            memcpy(rt->flv_data, rpkt.data, rpkt.data_size);

            ff_rtmp_packet_destroy(&rpkt);

            return 0;

        }

        ff_rtmp_packet_destroy(&rpkt);

    }

    return 0;

}

static int rtmp_close(URLContext *h)

{

    RTMPContext *rt = h->priv_data;

    if (!rt->is_input) {

        rt->flv_data = NULL;

        if (rt->out_pkt.data_size)

            ff_rtmp_packet_destroy(&rt->out_pkt);

        if (rt->state > STATE_FCPUBLISH)

            gen_fcunpublish_stream(h, rt);

    }

    if (rt->state > STATE_HANDSHAKED)

        gen_delete_stream(h, rt);

    av_freep(&rt->flv_data);

    url_close(rt->stream);

    av_free(rt);

    return 0;

}

/**

 * Opens RTMP connection and verifies that the stream can be played.

 *

 * URL syntax: rtmp://server[:port][/app][/playpath]

 *             where 'app' is first one or two directories in the path

 *             (e.g. /ondemand/, /flash/live/, etc.)

 *             and 'playpath' is a file name (the rest of the path,

 *             may be prefixed with "mp4:")

 */

static int rtmp_open(URLContext *s, const char *uri, int flags)

{

    RTMPContext *rt;

    char proto[8], hostname[256], path[1024], *fname;

    uint8_t buf[2048];

    int port;

    int ret;

    rt = av_mallocz(sizeof(RTMPContext));

    if (!rt)

        return AVERROR(ENOMEM);

    s->priv_data = rt;

    rt->is_input = !(flags & URL_WRONLY);

    url_split(proto, sizeof(proto), NULL, 0, hostname, sizeof(hostname), &port,

              path, sizeof(path), s->filename);

    if (port < 0)

        port = RTMP_DEFAULT_PORT;

    snprintf(buf, sizeof(buf), "tcp://%s:%d", hostname, port);

    if (url_open(&rt->stream, buf, URL_RDWR) < 0) {

        av_log(LOG_CONTEXT, AV_LOG_ERROR, "Cannot open connection %s\n", buf);

        goto fail;

    }

    rt->state = STATE_START;

    if (rtmp_handshake(s, rt))

        return -1;

    rt->chunk_size = 128;

    rt->state = STATE_HANDSHAKED;

    //extract "app" part from path

    if (!strncmp(path, "/ondemand/", 10)) {

        fname = path + 10;

        memcpy(rt->app, "ondemand", 9);

    } else {

        char *p = strchr(path + 1, '/');

        if (!p) {

            fname = path + 1;

            rt->app[0] = '\0';

        } else {

            char *c = strchr(p + 1, ':');

            fname = strchr(p + 1, '/');

            if (!fname || c < fname) {

                fname = p + 1;

                av_strlcpy(rt->app, path + 1, p - path);

            } else {

                fname++;

                av_strlcpy(rt->app, path + 1, fname - path - 1);

            }

        }

    }

    if (!strchr(fname, ':') &&

        (!strcmp(fname + strlen(fname) - 4, ".f4v") ||

         !strcmp(fname + strlen(fname) - 4, ".mp4"))) {

        memcpy(rt->playpath, "mp4:", 5);

    } else {

        rt->playpath[0] = 0;

    }

    strncat(rt->playpath, fname, sizeof(rt->playpath) - 5);

    av_log(LOG_CONTEXT, AV_LOG_DEBUG, "Proto = %s, path = %s, app = %s, fname = %s\n",

           proto, path, rt->app, rt->playpath);

    gen_connect(s, rt, proto, hostname, port);

    do {

        ret = get_packet(s, 1);

    } while (ret == EAGAIN);

    if (ret < 0)

        goto fail;

    if (rt->is_input) {

        // generate FLV header for demuxer

        rt->flv_size = 13;

        rt->flv_data = av_realloc(rt->flv_data, rt->flv_size);

        rt->flv_off  = 0;

        memcpy(rt->flv_data, "FLV\1\5\0\0\0\011\0\0\0\0", rt->flv_size);

    } else {

        rt->flv_size = 0;

        rt->flv_data = NULL;

        rt->flv_off  = 0;

    }

    s->max_packet_size = url_get_max_packet_size(rt->stream);

    s->is_streamed     = 1;

    return 0;

fail:

    rtmp_close(s);

    return AVERROR(EIO);

}

static int rtmp_read(URLContext *s, uint8_t *buf, int size)

{

    RTMPContext *rt = s->priv_data;

    int orig_size = size;

    int ret;

    while (size > 0) {

        int data_left = rt->flv_size - rt->flv_off;

        if (data_left >= size) {

            memcpy(buf, rt->flv_data + rt->flv_off, size);

            rt->flv_off += size;

            return orig_size;

        }

        if (data_left > 0) {

            memcpy(buf, rt->flv_data + rt->flv_off, data_left);

            buf  += data_left;

            size -= data_left;

            rt->flv_off = rt->flv_size;

        }

        if ((ret = get_packet(s, 0)) < 0)

           return ret;

    }

    return orig_size;

}

static int rtmp_write(URLContext *h, uint8_t *buf, int size)

{

    RTMPContext *rt = h->priv_data;

    int size_temp = size;

    int pktsize, pkttype;

    uint32_t ts;

    const uint8_t *buf_temp = buf;

    if (size < 11) {

        av_log(LOG_CONTEXT, AV_LOG_DEBUG, "FLV packet too small %d\n", size);

        return 0;

    }

    do {

        if (!rt->flv_off) {

            //skip flv header

            if (buf_temp[0] == 'F' && buf_temp[1] == 'L' && buf_temp[2] == 'V') {

                buf_temp += 9 + 4;

                size_temp -= 9 + 4;

            }

            pkttype = bytestream_get_byte(&buf_temp);

            pktsize = bytestream_get_be24(&buf_temp);

            ts = bytestream_get_be24(&buf_temp);

            ts |= bytestream_get_byte(&buf_temp) << 24;

            bytestream_get_be24(&buf_temp);

            size_temp -= 11;

            rt->flv_size = pktsize;

            //force 12bytes header

            if (((pkttype == RTMP_PT_VIDEO || pkttype == RTMP_PT_AUDIO) && ts == 0) ||

                pkttype == RTMP_PT_NOTIFY) {

                if (pkttype == RTMP_PT_NOTIFY)

                    pktsize += 16;

                rt->prev_pkt[1][RTMP_SOURCE_CHANNEL].channel_id = 0;

            }

            //this can be a big packet, it's better to send it right here

            ff_rtmp_packet_create(&rt->out_pkt, RTMP_SOURCE_CHANNEL, pkttype, ts, pktsize);

            rt->out_pkt.extra = rt->main_channel_id;

            rt->flv_data = rt->out_pkt.data;

            if (pkttype == RTMP_PT_NOTIFY)

                ff_amf_write_string(&rt->flv_data, "@setDataFrame");

        }

        if (rt->flv_size - rt->flv_off > size_temp) {

            bytestream_get_buffer(&buf_temp, rt->flv_data + rt->flv_off, size_temp);

            rt->flv_off += size_temp;

        } else {

            bytestream_get_buffer(&buf_temp, rt->flv_data + rt->flv_off, rt->flv_size - rt->flv_off);

            rt->flv_off += rt->flv_size - rt->flv_off;

        }

        if (rt->flv_off == rt->flv_size) {

            bytestream_get_be32(&buf_temp);

            ff_rtmp_packet_write(rt->stream, &rt->out_pkt, rt->chunk_size, rt->prev_pkt[1]);

            ff_rtmp_packet_destroy(&rt->out_pkt);

            rt->flv_size = 0;

            rt->flv_off = 0;

        }

    } while (buf_temp - buf < size_temp);

    return size;

}

URLProtocol rtmp_protocol = {

    "rtmp",

    rtmp_open,

    rtmp_read,

    rtmp_write,

    NULL, /* seek */

    rtmp_close,

};