rtmpdump librtmp 库和编译环境

/* * Copyright (C) 2005-2008 Team XBMC * http://www.xbmc.org * Copyright (C) 2008-2009 Andrej Stepanchuk * Copyright (C) 2009-2010 Howard Chu * * This file is part of librtmp. * * librtmp 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, * or (at your option) any later version. * * librtmp 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 General Public License for more details. * * You should have received a copy of the GNU Lesser General Public License * along with librtmp see the file COPYING. If not, write to * the Free Software Foundation, Inc., 51 Franklin Street, Fifth Floor, * Boston, MA 02110-1301, USA. * http://www.gnu.org/copyleft/lgpl.html */ #include <stdint.h> #include <stdlib.h> #include <string.h> #include <assert.h> #include "rtmp_sys.h" #include "log.h" #ifdef CRYPTO #ifdef USE_POLARSSL #include <polarssl/havege.h> #elif defined(USE_GNUTLS) #include <gnutls/gnutls.h> #else /* USE_OPENSSL */ #include <openssl/ssl.h> #include <openssl/rc4.h> #endif TLS_CTX RTMP_TLS_ctx; #endif #define RTMP_SIG_SIZE 1536 #define RTMP_LARGE_HEADER_SIZE 12 static const int packetSize[] = { 12, 8, 4, 1 }; int RTMP_ctrlC; const char RTMPProtocolStrings[][7] = { "RTMP", "RTMPT", "RTMPE", "RTMPTE", "RTMPS", "RTMPTS", "", "", "RTMFP" }; const char RTMPProtocolStringsLower[][7] = { "rtmp", "rtmpt", "rtmpe", "rtmpte", "rtmps", "rtmpts", "", "", "rtmfp" }; static const char *RTMPT_cmds[] = { "open", "send", "idle", "close" }; typedef enum { RTMPT_OPEN=0, RTMPT_SEND, RTMPT_IDLE, RTMPT_CLOSE } RTMPTCmd; static int DumpMetaData(AMFObject *obj); static int HandShake(RTMP *r, int FP9HandShake); static int SocksNegotiate(RTMP *r); static int SendConnectPacket(RTMP *r, RTMPPacket *cp); static int SendCheckBW(RTMP *r); static int SendCheckBWResult(RTMP *r, double txn); static int SendDeleteStream(RTMP *r, double dStreamId); static int SendFCSubscribe(RTMP *r, AVal *subscribepath); static int SendPlay(RTMP *r); static int SendBytesReceived(RTMP *r); #if 0 /* unused */ static int SendBGHasStream(RTMP *r, double dId, AVal *playpath); #endif static int HandleInvoke(RTMP *r, const char *body, unsigned int nBodySize); static int HandleMetadata(RTMP *r, char *body, unsigned int len); static void HandleChangeChunkSize(RTMP *r, const RTMPPacket *packet); static void HandleAudio(RTMP *r, const RTMPPacket *packet); static void HandleVideo(RTMP *r, const RTMPPacket *packet); static void HandleCtrl(RTMP *r, const RTMPPacket *packet); static void HandleServerBW(RTMP *r, const RTMPPacket *packet); static void HandleClientBW(RTMP *r, const RTMPPacket *packet); static int ReadN(RTMP *r, char *buffer, int n); static int WriteN(RTMP *r, const char *buffer, int n); static void DecodeTEA(AVal *key, AVal *text); static int HTTP_Post(RTMP *r, RTMPTCmd cmd, const char *buf, int len); static int HTTP_read(RTMP *r, int fill); #ifndef _WIN32 static int clk_tck; #endif #ifdef CRYPTO #include "handshake.h" #endif uint32_t RTMP_GetTime() { #ifdef _DEBUG return 0; #elif defined(_WIN32) return timeGetTime(); #else struct tms t; if (!clk_tck) clk_tck = sysconf(_SC_CLK_TCK); return times(&t) * 1000 / clk_tck; #endif } void RTMP_UserInterrupt() { RTMP_ctrlC = TRUE; } void RTMPPacket_Reset(RTMPPacket *p) { p->m_headerType = 0; p->m_packetType = 0; p->m_nChannel = 0; p->m_nTimeStamp = 0; p->m_nInfoField2 = 0; p->m_hasAbsTimestamp = FALSE; p->m_nBodySize = 0; p->m_nBytesRead = 0; } int RTMPPacket_Alloc(RTMPPacket *p, int nSize) { char *ptr = calloc(1, nSize + RTMP_MAX_HEADER_SIZE); if (!ptr) return FALSE; p->m_body = ptr + RTMP_MAX_HEADER_SIZE; p->m_nBytesRead = 0; return TRUE; } void RTMPPacket_Free(RTMPPacket *p) { if (p->m_body) { free(p->m_body - RTMP_MAX_HEADER_SIZE); p->m_body = NULL; } } void RTMPPacket_Dump(RTMPPacket *p) { RTMP_Log(RTMP_LOGDEBUG, "RTMP PACKET: packet type: 0x%02x. channel: 0x%02x. info 1: %d info 2: %d. Body size: %lu. body: 0x%02x", p->m_packetType, p->m_nChannel, p->m_nTimeStamp, p->m_nInfoField2, p->m_nBodySize, p->m_body ? (unsigned char)p->m_body[0] : 0); } int RTMP_LibVersion() { return RTMP_LIB_VERSION; } void RTMP_TLS_Init() { #ifdef CRYPTO #ifdef USE_POLARSSL /* Do this regardless of NO_SSL, we use havege for rtmpe too */ RTMP_TLS_ctx = calloc(1,sizeof(struct tls_ctx)); havege_init(&RTMP_TLS_ctx->hs); #elif defined(USE_GNUTLS) && !defined(NO_SSL) /* Technically we need to initialize libgcrypt ourselves if * we're not going to call gnutls_global_init(). Ignoring this * for now. */ gnutls_global_init(); RTMP_TLS_ctx = malloc(sizeof(struct tls_ctx)); gnutls_certificate_allocate_credentials(&RTMP_TLS_ctx->cred); gnutls_priority_init(&RTMP_TLS_ctx->prios, "NORMAL", NULL); gnutls_certificate_set_x509_trust_file(RTMP_TLS_ctx->cred, "ca.pem", GNUTLS_X509_FMT_PEM); #elif !defined(NO_SSL) /* USE_OPENSSL */ /* libcrypto doesn't need anything special */ SSL_load_error_strings(); SSL_library_init(); OpenSSL_add_all_digests(); RTMP_TLS_ctx = SSL_CTX_new(SSLv23_method()); SSL_CTX_set_options(RTMP_TLS_ctx, SSL_OP_ALL); SSL_CTX_set_default_verify_paths(RTMP_TLS_ctx); #endif #endif } RTMP * RTMP_Alloc() { return calloc(1, sizeof(RTMP)); } void RTMP_Free(RTMP *r) { free(r); } void RTMP_Init(RTMP *r) { #ifdef CRYPTO if (!RTMP_TLS_ctx) RTMP_TLS_Init(); #endif memset(r, 0, sizeof(RTMP)); r->m_sb.sb_socket = -1; r->m_inChunkSize = RTMP_DEFAULT_CHUNKSIZE; r->m_outChunkSize = RTMP_DEFAULT_CHUNKSIZE; r->m_nBufferMS = 30000; r->m_nClientBW = 2500000; r->m_nClientBW2 = 2; r->m_nServerBW = 2500000; r->m_fAudioCodecs = 3191.0; r->m_fVideoCodecs = 252.0; r->Link.timeout = 30; r->Link.swfAge = 30; } void RTMP_EnableWrite(RTMP *r) { r->Link.protocol |= RTMP_FEATURE_WRITE; } double RTMP_GetDuration(RTMP *r) { return r->m_fDuration; } int RTMP_IsConnected(RTMP *r) { return r->m_sb.sb_socket != -1; } int RTMP_Socket(RTMP *r) { return r->m_sb.sb_socket; } int RTMP_IsTimedout(RTMP *r) { return r->m_sb.sb_timedout; } void RTMP_SetBufferMS(RTMP *r, int size) { r->m_nBufferMS = size; } void RTMP_UpdateBufferMS(RTMP *r) { RTMP_SendCtrl(r, 3, r->m_stream_id, r->m_nBufferMS); } #undef OSS #ifdef _WIN32 #define OSS "WIN" #elif defined(__sun__) #define OSS "SOL" #elif defined(__APPLE__) #define OSS "MAC" #elif defined(__linux__) #define OSS "LNX" #else #define OSS "GNU" #endif #define DEF_VERSTR OSS " 10,0,32,18" static const char DEFAULT_FLASH_VER[] = DEF_VERSTR; const AVal RTMP_DefaultFlashVer = { (char *)DEFAULT_FLASH_VER, sizeof(DEFAULT_FLASH_VER) - 1 }; void RTMP_SetupStream(RTMP *r, int protocol, AVal *host, unsigned int port, AVal *sockshost, AVal *playpath, AVal *tcUrl, AVal *swfUrl, AVal *pageUrl, AVal *app, AVal *auth, AVal *swfSHA256Hash, uint32_t swfSize, AVal *flashVer, AVal *subscribepath, int dStart, int dStop, int bLiveStream, long int timeout) { RTMP_Log(RTMP_LOGDEBUG, "Protocol : %s", RTMPProtocolStrings[protocol&7]); RTMP_Log(RTMP_LOGDEBUG, "Hostname : %.*s", host->av_len, host->av_val); RTMP_Log(RTMP_LOGDEBUG, "Port : %d", port); RTMP_Log(RTMP_LOGDEBUG, "Playpath : %s", playpath->av_val); if (tcUrl && tcUrl->av_val) RTMP_Log(RTMP_LOGDEBUG, "tcUrl : %s", tcUrl->av_val); if (swfUrl && swfUrl->av_val) RTMP_Log(RTMP_LOGDEBUG, "swfUrl : %s", swfUrl->av_val); if (pageUrl && pageUrl->av_val) RTMP_Log(RTMP_LOGDEBUG, "pageUrl : %s", pageUrl->av_val); if (app && app->av_val) RTMP_Log(RTMP_LOGDEBUG, "app : %.*s", app->av_len,