FFmpeg/libavformat/rtmpproto.c

/*
 * 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
 * RTMP protocol
 */

#include "libavcodec/bytestream.h"
#include "libavutil/avstring.h"
#include "libavutil/intfloat.h"
#include "libavutil/lfg.h"
#include "libavutil/opt.h"
#include "libavutil/sha.h"
#include "avformat.h"
#include "internal.h"

#include "network.h"

#include "flv.h"
#include "rtmp.h"
#include "rtmpcrypt.h"
#include "rtmppkt.h"
#include "url.h"

//#define DEBUG

#define APP_MAX_LENGTH 128
#define PLAYPATH_MAX_LENGTH 256
#define TCURL_MAX_LENGTH 512
#define FLASHVER_MAX_LENGTH 64

/** 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)
    STATE_STOPPED,    ///< the broadcast has been stopped
} ClientState;

/** protocol handler context */
typedef struct RTMPContext {
    const AVClass *class;
    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;                  ///< stream identifier to play (with possible "mp4:" prefix)
    int           live;                       ///< 0: recorded, -1: live, -2: both
    char          *app;                       ///< name of application
    char          *conn;                      ///< append arbitrary AMF data to the Connect message
    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
    int           flv_nb_packets;             ///< number of flv packets published
    RTMPPacket    out_pkt;                    ///< rtmp packet, created from flv a/v or metadata (for output)
    uint32_t      client_report_size;         ///< number of bytes after which client should report to server
    uint32_t      bytes_read;                 ///< number of bytes read from server
    uint32_t      last_bytes_read;            ///< number of bytes read last reported to server
    int           skip_bytes;                 ///< number of bytes to skip from the input FLV stream in the next write call
    uint8_t       flv_header[11];             ///< partial incoming flv packet header
    int           flv_header_bytes;           ///< number of initialized bytes in flv_header
    int           nb_invokes;                 ///< keeps track of invoke messages
    int           create_stream_invoke;       ///< invoke id for the create stream command
    char*         tcurl;                      ///< url of the target stream
    char*         flashver;                   ///< version of the flash plugin
    char*         swfurl;                     ///< url of the swf player
    char*         pageurl;                    ///< url of the web page
    int           server_bw;                  ///< server bandwidth
    int           client_buffer_time;         ///< client buffer time in ms
    int           flush_interval;             ///< number of packets flushed in the same request (RTMPT only)
    int           encrypted;                  ///< use an encrypted connection (RTMPE only)
} 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
};

static int rtmp_write_amf_data(URLContext *s, char *param, uint8_t **p)
{
    char *field, *value;
    char type;

    /* The type must be B for Boolean, N for number, S for string, O for
     * object, or Z for null. For Booleans the data must be either 0 or 1 for
     * FALSE or TRUE, respectively. Likewise for Objects the data must be
     * 0 or 1 to end or begin an object, respectively. Data items in subobjects
     * may be named, by prefixing the type with 'N' and specifying the name
     * before the value (ie. NB:myFlag:1). This option may be used multiple times
     * to construct arbitrary AMF sequences. */
    if (param[0] && param[1] == ':') {
        type = param[0];
        value = param + 2;
    } else if (param[0] == 'N' && param[1] && param[2] == ':') {
        type = param[1];
        field = param + 3;
        value = strchr(field, ':');
        if (!value)
            goto fail;
        *value = '\0';
        value++;

        if (!field || !value)
            goto fail;

        ff_amf_write_field_name(p, field);
    } else {
        goto fail;
    }

    switch (type) {
    case 'B':
        ff_amf_write_bool(p, value[0] != '0');
        break;
    case 'S':
        ff_amf_write_string(p, value);
        break;
    case 'N':
        ff_amf_write_number(p, strtod(value, NULL));
        break;
    case 'Z':
        ff_amf_write_null(p);
        break;
    case 'O':
        if (value[0] != '0')
            ff_amf_write_object_start(p);
        else
            ff_amf_write_object_end(p);
        break;
    default:
        goto fail;
        break;
    }

    return 0;

fail:
    av_log(s, AV_LOG_ERROR, "Invalid AMF parameter: %s\n", param);
    return AVERROR(EINVAL);
}

/**
 * Generate 'connect' call and send it to the server.
 */
static int gen_connect(URLContext *s, RTMPContext *rt)
{
    RTMPPacket pkt;
    uint8_t *p;
    int ret;

    if ((ret = ff_rtmp_packet_create(&pkt, RTMP_SYSTEM_CHANNEL, RTMP_PT_INVOKE,
                                     0, 4096)) < 0)
        return ret;

    p = pkt.data;

    ff_amf_write_string(&p, "connect");
    ff_amf_write_number(&p, ++rt->nb_invokes);
    ff_amf_write_object_start(&p);
    ff_amf_write_field_name(&p, "app");
    ff_amf_write_string(&p, rt->app);

    if (!rt->is_input) {
        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, rt->flashver);

    if (rt->swfurl) {
        ff_amf_write_field_name(&p, "swfUrl");
        ff_amf_write_string(&p, rt->swfurl);
    }

    ff_amf_write_field_name(&p, "tcUrl");
    ff_amf_write_string(&p, rt->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);

        /* Tell the server we support all the audio codecs except
         * SUPPORT_SND_INTEL (0x0008) and SUPPORT_SND_UNUSED (0x0010)
         * which are unused in the RTMP protocol implementation. */
        ff_amf_write_field_name(&p, "audioCodecs");
        ff_amf_write_number(&p, 4071.0);
        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);

        if (rt->pageurl) {
            ff_amf_write_field_name(&p, "pageUrl");
            ff_amf_write_string(&p, rt->pageurl);
        }
    }
    ff_amf_write_object_end(&p);

    if (rt->conn) {
        char *param = rt->conn;

        // Write arbitrary AMF data to the Connect message.
        while (param != NULL) {
            char *sep;
            param += strspn(param, " ");
            if (!*param)
                break;
            sep = strchr(param, ' ');
            if (sep)
                *sep = '\0';
            if ((ret = rtmp_write_amf_data(s, param, &p)) < 0) {
                // Invalid AMF parameter.
                ff_rtmp_packet_destroy(&pkt);
                return ret;
            }

            if (sep)
                param = sep + 1;
            else
                break;
        }
    }

    pkt.data_size = p - pkt.data;

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

    return ret;
}

/**
 * Generate 'releaseStream' call and send it to the server. It should make
 * the server release some channel for media streams.
 */
static int gen_release_stream(URLContext *s, RTMPContext *rt)
{
    RTMPPacket pkt;
    uint8_t *p;
    int ret;

    if ((ret = ff_rtmp_packet_create(&pkt, RTMP_SYSTEM_CHANNEL, RTMP_PT_INVOKE,
                                     0, 29 + strlen(rt->playpath))) < 0)
        return ret;

    av_log(s, AV_LOG_DEBUG, "Releasing stream...\n");
    p = pkt.data;
    ff_amf_write_string(&p, "releaseStream");
    ff_amf_write_number(&p, ++rt->nb_invokes);
    ff_amf_write_null(&p);
    ff_amf_write_string(&p, rt->playpath);

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

    return ret;
}

/**
 * Generate 'FCPublish' call and send it to the server. It should make
 * the server preapare for receiving media streams.
 */
static int gen_fcpublish_stream(URLContext *s, RTMPContext *rt)
{
    RTMPPacket pkt;
    uint8_t *p;
    int ret;

    if ((ret = ff_rtmp_packet_create(&pkt, RTMP_SYSTEM_CHANNEL, RTMP_PT_INVOKE,
                                     0, 25 + strlen(rt->playpath))) < 0)
        return ret;

    av_log(s, AV_LOG_DEBUG, "FCPublish stream...\n");
    p = pkt.data;
    ff_amf_write_string(&p, "FCPublish");
    ff_amf_write_number(&p, ++rt->nb_invokes);
    ff_amf_write_null(&p);
    ff_amf_write_string(&p, rt->playpath);

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

    return ret;
}

/**
 * Generate 'FCUnpublish' call and send it to the server. It should make
 * the server destroy stream.
 */
static int gen_fcunpublish_stream(URLContext *s, RTMPContext *rt)
{
    RTMPPacket pkt;
    uint8_t *p;
    int ret;

    if ((ret = ff_rtmp_packet_create(&pkt, RTMP_SYSTEM_CHANNEL, RTMP_PT_INVOKE,
                                     0, 27 + strlen(rt->playpath))) < 0)
        return ret;

    av_log(s, AV_LOG_DEBUG, "UnPublishing stream...\n");
    p = pkt.data;
    ff_amf_write_string(&p, "FCUnpublish");
    ff_amf_write_number(&p, ++rt->nb_invokes);
    ff_amf_write_null(&p);
    ff_amf_write_string(&p, rt->playpath);

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

    return ret;
}

/**
 * Generate 'createStream' call and send it to the server. It should make
 * the server allocate some channel for media streams.
 */
static int gen_create_stream(URLContext *s, RTMPContext *rt)
{
    RTMPPacket pkt;
    uint8_t *p;
    int ret;

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

    if ((ret = ff_rtmp_packet_create(&pkt, RTMP_SYSTEM_CHANNEL, RTMP_PT_INVOKE,
                                     0, 25)) < 0)
        return ret;

    p = pkt.data;
    ff_amf_write_string(&p, "createStream");
    ff_amf_write_number(&p, ++rt->nb_invokes);
    ff_amf_write_null(&p);
    rt->create_stream_invoke = rt->nb_invokes;

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

    return ret;
}


/**
 * Generate 'deleteStream' call and send it to the server. It should make
 * the server remove some channel for media streams.
 */
static int gen_delete_stream(URLContext *s, RTMPContext *rt)
{
    RTMPPacket pkt;
    uint8_t *p;
    int ret;

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

    if ((ret = ff_rtmp_packet_create(&pkt, RTMP_SYSTEM_CHANNEL, RTMP_PT_INVOKE,
                                     0, 34)) < 0)
        return ret;

    p = pkt.data;
    ff_amf_write_string(&p, "deleteStream");
    ff_amf_write_number(&p, ++rt->nb_invokes);
    ff_amf_write_null(&p);
    ff_amf_write_number(&p, rt->main_channel_id);

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

    return ret;
}

/**
 * Generate client buffer time and send it to the server.
 */
static int gen_buffer_time(URLContext *s, RTMPContext *rt)
{
    RTMPPacket pkt;
    uint8_t *p;
    int ret;

    if ((ret = ff_rtmp_packet_create(&pkt, RTMP_NETWORK_CHANNEL, RTMP_PT_PING,
                                     1, 10)) < 0)
        return ret;

    p = pkt.data;
    bytestream_put_be16(&p, 3);
    bytestream_put_be32(&p, rt->main_channel_id);
    bytestream_put_be32(&p, rt->client_buffer_time);

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

    return ret;
}

/**
 * Generate 'play' call and send it to the server, then ping the server
 * to start actual playing.
 */
static int gen_play(URLContext *s, RTMPContext *rt)
{
    RTMPPacket pkt;
    uint8_t *p;
    int ret;

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

    if ((ret = ff_rtmp_packet_create(&pkt, RTMP_VIDEO_CHANNEL, RTMP_PT_INVOKE,
                                     0, 29 + strlen(rt->playpath))) < 0)
        return ret;

    pkt.extra = rt->main_channel_id;

    p = pkt.data;
    ff_amf_write_string(&p, "play");
    ff_amf_write_number(&p, ++rt->nb_invokes);
    ff_amf_write_null(&p);
    ff_amf_write_string(&p, rt->playpath);
    ff_amf_write_number(&p, rt->live);

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

    return ret;
}

/**
 * Generate 'publish' call and send it to the server.
 */
static int gen_publish(URLContext *s, RTMPContext *rt)
{
    RTMPPacket pkt;
    uint8_t *p;
    int ret;

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

    if ((ret = ff_rtmp_packet_create(&pkt, RTMP_SOURCE_CHANNEL, RTMP_PT_INVOKE,
                                     0, 30 + strlen(rt->playpath))) < 0)
        return ret;

    pkt.extra = rt->main_channel_id;

    p = pkt.data;
    ff_amf_write_string(&p, "publish");
    ff_amf_write_number(&p, ++rt->nb_invokes);
    ff_amf_write_null(&p);
    ff_amf_write_string(&p, rt->playpath);
    ff_amf_write_string(&p, "live");

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

    return ret;
}

/**
 * Generate ping reply and send it to the server.
 */
static int gen_pong(URLContext *s, RTMPContext *rt, RTMPPacket *ppkt)
{
    RTMPPacket pkt;
    uint8_t *p;
    int ret;

    if (ppkt->data_size < 6) {
        av_log(s, AV_LOG_ERROR, "Too short ping packet (%d)\n",
               ppkt->data_size);
        return AVERROR_INVALIDDATA;
    }

    if ((ret = ff_rtmp_packet_create(&pkt, RTMP_NETWORK_CHANNEL, RTMP_PT_PING,
                                     ppkt->timestamp + 1, 6)) < 0)
        return ret;

    p = pkt.data;
    bytestream_put_be16(&p, 7);
    bytestream_put_be32(&p, AV_RB32(ppkt->data+2));
    ret = ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size,
                               rt->prev_pkt[1]);
    ff_rtmp_packet_destroy(&pkt);

    return ret;
}

/**
 * Generate server bandwidth message and send it to the server.
 */
static int gen_server_bw(URLContext *s, RTMPContext *rt)
{
    RTMPPacket pkt;
    uint8_t *p;
    int ret;

    if ((ret = ff_rtmp_packet_create(&pkt, RTMP_NETWORK_CHANNEL, RTMP_PT_SERVER_BW,
                                     0, 4)) < 0)
        return ret;

    p = pkt.data;
    bytestream_put_be32(&p, rt->server_bw);
    ret = ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size,
                               rt->prev_pkt[1]);
    ff_rtmp_packet_destroy(&pkt);

    return ret;
}

/**
 * Generate check bandwidth message and send it to the server.
 */
static int gen_check_bw(URLContext *s, RTMPContext *rt)
{
    RTMPPacket pkt;
    uint8_t *p;
    int ret;

    if ((ret = ff_rtmp_packet_create(&pkt, RTMP_SYSTEM_CHANNEL, RTMP_PT_INVOKE,
                                     0, 21)) < 0)
        return ret;

    p = pkt.data;
    ff_amf_write_string(&p, "_checkbw");
    ff_amf_write_number(&p, ++rt->nb_invokes);
    ff_amf_write_null(&p);

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

    return ret;
}

/**
 * Generate report on bytes read so far and send it to the server.
 */
static int gen_bytes_read(URLContext *s, RTMPContext *rt, uint32_t ts)
{
    RTMPPacket pkt;
    uint8_t *p;
    int ret;

    if ((ret = ff_rtmp_packet_create(&pkt, RTMP_NETWORK_CHANNEL, RTMP_PT_BYTES_READ,
                                     ts, 4)) < 0)
        return ret;

    p = pkt.data;
    bytestream_put_be32(&p, rt->bytes_read);
    ret = ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size,
                               rt->prev_pkt[1]);
    ff_rtmp_packet_destroy(&pkt);

    return ret;
}

int ff_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 (!sha)
        return AVERROR(ENOMEM);

    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);

    return 0;
}

int ff_rtmp_calc_digest_pos(const uint8_t *buf, int off, int mod_val,
                            int add_val)
{
    int i, digest_pos = 0;

    for (i = 0; i < 4; i++)
        digest_pos += buf[i + off];
    digest_pos = digest_pos % mod_val + add_val;

    return digest_pos;
}

/**
 * Put 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)
 * @param encrypted use an encrypted connection (RTMPE)
 * @return offset to the digest inside input data
 */
static int rtmp_handshake_imprint_with_digest(uint8_t *buf, int encrypted)
{
    int ret, digest_pos;

    if (encrypted)
        digest_pos = ff_rtmp_calc_digest_pos(buf, 772, 728, 776);
    else
        digest_pos = ff_rtmp_calc_digest_pos(buf, 8, 728, 12);

    ret = ff_rtmp_calc_digest(buf, RTMP_HANDSHAKE_PACKET_SIZE, digest_pos,
                              rtmp_player_key, PLAYER_KEY_OPEN_PART_LEN,
                              buf + digest_pos);
    if (ret < 0)
        return ret;

    return digest_pos;
}

/**
 * Verify 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)
{
    uint8_t digest[32];
    int ret, digest_pos;

    digest_pos = ff_rtmp_calc_digest_pos(buf, off, 728, off + 4);

    ret = ff_rtmp_calc_digest(buf, RTMP_HANDSHAKE_PACKET_SIZE, digest_pos,
                              rtmp_server_key, SERVER_KEY_OPEN_PART_LEN,
                              digest);
    if (ret < 0)
        return ret;

    if (!memcmp(digest, buf + digest_pos, 32))
        return digest_pos;
    return 0;
}

/**
 * Perform 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], signature[32];
    int ret, type = 0;

    av_log(s, 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;

    if (rt->encrypted && CONFIG_FFRTMPCRYPT_PROTOCOL) {
        /* When the client wants to use RTMPE, we have to change the command
         * byte to 0x06 which means to use encrypted data and we have to set
         * the flash version to at least 9.0.115.0. */
        tosend[0] = 6;
        tosend[5] = 128;
        tosend[6] = 0;
        tosend[7] = 3;
        tosend[8] = 2;

        /* Initialize the Diffie-Hellmann context and generate the public key
         * to send to the server. */
        if ((ret = ff_rtmpe_gen_pub_key(rt->stream, tosend + 1)) < 0)
            return ret;
    }

    client_pos = rtmp_handshake_imprint_with_digest(tosend + 1, rt->encrypted);
    if (client_pos < 0)
        return client_pos;

    if ((ret = ffurl_write(rt->stream, tosend,
                           RTMP_HANDSHAKE_PACKET_SIZE + 1)) < 0) {
        av_log(s, AV_LOG_ERROR, "Cannot write RTMP handshake request\n");
        return ret;
    }

    if ((ret = ffurl_read_complete(rt->stream, serverdata,
                                   RTMP_HANDSHAKE_PACKET_SIZE + 1)) < 0) {
        av_log(s, AV_LOG_ERROR, "Cannot read RTMP handshake response\n");
        return ret;
    }

    if ((ret = ffurl_read_complete(rt->stream, clientdata,
                                   RTMP_HANDSHAKE_PACKET_SIZE)) < 0) {
        av_log(s, AV_LOG_ERROR, "Cannot read RTMP handshake response\n");
        return ret;
    }

    av_log(s, AV_LOG_DEBUG, "Type answer %d\n", serverdata[0]);
    av_log(s, AV_LOG_DEBUG, "Server version %d.%d.%d.%d\n",
           serverdata[5], serverdata[6], serverdata[7], serverdata[8]);

    if (rt->is_input && serverdata[5] >= 3) {
        server_pos = rtmp_validate_digest(serverdata + 1, 772);
        if (server_pos < 0)
            return server_pos;

        if (!server_pos) {
            type = 1;
            server_pos = rtmp_validate_digest(serverdata + 1, 8);
            if (server_pos < 0)
                return server_pos;

            if (!server_pos) {
                av_log(s, AV_LOG_ERROR, "Server response validating failed\n");
                return AVERROR(EIO);
            }
        }

        ret = ff_rtmp_calc_digest(tosend + 1 + client_pos, 32, 0,
                                  rtmp_server_key, sizeof(rtmp_server_key),
                                  digest);
        if (ret < 0)
            return ret;

        ret = ff_rtmp_calc_digest(clientdata, RTMP_HANDSHAKE_PACKET_SIZE - 32,
                                  0, digest, 32, signature);
        if (ret < 0)
            return ret;

        if (rt->encrypted && CONFIG_FFRTMPCRYPT_PROTOCOL) {
            /* Compute the shared secret key sent by the server and initialize
             * the RC4 encryption. */
            if ((ret = ff_rtmpe_compute_secret_key(rt->stream, serverdata + 1,
                                                   tosend + 1, type)) < 0)
                return ret;

            /* Encrypt the signature received by the server. */
            ff_rtmpe_encrypt_sig(rt->stream, signature, digest, serverdata[0]);
        }

        if (memcmp(signature, clientdata + RTMP_HANDSHAKE_PACKET_SIZE - 32, 32)) {
            av_log(s, AV_LOG_ERROR, "Signature mismatch\n");
            return AVERROR(EIO);
        }

        for (i = 0; i < RTMP_HANDSHAKE_PACKET_SIZE; i++)
            tosend[i] = av_lfg_get(&rnd) >> 24;
        ret = ff_rtmp_calc_digest(serverdata + 1 + server_pos, 32, 0,
                                  rtmp_player_key, sizeof(rtmp_player_key),
                                  digest);
        if (ret < 0)
            return ret;

        ret = ff_rtmp_calc_digest(tosend, RTMP_HANDSHAKE_PACKET_SIZE - 32, 0,
                                  digest, 32,
                                  tosend + RTMP_HANDSHAKE_PACKET_SIZE - 32);
        if (ret < 0)
            return ret;

        if (rt->encrypted && CONFIG_FFRTMPCRYPT_PROTOCOL) {
            /* Encrypt the signature to be send to the server. */
            ff_rtmpe_encrypt_sig(rt->stream, tosend +
                                 RTMP_HANDSHAKE_PACKET_SIZE - 32, digest,
                                 serverdata[0]);
        }

        // write reply back to the server
        if ((ret = ffurl_write(rt->stream, tosend,
                               RTMP_HANDSHAKE_PACKET_SIZE)) < 0)
            return ret;

        if (rt->encrypted && CONFIG_FFRTMPCRYPT_PROTOCOL) {
            /* Set RC4 keys for encryption and update the keystreams. */
            if ((ret = ff_rtmpe_update_keystream(rt->stream)) < 0)
                return ret;
        }
    } else {
        if (rt->encrypted && CONFIG_FFRTMPCRYPT_PROTOCOL) {
            /* Compute the shared secret key sent by the server and initialize
             * the RC4 encryption. */
            if ((ret = ff_rtmpe_compute_secret_key(rt->stream, serverdata + 1,
                            tosend + 1, 1)) < 0)
                return ret;

            if (serverdata[0] == 9) {
                /* Encrypt the signature received by the server. */
                ff_rtmpe_encrypt_sig(rt->stream, signature, digest,
                                     serverdata[0]);
            }
        }

        if ((ret = ffurl_write(rt->stream, serverdata + 1,
                               RTMP_HANDSHAKE_PACKET_SIZE)) < 0)
            return ret;

        if (rt->encrypted && CONFIG_FFRTMPCRYPT_PROTOCOL) {
            /* Set RC4 keys for encryption and update the keystreams. */
            if ((ret = ff_rtmpe_update_keystream(rt->stream)) < 0)
                return ret;
        }
    }

    return 0;
}

static int handle_chunk_size(URLContext *s, RTMPPacket *pkt)
{
    RTMPContext *rt = s->priv_data;
    int ret;

    if (pkt->data_size < 4) {
        av_log(s, AV_LOG_ERROR,
               "Too short chunk size change packet (%d)\n",
               pkt->data_size);
        return AVERROR_INVALIDDATA;
    }

    if (!rt->is_input) {
        if ((ret = ff_rtmp_packet_write(rt->stream, pkt, rt->chunk_size,
                                        rt->prev_pkt[1])) < 0)
            return ret;
    }

    rt->chunk_size = AV_RB32(pkt->data);
    if (rt->chunk_size <= 0) {
        av_log(s, AV_LOG_ERROR, "Incorrect chunk size %d\n", rt->chunk_size);
        return AVERROR_INVALIDDATA;
    }
    av_log(s, AV_LOG_DEBUG, "New chunk size = %d\n", rt->chunk_size);

    return 0;
}

static int handle_ping(URLContext *s, RTMPPacket *pkt)
{
    RTMPContext *rt = s->priv_data;
    int t, ret;

    if (pkt->data_size < 2) {
        av_log(s, AV_LOG_ERROR, "Too short ping packet (%d)\n",
               pkt->data_size);
        return AVERROR_INVALIDDATA;
    }

    t = AV_RB16(pkt->data);
    if (t == 6) {
        if ((ret = gen_pong(s, rt, pkt)) < 0)
            return ret;
    }

    return 0;
}

static int handle_client_bw(URLContext *s, RTMPPacket *pkt)
{
    RTMPContext *rt = s->priv_data;

    if (pkt->data_size < 4) {
        av_log(s, AV_LOG_ERROR,
               "Client bandwidth report packet is less than 4 bytes long (%d)\n",
               pkt->data_size);
        return AVERROR_INVALIDDATA;
    }

    rt->client_report_size = AV_RB32(pkt->data);
    if (rt->client_report_size <= 0) {
        av_log(s, AV_LOG_ERROR, "Incorrect client bandwidth %d\n",
                rt->client_report_size);
        return AVERROR_INVALIDDATA;

    }
    av_log(s, AV_LOG_DEBUG, "Client bandwidth = %d\n", rt->client_report_size);
    rt->client_report_size >>= 1;

    return 0;
}

static int handle_server_bw(URLContext *s, RTMPPacket *pkt)
{
    RTMPContext *rt = s->priv_data;

    if (pkt->data_size < 4) {
        av_log(s, AV_LOG_ERROR,
               "Too short server bandwidth report packet (%d)\n",
               pkt->data_size);
        return AVERROR_INVALIDDATA;
    }

    rt->server_bw = AV_RB32(pkt->data);
    if (rt->server_bw <= 0) {
        av_log(s, AV_LOG_ERROR, "Incorrect server bandwidth %d\n",
               rt->server_bw);
        return AVERROR_INVALIDDATA;
    }
    av_log(s, AV_LOG_DEBUG, "Server bandwidth = %d\n", rt->server_bw);

    return 0;
}

static int handle_invoke(URLContext *s, RTMPPacket *pkt)
{
    RTMPContext *rt = s->priv_data;
    int i, t;
    const uint8_t *data_end = pkt->data + pkt->data_size;
    int ret;

    //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(s, 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) {
                    if ((ret = gen_release_stream(s, rt)) < 0)
                        return ret;
                    if ((ret = gen_fcpublish_stream(s, rt)) < 0)
                        return ret;
                    rt->state = STATE_RELEASING;
                } else {
                    if ((ret = gen_server_bw(s, rt)) < 0)
                        return ret;
                    rt->state = STATE_CONNECTING;
                }
                if ((ret = gen_create_stream(s, rt)) < 0)
                    return ret;
                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 = av_int2double(AV_RB64(pkt->data + 11));
                    if (pkt_id == rt->create_stream_invoke)
                        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(s, AV_LOG_WARNING, "Unexpected reply on connect()\n");
                } else {
                    rt->main_channel_id = av_int2double(AV_RB64(pkt->data + 21));
                }
                if (rt->is_input) {
                    if ((ret = gen_play(s, rt)) < 0)
                        return ret;
                    if ((ret = gen_buffer_time(s, rt)) < 0)
                        return ret;
                } else {
                    if ((ret = gen_publish(s, rt)) < 0)
                        return ret;
                }
                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(s, 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.Play.Stop")) rt->state = STATE_STOPPED;
        if (!t && !strcmp(tmpstr, "NetStream.Play.UnpublishNotify")) rt->state = STATE_STOPPED;
        if (!t && !strcmp(tmpstr, "NetStream.Publish.Start")) rt->state = STATE_PUBLISHING;
    } else if (!memcmp(pkt->data, "\002\000\010onBWDone", 11)) {
        if ((ret = gen_check_bw(s, rt)) < 0)
            return ret;
    }

    return 0;
}

/**
 * Parse received packet and possibly 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 ret;

#ifdef DEBUG
    ff_rtmp_packet_dump(s, pkt);
#endif

    switch (pkt->type) {
    case RTMP_PT_CHUNK_SIZE:
        if ((ret = handle_chunk_size(s, pkt)) < 0)
            return ret;
        break;
    case RTMP_PT_PING:
        if ((ret = handle_ping(s, pkt)) < 0)
            return ret;
        break;
    case RTMP_PT_CLIENT_BW:
        if ((ret = handle_client_bw(s, pkt)) < 0)
            return ret;
        break;
    case RTMP_PT_SERVER_BW:
        if ((ret = handle_server_bw(s, pkt)) < 0)
            return ret;
        break;
    case RTMP_PT_INVOKE:
        if ((ret = handle_invoke(s, pkt)) < 0)
            return ret;
        break;
    case RTMP_PT_VIDEO:
    case RTMP_PT_AUDIO:
    case RTMP_PT_METADATA:
        /* Audio, Video and Metadata packets are parsed in get_packet() */
        break;
    default:
        av_log(s, AV_LOG_VERBOSE, "Unknown packet type received 0x%02X\n", pkt->type);
        break;
    }
    return 0;
}

/**
 * Interact 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;
    uint8_t *p;
    const uint8_t *next;
    uint32_t data_size;
    uint32_t ts, cts, pts=0;

    if (rt->state == STATE_STOPPED)
        return AVERROR_EOF;

    for (;;) {
        RTMPPacket rpkt = { 0 };
        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);
            }
        }
        rt->bytes_read += ret;
        if (rt->bytes_read - rt->last_bytes_read > rt->client_report_size) {
            av_log(s, AV_LOG_DEBUG, "Sending bytes read report\n");
            if ((ret = gen_bytes_read(s, rt, rpkt.timestamp + 1)) < 0)
                return ret;
            rt->last_bytes_read = rt->bytes_read;
        }

        ret = rtmp_parse_result(s, rt, &rpkt);
        if (ret < 0) {//serious error in current packet
            ff_rtmp_packet_destroy(&rpkt);
            return ret;
        }
        if (rt->state == STATE_STOPPED) {
            ff_rtmp_packet_destroy(&rpkt);
            return AVERROR_EOF;
        }
        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))) {
            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);
            /* rewrite timestamps */
            next = rpkt.data;
            ts = rpkt.timestamp;
            while (next - rpkt.data < rpkt.data_size - 11) {
                next++;
                data_size = bytestream_get_be24(&next);
                p=next;
                cts = bytestream_get_be24(&next);
                cts |= bytestream_get_byte(&next) << 24;
                if (pts==0)
                    pts=cts;
                ts += cts - pts;
                pts = cts;
                bytestream_put_be24(&p, ts);
                bytestream_put_byte(&p, ts >> 24);
                next += data_size + 3 + 4;
            }
            memcpy(rt->flv_data, rpkt.data, rpkt.data_size);
            ff_rtmp_packet_destroy(&rpkt);
            return 0;
        }
        ff_rtmp_packet_destroy(&rpkt);
    }
}

static int rtmp_close(URLContext *h)
{
    RTMPContext *rt = h->priv_data;
    int ret = 0;

    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)
            ret = gen_fcunpublish_stream(h, rt);
    }
    if (rt->state > STATE_HANDSHAKED)
        ret = gen_delete_stream(h, rt);

    av_freep(&rt->flv_data);
    ffurl_close(rt->stream);
    return ret;
}

/**
 * Open RTMP connection and verify 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 = s->priv_data;
    char proto[8], hostname[256], path[1024], *fname;
    char *old_app;
    uint8_t buf[2048];
    int port;
    AVDictionary *opts = NULL;
    int ret;

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

    av_url_split(proto, sizeof(proto), NULL, 0, hostname, sizeof(hostname), &port,
                 path, sizeof(path), s->filename);

    if (!strcmp(proto, "rtmpt") || !strcmp(proto, "rtmpts")) {
        if (!strcmp(proto, "rtmpts"))
            av_dict_set(&opts, "ffrtmphttp_tls", "1", 1);

        /* open the http tunneling connection */
        ff_url_join(buf, sizeof(buf), "ffrtmphttp", NULL, hostname, port, NULL);
    } else if (!strcmp(proto, "rtmps")) {
        /* open the tls connection */
        if (port < 0)
            port = RTMPS_DEFAULT_PORT;
        ff_url_join(buf, sizeof(buf), "tls", NULL, hostname, port, NULL);
    } else if (!strcmp(proto, "rtmpe") || (!strcmp(proto, "rtmpte"))) {
        if (!strcmp(proto, "rtmpte"))
            av_dict_set(&opts, "ffrtmpcrypt_tunneling", "1", 1);

        /* open the encrypted connection */
        ff_url_join(buf, sizeof(buf), "ffrtmpcrypt", NULL, hostname, port, NULL);
        rt->encrypted = 1;
    } else {
        /* open the tcp connection */
        if (port < 0)
            port = RTMP_DEFAULT_PORT;
        ff_url_join(buf, sizeof(buf), "tcp", NULL, hostname, port, NULL);
    }

    if ((ret = ffurl_open(&rt->stream, buf, AVIO_FLAG_READ_WRITE,
                          &s->interrupt_callback, &opts)) < 0) {
        av_log(s , AV_LOG_ERROR, "Cannot open connection %s\n", buf);
        goto fail;
    }

    rt->state = STATE_START;
    if ((ret = rtmp_handshake(s, rt)) < 0)
        goto fail;

    rt->chunk_size = 128;
    rt->state = STATE_HANDSHAKED;

    // Keep the application name when it has been defined by the user.
    old_app = rt->app;

    rt->app = av_malloc(APP_MAX_LENGTH);
    if (!rt->app) {
        ret = AVERROR(ENOMEM);
        goto fail;
    }

    //extract "app" part from path
    if (!strncmp(path, "/ondemand/", 10)) {
        fname = path + 10;
        memcpy(rt->app, "ondemand", 9);
    } else {
        char *next = *path ? path + 1 : path;
        char *p = strchr(next, '/');
        if (!p) {
            fname = next;
            rt->app[0] = '\0';
        } else {
            // make sure we do not mismatch a playpath for an application instance
            char *c = strchr(p + 1, ':');
            fname = strchr(p + 1, '/');
            if (!fname || (c && 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 (old_app) {
        // The name of application has been defined by the user, override it.
        av_free(rt->app);
        rt->app = old_app;
    }

    if (!rt->playpath) {
        int len = strlen(fname);

        rt->playpath = av_malloc(PLAYPATH_MAX_LENGTH);
        if (!rt->playpath) {
            ret = AVERROR(ENOMEM);
            goto fail;
        }

        if (!strchr(fname, ':') && len >= 4 &&
            (!strcmp(fname + len - 4, ".f4v") ||
             !strcmp(fname + len - 4, ".mp4"))) {
            memcpy(rt->playpath, "mp4:", 5);
        } else if (len >= 4 && !strcmp(fname + len - 4, ".flv")) {
            fname[len - 4] = '\0';
        } else {
            rt->playpath[0] = 0;
        }
        strncat(rt->playpath, fname, PLAYPATH_MAX_LENGTH - 5);
    }

    if (!rt->tcurl) {
        rt->tcurl = av_malloc(TCURL_MAX_LENGTH);
        if (!rt->tcurl) {
            ret = AVERROR(ENOMEM);
            goto fail;
        }
        ff_url_join(rt->tcurl, TCURL_MAX_LENGTH, proto, NULL, hostname,
                    port, "/%s", rt->app);
    }

    if (!rt->flashver) {
        rt->flashver = av_malloc(FLASHVER_MAX_LENGTH);
        if (!rt->flashver) {
            ret = AVERROR(ENOMEM);
            goto fail;
        }
        if (rt->is_input) {
            snprintf(rt->flashver, FLASHVER_MAX_LENGTH, "%s %d,%d,%d,%d",
                    RTMP_CLIENT_PLATFORM, RTMP_CLIENT_VER1, RTMP_CLIENT_VER2,
                    RTMP_CLIENT_VER3, RTMP_CLIENT_VER4);
        } else {
            snprintf(rt->flashver, FLASHVER_MAX_LENGTH,
                    "FMLE/3.0 (compatible; %s)", LIBAVFORMAT_IDENT);
        }
    }

    rt->client_report_size = 1048576;
    rt->bytes_read = 0;
    rt->last_bytes_read = 0;
    rt->server_bw = 2500000;

    av_log(s, AV_LOG_DEBUG, "Proto = %s, path = %s, app = %s, fname = %s\n",
           proto, path, rt->app, rt->playpath);
    if ((ret = gen_connect(s, rt)) < 0)
        goto fail;

    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;
        rt->skip_bytes = 13;
    }

    s->max_packet_size = rt->stream->max_packet_size;
    s->is_streamed     = 1;
    return 0;

fail:
    av_dict_free(&opts);
    rtmp_close(s);
    return ret;
}

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;
            return data_left;
        }
        if ((ret = get_packet(s, 0)) < 0)
           return ret;
    }
    return orig_size;
}

static int rtmp_write(URLContext *s, const uint8_t *buf, int size)
{
    RTMPContext *rt = s->priv_data;
    int size_temp = size;
    int pktsize, pkttype;
    uint32_t ts;
    const uint8_t *buf_temp = buf;
    uint8_t c;
    int ret;

    do {
        if (rt->skip_bytes) {
            int skip = FFMIN(rt->skip_bytes, size_temp);
            buf_temp       += skip;
            size_temp      -= skip;
            rt->skip_bytes -= skip;
            continue;
        }

        if (rt->flv_header_bytes < 11) {
            const uint8_t *header = rt->flv_header;
            int copy = FFMIN(11 - rt->flv_header_bytes, size_temp);
            bytestream_get_buffer(&buf_temp, rt->flv_header + rt->flv_header_bytes, copy);
            rt->flv_header_bytes += copy;
            size_temp            -= copy;
            if (rt->flv_header_bytes < 11)
                break;

            pkttype = bytestream_get_byte(&header);
            pktsize = bytestream_get_be24(&header);
            ts = bytestream_get_be24(&header);
            ts |= bytestream_get_byte(&header) << 24;
            bytestream_get_be24(&header);
            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
            if ((ret = ff_rtmp_packet_create(&rt->out_pkt, RTMP_SOURCE_CHANNEL,
                                             pkttype, ts, pktsize)) < 0)
                return ret;

            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;
            size_temp = 0;
        } else {
            bytestream_get_buffer(&buf_temp, rt->flv_data + rt->flv_off, rt->flv_size - rt->flv_off);
            size_temp   -= rt->flv_size - rt->flv_off;
            rt->flv_off += rt->flv_size - rt->flv_off;
        }

        if (rt->flv_off == rt->flv_size) {
            rt->skip_bytes = 4;

            if ((ret = ff_rtmp_packet_write(rt->stream, &rt->out_pkt,
                                            rt->chunk_size, rt->prev_pkt[1])) < 0)
                return ret;
            ff_rtmp_packet_destroy(&rt->out_pkt);
            rt->flv_size = 0;
            rt->flv_off = 0;
            rt->flv_header_bytes = 0;
            rt->flv_nb_packets++;
        }
    } while (buf_temp - buf < size);

    if (rt->flv_nb_packets < rt->flush_interval)
        return size;
    rt->flv_nb_packets = 0;

    /* set stream into nonblocking mode */
    rt->stream->flags |= AVIO_FLAG_NONBLOCK;

    /* try to read one byte from the stream */
    ret = ffurl_read(rt->stream, &c, 1);

    /* switch the stream back into blocking mode */
    rt->stream->flags &= ~AVIO_FLAG_NONBLOCK;

    if (ret == AVERROR(EAGAIN)) {
        /* no incoming data to handle */
        return size;
    } else if (ret < 0) {
        return ret;
    } else if (ret == 1) {
        RTMPPacket rpkt = { 0 };

        if ((ret = ff_rtmp_packet_read_internal(rt->stream, &rpkt,
                                                rt->chunk_size,
                                                rt->prev_pkt[0], c)) <= 0)
             return ret;

        if ((ret = rtmp_parse_result(s, rt, &rpkt)) < 0)
            return ret;

        ff_rtmp_packet_destroy(&rpkt);
    }

    return size;
}

#define OFFSET(x) offsetof(RTMPContext, x)
#define DEC AV_OPT_FLAG_DECODING_PARAM
#define ENC AV_OPT_FLAG_ENCODING_PARAM

static const AVOption rtmp_options[] = {
    {"rtmp_app", "Name of application to connect to on the RTMP server", OFFSET(app), AV_OPT_TYPE_STRING, {.str = NULL }, 0, 0, DEC|ENC},
    {"rtmp_buffer", "Set buffer time in milliseconds. The default is 3000.", OFFSET(client_buffer_time), AV_OPT_TYPE_INT, {3000}, 0, INT_MAX, DEC|ENC},
    {"rtmp_conn", "Append arbitrary AMF data to the Connect message", OFFSET(conn), AV_OPT_TYPE_STRING, {.str = NULL }, 0, 0, DEC|ENC},
    {"rtmp_flashver", "Version of the Flash plugin used to run the SWF player.", OFFSET(flashver), AV_OPT_TYPE_STRING, {.str = NULL }, 0, 0, DEC|ENC},
    {"rtmp_flush_interval", "Number of packets flushed in the same request (RTMPT only).", OFFSET(flush_interval), AV_OPT_TYPE_INT, {10}, 0, INT_MAX, ENC},
    {"rtmp_live", "Specify that the media is a live stream.", OFFSET(live), AV_OPT_TYPE_INT, {-2}, INT_MIN, INT_MAX, DEC, "rtmp_live"},
    {"any", "both", 0, AV_OPT_TYPE_CONST, {-2}, 0, 0, DEC, "rtmp_live"},
    {"live", "live stream", 0, AV_OPT_TYPE_CONST, {-1}, 0, 0, DEC, "rtmp_live"},
    {"recorded", "recorded stream", 0, AV_OPT_TYPE_CONST, {0}, 0, 0, DEC, "rtmp_live"},
    {"rtmp_pageurl", "URL of the web page in which the media was embedded. By default no value will be sent.", OFFSET(pageurl), AV_OPT_TYPE_STRING, {.str = NULL }, 0, 0, DEC},
    {"rtmp_playpath", "Stream identifier to play or to publish", OFFSET(playpath), AV_OPT_TYPE_STRING, {.str = NULL }, 0, 0, DEC|ENC},
    {"rtmp_swfurl", "URL of the SWF player. By default no value will be sent", OFFSET(swfurl), AV_OPT_TYPE_STRING, {.str = NULL }, 0, 0, DEC|ENC},
    {"rtmp_tcurl", "URL of the target stream. Defaults to proto://host[:port]/app.", OFFSET(tcurl), AV_OPT_TYPE_STRING, {.str = NULL }, 0, 0, DEC|ENC},
    { NULL },
};

static const AVClass rtmp_class = {
    .class_name = "rtmp",
    .item_name  = av_default_item_name,
    .option     = rtmp_options,
    .version    = LIBAVUTIL_VERSION_INT,
};

URLProtocol ff_rtmp_protocol = {
    .name           = "rtmp",
    .url_open       = rtmp_open,
    .url_read       = rtmp_read,
    .url_write      = rtmp_write,
    .url_close      = rtmp_close,
    .priv_data_size = sizeof(RTMPContext),
    .flags          = URL_PROTOCOL_FLAG_NETWORK,
    .priv_data_class= &rtmp_class,
};

static const AVClass rtmpe_class = {
    .class_name = "rtmpe",
    .item_name  = av_default_item_name,
    .option     = rtmp_options,
    .version    = LIBAVUTIL_VERSION_INT,
};

URLProtocol ff_rtmpe_protocol = {
    .name            = "rtmpe",
    .url_open        = rtmp_open,
    .url_read        = rtmp_read,
    .url_write       = rtmp_write,
    .url_close       = rtmp_close,
    .priv_data_size  = sizeof(RTMPContext),
    .flags           = URL_PROTOCOL_FLAG_NETWORK,
    .priv_data_class = &rtmpe_class,
};

static const AVClass rtmps_class = {
    .class_name = "rtmps",
    .item_name  = av_default_item_name,
    .option     = rtmp_options,
    .version    = LIBAVUTIL_VERSION_INT,
};

URLProtocol ff_rtmps_protocol = {
    .name            = "rtmps",
    .url_open        = rtmp_open,
    .url_read        = rtmp_read,
    .url_write       = rtmp_write,
    .url_close       = rtmp_close,
    .priv_data_size  = sizeof(RTMPContext),
    .flags           = URL_PROTOCOL_FLAG_NETWORK,
    .priv_data_class = &rtmps_class,
};

static const AVClass rtmpt_class = {
    .class_name = "rtmpt",
    .item_name  = av_default_item_name,
    .option     = rtmp_options,
    .version    = LIBAVUTIL_VERSION_INT,
};

URLProtocol ff_rtmpt_protocol = {
    .name            = "rtmpt",
    .url_open        = rtmp_open,
    .url_read        = rtmp_read,
    .url_write       = rtmp_write,
    .url_close       = rtmp_close,
    .priv_data_size  = sizeof(RTMPContext),
    .flags           = URL_PROTOCOL_FLAG_NETWORK,
    .priv_data_class = &rtmpt_class,
};

static const AVClass rtmpte_class = {
    .class_name = "rtmpte",
    .item_name  = av_default_item_name,
    .option     = rtmp_options,
    .version    = LIBAVUTIL_VERSION_INT,
};

URLProtocol ff_rtmpte_protocol = {
    .name            = "rtmpte",
    .url_open        = rtmp_open,
    .url_read        = rtmp_read,
    .url_write       = rtmp_write,
    .url_close       = rtmp_close,
    .priv_data_size  = sizeof(RTMPContext),
    .flags           = URL_PROTOCOL_FLAG_NETWORK,
    .priv_data_class = &rtmpte_class,
};

static const AVClass rtmpts_class = {
    .class_name = "rtmpts",
    .item_name  = av_default_item_name,
    .option     = rtmp_options,
    .version    = LIBAVUTIL_VERSION_INT,
};

URLProtocol ff_rtmpts_protocol = {
    .name            = "rtmpts",
    .url_open        = rtmp_open,
    .url_read        = rtmp_read,
    .url_write       = rtmp_write,
    .url_close       = rtmp_close,
    .priv_data_size  = sizeof(RTMPContext),
    .flags           = URL_PROTOCOL_FLAG_NETWORK,
    .priv_data_class = &rtmpts_class,
};