Mercurial > libavformat.hg
view rtmpproto.c @ 6291:3a8dfc455ff0 libavformat
Fix a compile warning when compiling with DEBUG=1.
Patch by Zhentan Feng <spyfeng gmail com>.
| author | rbultje |
|---|---|
| date | Tue, 20 Jul 2010 15:05:45 +0000 |
| parents | fa244b4fe683 |
| children |
line wrap: on
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/* * 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" /* we can't use av_log() with URLContext yet... */ #if LIBAVFORMAT_VERSION_MAJOR < 53 #define LOG_CONTEXT NULL #else #define LOG_CONTEXT s #endif //#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(LOG_CONTEXT, AV_LOG_DEBUG, "Releasing stream...\n"); p = pkt.data; ff_amf_write_string(&p, "releaseStream"); ff_amf_write_number(&p, 2.0); ff_amf_write_null(&p); ff_amf_write_string(&p, rt->playpath); ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size, rt->prev_pkt[1]); ff_rtmp_packet_destroy(&pkt); } /** * 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(LOG_CONTEXT, AV_LOG_DEBUG, "FCPublish stream...\n"); p = pkt.data; ff_amf_write_string(&p, "FCPublish"); ff_amf_write_number(&p, 3.0); ff_amf_write_null(&p); ff_amf_write_string(&p, rt->playpath); ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size, rt->prev_pkt[1]); ff_rtmp_packet_destroy(&pkt); } /** * 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(LOG_CONTEXT, AV_LOG_DEBUG, "UnPublishing stream...\n"); p = pkt.data; ff_amf_write_string(&p, "FCUnpublish"); ff_amf_write_number(&p, 5.0); ff_amf_write_null(&p); ff_amf_write_string(&p, rt->playpath); ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size, rt->prev_pkt[1]); ff_rtmp_packet_destroy(&pkt); } /** * 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(LOG_CONTEXT, AV_LOG_DEBUG, "Creating stream...\n"); ff_rtmp_packet_create(&pkt, RTMP_SYSTEM_CHANNEL, RTMP_PT_INVOKE, 0, 25); p = pkt.data; ff_amf_write_string(&p, "createStream"); ff_amf_write_number(&p, rt->is_input ? 3.0 : 4.0); ff_amf_write_null(&p); ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size, rt->prev_pkt[1]); ff_rtmp_packet_destroy(&pkt); } /** * 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(LOG_CONTEXT, AV_LOG_DEBUG, "Deleting stream...\n"); ff_rtmp_packet_create(&pkt, RTMP_SYSTEM_CHANNEL, RTMP_PT_INVOKE, 0, 34); p = pkt.data; ff_amf_write_string(&p, "deleteStream"); ff_amf_write_number(&p, 0.0); ff_amf_write_null(&p); ff_amf_write_number(&p, rt->main_channel_id); ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size, rt->prev_pkt[1]); ff_rtmp_packet_destroy(&pkt); } /** * 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(LOG_CONTEXT, AV_LOG_DEBUG, "Sending play command for '%s'\n", rt->playpath); ff_rtmp_packet_create(&pkt, RTMP_VIDEO_CHANNEL, RTMP_PT_INVOKE, 0, 20 + strlen(rt->playpath)); pkt.extra = rt->main_channel_id; p = pkt.data; ff_amf_write_string(&p, "play"); ff_amf_write_number(&p, 0.0); ff_amf_write_null(&p); ff_amf_write_string(&p, rt->playpath); ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size, rt->prev_pkt[1]); ff_rtmp_packet_destroy(&pkt); // set client buffer time disguised in ping packet ff_rtmp_packet_create(&pkt, RTMP_NETWORK_CHANNEL, RTMP_PT_PING, 1, 10); p = pkt.data; bytestream_put_be16(&p, 3); bytestream_put_be32(&p, 1); bytestream_put_be32(&p, 256); //TODO: what is a good value here? ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size, rt->prev_pkt[1]); ff_rtmp_packet_destroy(&pkt); } /** * Generate 'publish' call and send it to the server. */ static void gen_publish(URLContext *s, RTMPContext *rt) { RTMPPacket pkt; uint8_t *p; av_log(LOG_CONTEXT, AV_LOG_DEBUG, "Sending publish command for '%s'\n", rt->playpath); ff_rtmp_packet_create(&pkt, RTMP_SOURCE_CHANNEL, RTMP_PT_INVOKE, 0, 30 + strlen(rt->playpath)); pkt.extra = rt->main_channel_id; p = pkt.data; ff_amf_write_string(&p, "publish"); ff_amf_write_number(&p, 0.0); ff_amf_write_null(&p); ff_amf_write_string(&p, rt->playpath); ff_amf_write_string(&p, "live"); ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size, rt->prev_pkt[1]); ff_rtmp_packet_destroy(&pkt); } /** * 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(LOG_CONTEXT, AV_LOG_DEBUG, "Handshaking...\n"); av_lfg_init(&rnd, 0xDEADC0DE); // generate handshake packet - 1536 bytes of pseudorandom data for (i = 9; i <= RTMP_HANDSHAKE_PACKET_SIZE; i++) tosend[i] = av_lfg_get(&rnd) >> 24; client_pos = rtmp_handshake_imprint_with_digest(tosend + 1); url_write(rt->stream, tosend, RTMP_HANDSHAKE_PACKET_SIZE + 1); i = url_read_complete(rt->stream, serverdata, RTMP_HANDSHAKE_PACKET_SIZE + 1); if (i != RTMP_HANDSHAKE_PACKET_SIZE + 1) { av_log(LOG_CONTEXT, AV_LOG_ERROR, "Cannot read RTMP handshake response\n"); return -1; } i = url_read_complete(rt->stream, clientdata, RTMP_HANDSHAKE_PACKET_SIZE); if (i != RTMP_HANDSHAKE_PACKET_SIZE) { av_log(LOG_CONTEXT, AV_LOG_ERROR, "Cannot read RTMP handshake response\n"); return -1; } av_log(LOG_CONTEXT, AV_LOG_DEBUG, "Server version %d.%d.%d.%d\n", serverdata[5], serverdata[6], serverdata[7], serverdata[8]); if (rt->is_input && 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(LOG_CONTEXT, AV_LOG_ERROR, "Server response validating failed\n"); return -1; } } rtmp_calc_digest(tosend + 1 + client_pos, 32, 0, rtmp_server_key, sizeof(rtmp_server_key), digest); rtmp_calc_digest(clientdata, RTMP_HANDSHAKE_PACKET_SIZE-32, 0, digest, 32, digest); if (memcmp(digest, clientdata + RTMP_HANDSHAKE_PACKET_SIZE - 32, 32)) { av_log(LOG_CONTEXT, AV_LOG_ERROR, "Signature mismatch\n"); return -1; } for (i = 0; i < RTMP_HANDSHAKE_PACKET_SIZE; i++) tosend[i] = av_lfg_get(&rnd) >> 24; rtmp_calc_digest(serverdata + 1 + server_pos, 32, 0, rtmp_player_key, sizeof(rtmp_player_key), digest); rtmp_calc_digest(tosend, RTMP_HANDSHAKE_PACKET_SIZE - 32, 0, digest, 32, tosend + RTMP_HANDSHAKE_PACKET_SIZE - 32); // write reply back to the server url_write(rt->stream, tosend, RTMP_HANDSHAKE_PACKET_SIZE); } else { url_write(rt->stream, serverdata+1, RTMP_HANDSHAKE_PACKET_SIZE); } return 0; } /** * 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(LOG_CONTEXT, pkt); #endif switch (pkt->type) { case RTMP_PT_CHUNK_SIZE: if (pkt->data_size != 4) { av_log(LOG_CONTEXT, AV_LOG_ERROR, "Chunk size change packet is not 4 bytes long (%d)\n", pkt->data_size); return -1; } if (!rt->is_input) ff_rtmp_packet_write(rt->stream, pkt, rt->chunk_size, rt->prev_pkt[1]); rt->chunk_size = AV_RB32(pkt->data); if (rt->chunk_size <= 0) { av_log(LOG_CONTEXT, AV_LOG_ERROR, "Incorrect chunk size %d\n", rt->chunk_size); return -1; } av_log(LOG_CONTEXT, AV_LOG_DEBUG, "New chunk size = %d\n", rt->chunk_size); break; case RTMP_PT_PING: t = AV_RB16(pkt->data); if (t == 6) gen_pong(s, rt, pkt); break; case RTMP_PT_CLIENT_BW: if (pkt->data_size < 4) { av_log(LOG_CONTEXT, AV_LOG_ERROR, "Client bandwidth report packet is less than 4 bytes long (%d)\n", pkt->data_size); return -1; } av_log(LOG_CONTEXT, 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(LOG_CONTEXT, AV_LOG_ERROR, "Server error: %s\n",tmpstr); return -1; } else if (!memcmp(pkt->data, "\002\000\007_result", 10)) { switch (rt->state) { case STATE_HANDSHAKED: if (!rt->is_input) { gen_release_stream(s, rt); gen_fcpublish_stream(s, rt); rt->state = STATE_RELEASING; } else { rt->state = STATE_CONNECTING; } gen_create_stream(s, rt); break; case STATE_FCPUBLISH: rt->state = STATE_CONNECTING; break; case STATE_RELEASING: rt->state = STATE_FCPUBLISH; /* hack for Wowza Media Server, it does not send result for * releaseStream and FCPublish calls */ if (!pkt->data[10]) { int pkt_id = (int) av_int2dbl(AV_RB64(pkt->data + 11)); if (pkt_id == 4) rt->state = STATE_CONNECTING; } if (rt->state != STATE_CONNECTING) break; case STATE_CONNECTING: //extract a number from the result if (pkt->data[10] || pkt->data[19] != 5 || pkt->data[20]) { av_log(LOG_CONTEXT, AV_LOG_WARNING, "Unexpected reply on connect()\n"); } else { rt->main_channel_id = (int) av_int2dbl(AV_RB64(pkt->data + 21)); } if (rt->is_input) { gen_play(s, rt); } else { gen_publish(s, rt); } rt->state = STATE_READY; break; } } else if (!memcmp(pkt->data, "\002\000\010onStatus", 11)) { const uint8_t* ptr = pkt->data + 11; uint8_t tmpstr[256]; for (i = 0; i < 2; i++) { t = ff_amf_tag_size(ptr, data_end); if (t < 0) return 1; ptr += t; } t = ff_amf_get_field_value(ptr, data_end, "level", tmpstr, sizeof(tmpstr)); if (!t && !strcmp(tmpstr, "error")) { if (!ff_amf_get_field_value(ptr, data_end, "description", tmpstr, sizeof(tmpstr))) av_log(LOG_CONTEXT, AV_LOG_ERROR, "Server error: %s\n",tmpstr); return -1; } t = ff_amf_get_field_value(ptr, data_end, "code", tmpstr, sizeof(tmpstr)); if (!t && !strcmp(tmpstr, "NetStream.Play.Start")) rt->state = STATE_PLAYING; if (!t && !strcmp(tmpstr, "NetStream.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; 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(LOG_CONTEXT, 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); url_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 & URL_WRONLY); 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 (url_open(&rt->stream, buf, URL_RDWR) < 0) { av_log(LOG_CONTEXT, AV_LOG_ERROR, "Cannot open connection %s\n", buf); goto fail; } rt->state = STATE_START; if (rtmp_handshake(s, rt)) return -1; rt->chunk_size = 128; rt->state = STATE_HANDSHAKED; //extract "app" part from path if (!strncmp(path, "/ondemand/", 10)) { fname = path + 10; memcpy(rt->app, "ondemand", 9); } else { char *p = strchr(path + 1, '/'); if (!p) { fname = path + 1; rt->app[0] = '\0'; } else { char *c = strchr(p + 1, ':'); fname = strchr(p + 1, '/'); if (!fname || c < fname) { fname = p + 1; av_strlcpy(rt->app, path + 1, p - path); } else { fname++; av_strlcpy(rt->app, path + 1, fname - path - 1); } } } if (!strchr(fname, ':') && (!strcmp(fname + strlen(fname) - 4, ".f4v") || !strcmp(fname + strlen(fname) - 4, ".mp4"))) { memcpy(rt->playpath, "mp4:", 5); } else { rt->playpath[0] = 0; } strncat(rt->playpath, fname, sizeof(rt->playpath) - 5); rt->client_report_size = 1048576; rt->bytes_read = 0; rt->last_bytes_read = 0; av_log(LOG_CONTEXT, AV_LOG_DEBUG, "Proto = %s, path = %s, app = %s, fname = %s\n", proto, path, rt->app, rt->playpath); gen_connect(s, rt, proto, hostname, port); do { ret = get_packet(s, 1); } while (ret == EAGAIN); if (ret < 0) goto fail; if (rt->is_input) { // generate FLV header for demuxer rt->flv_size = 13; rt->flv_data = av_realloc(rt->flv_data, rt->flv_size); rt->flv_off = 0; memcpy(rt->flv_data, "FLV\1\5\0\0\0\011\0\0\0\0", rt->flv_size); } else { rt->flv_size = 0; rt->flv_data = NULL; rt->flv_off = 0; } s->max_packet_size = url_get_max_packet_size(rt->stream); s->is_streamed = 1; return 0; fail: rtmp_close(s); return AVERROR(EIO); } static int rtmp_read(URLContext *s, uint8_t *buf, int size) { RTMPContext *rt = s->priv_data; int orig_size = size; int ret; while (size > 0) { int data_left = rt->flv_size - rt->flv_off; if (data_left >= size) { memcpy(buf, rt->flv_data + rt->flv_off, size); rt->flv_off += size; return orig_size; } if (data_left > 0) { memcpy(buf, rt->flv_data + rt->flv_off, data_left); buf += data_left; size -= data_left; rt->flv_off = rt->flv_size; return data_left; } if ((ret = get_packet(s, 0)) < 0) return ret; } return orig_size; } static int rtmp_write(URLContext *h, const uint8_t *buf, int size) { RTMPContext *rt = h->priv_data; int size_temp = size; int pktsize, pkttype; uint32_t ts; const uint8_t *buf_temp = buf; if (size < 11) { av_log(LOG_CONTEXT, AV_LOG_DEBUG, "FLV packet too small %d\n", size); return 0; } do { if (!rt->flv_off) { //skip flv header if (buf_temp[0] == 'F' && buf_temp[1] == 'L' && buf_temp[2] == 'V') { buf_temp += 9 + 4; size_temp -= 9 + 4; } pkttype = bytestream_get_byte(&buf_temp); pktsize = bytestream_get_be24(&buf_temp); ts = bytestream_get_be24(&buf_temp); ts |= bytestream_get_byte(&buf_temp) << 24; bytestream_get_be24(&buf_temp); size_temp -= 11; rt->flv_size = pktsize; //force 12bytes header if (((pkttype == RTMP_PT_VIDEO || pkttype == RTMP_PT_AUDIO) && ts == 0) || pkttype == RTMP_PT_NOTIFY) { if (pkttype == RTMP_PT_NOTIFY) pktsize += 16; rt->prev_pkt[1][RTMP_SOURCE_CHANNEL].channel_id = 0; } //this can be a big packet, it's better to send it right here ff_rtmp_packet_create(&rt->out_pkt, RTMP_SOURCE_CHANNEL, pkttype, ts, pktsize); rt->out_pkt.extra = rt->main_channel_id; rt->flv_data = rt->out_pkt.data; if (pkttype == RTMP_PT_NOTIFY) ff_amf_write_string(&rt->flv_data, "@setDataFrame"); } if (rt->flv_size - rt->flv_off > size_temp) { bytestream_get_buffer(&buf_temp, rt->flv_data + rt->flv_off, size_temp); rt->flv_off += size_temp; } else { bytestream_get_buffer(&buf_temp, rt->flv_data + rt->flv_off, rt->flv_size - rt->flv_off); rt->flv_off += rt->flv_size - rt->flv_off; } if (rt->flv_off == rt->flv_size) { bytestream_get_be32(&buf_temp); ff_rtmp_packet_write(rt->stream, &rt->out_pkt, rt->chunk_size, rt->prev_pkt[1]); ff_rtmp_packet_destroy(&rt->out_pkt); rt->flv_size = 0; rt->flv_off = 0; } } while (buf_temp - buf < size_temp); return size; } URLProtocol rtmp_protocol = { "rtmp", rtmp_open, rtmp_read, rtmp_write, NULL, /* seek */ rtmp_close, };