/*
  * 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/lfg.h"
 #include "libavutil/sha.h"
 #include "avformat.h"

 #include "internal.h"

 
 #include "network.h"
 
 #include "flv.h"
 #include "rtmp.h"
 #include "rtmppkt.h"

 #include "url.h"

 //#define DEBUG
 

 /** 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 {
     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)

     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

 } 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
 };
 
 /**

  * Generate 'connect' call and send 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;
 

     ff_url_join(tcurl, sizeof(tcurl), proto, NULL, host, port, "/%s", 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]);

     ff_rtmp_packet_destroy(&pkt);

 }
 
 /**

  * Generate 'releaseStream' call and send 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(s, 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);
 }
 
 /**

  * Generate 'FCPublish' call and send 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(s, 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);
 }
 
 /**

  * Generate 'FCUnpublish' call and send 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(s, 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);
 }
 
 /**

  * Generate 'createStream' call and send 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(s, 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);
 }
 
 
 /**

  * Generate 'deleteStream' call and send 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(s, 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);
 }
 
 /**

  * Generate 'play' call and send it to the server, then ping the server

  * to start actual playing.
  */
 static void gen_play(URLContext *s, RTMPContext *rt)
 {
     RTMPPacket pkt;
     uint8_t *p;
 

     av_log(s, 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);
 }
 
 /**

  * Generate 'publish' call and send it to the server.

  */
 static void gen_publish(URLContext *s, RTMPContext *rt)
 {
     RTMPPacket pkt;
     uint8_t *p;
 

     av_log(s, 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);
 }
 
 /**

  * Generate ping reply and send 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));

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

 /**

  * Generate report on bytes read so far and send it to the server.

  */
 static void gen_bytes_read(URLContext *s, RTMPContext *rt, uint32_t ts)
 {
     RTMPPacket pkt;
     uint8_t *p;
 
     ff_rtmp_packet_create(&pkt, RTMP_NETWORK_CHANNEL, RTMP_PT_BYTES_READ, ts, 4);
     p = pkt.data;
     bytestream_put_be32(&p, rt->bytes_read);
     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
 
 /**

  * Calculate 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);
 }
 
 /**

  * 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)
  * @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;
 }
 
 /**

  * 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)
 {
     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;
 }
 
 /**

  * 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];
 

     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;
     client_pos = rtmp_handshake_imprint_with_digest(tosend + 1);
 

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

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

     if (i != RTMP_HANDSHAKE_PACKET_SIZE + 1) {

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

         return -1;
     }

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

     if (i != RTMP_HANDSHAKE_PACKET_SIZE) {

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

         return -1;
     }
 

     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) {

             server_pos = rtmp_validate_digest(serverdata + 1, 8);
             if (!server_pos) {

                 av_log(s, 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(s, 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

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

     } else {

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

     }
 

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

 #ifdef DEBUG

     ff_rtmp_packet_dump(s, pkt);

 #endif
 

     switch (pkt->type) {
     case RTMP_PT_CHUNK_SIZE:
         if (pkt->data_size != 4) {

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

             return -1;
         }

         av_log(s, 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_CLIENT_BW:
         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 -1;
         }

         av_log(s, AV_LOG_DEBUG, "Client bandwidth = %d\n", AV_RB32(pkt->data));

         rt->client_report_size = AV_RB32(pkt->data) >> 1;
         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(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) {
                     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(s, 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(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;

         }
         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");

             gen_bytes_read(s, rt, rpkt.timestamp + 1);
             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 -1;
         }

         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);
     }
     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);

     ffurl_close(rt->stream);

     av_free(rt);
     return 0;
 }
 
 /**

  * 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;

     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 & AVIO_FLAG_WRITE);

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

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

 
     if (port < 0)
         port = RTMP_DEFAULT_PORT;

     ff_url_join(buf, sizeof(buf), "tcp", NULL, hostname, port, NULL);

     if (ffurl_open(&rt->stream, buf, AVIO_FLAG_READ_WRITE) < 0) {
         av_log(s , 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);

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

     av_log(s, 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 = rt->stream->max_packet_size;

     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;

             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;
 
     if (size < 11) {

         av_log(s, 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 ff_rtmp_protocol = {

     .name      = "rtmp",
     .url_open  = rtmp_open,
     .url_read  = rtmp_read,
     .url_write = rtmp_write,
     .url_close = rtmp_close,

 };