X-Git-Url: https://git.sesse.net/?a=blobdiff_plain;f=libavformat%2Frtmpproto.c;h=5c40eb5517b948e1d7de5db666742fbcc3e9e984;hb=f5a9c35f886508b851011b7dd4ec18cc67b57d37;hp=8148fbff19d3a1dc59f11449d30911997a5dc946;hpb=e6b244a3b986e513779aec83beab4fb25a130aa2;p=ffmpeg diff --git a/libavformat/rtmpproto.c b/libavformat/rtmpproto.c index 8148fbff19d..5c40eb5517b 100644 --- a/libavformat/rtmpproto.c +++ b/libavformat/rtmpproto.c @@ -2,67 +2,96 @@ * RTMP network protocol * Copyright (c) 2009 Kostya Shishkov * - * This file is part of FFmpeg. + * This file is part of Libav. * - * FFmpeg is free software; you can redistribute it and/or + * Libav 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, + * Libav 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 + * License along with Libav; if not, write to the Free Software * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA */ /** - * @file libavformat/rtmpproto.c + * @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 "rtmppkt.h" +#include "url.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 + +#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 - char playpath[256]; ///< path to filename to play (with possible "mp4:" prefix) + 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 + 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) } RTMPContext; #define PLAYER_KEY_OPEN_PART_LEN 30 ///< length of partial key used for first client digest signing @@ -88,119 +117,477 @@ static const uint8_t rtmp_server_key[] = { 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); +} + /** - * Generates 'connect' call and sends it to the server. + * 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, const char *app) +static int gen_connect(URLContext *s, RTMPContext *rt) { RTMPPacket pkt; - uint8_t ver[32], *p; - char tcurl[512]; + uint8_t *p; + int ret; + + if ((ret = ff_rtmp_packet_create(&pkt, RTMP_SYSTEM_CHANNEL, RTMP_PT_INVOKE, + 0, 4096)) < 0) + return ret; - ff_rtmp_packet_create(&pkt, RTMP_VIDEO_CHANNEL, RTMP_PT_INVOKE, 0, 4096); p = pkt.data; - snprintf(tcurl, sizeof(tcurl), "%s://%s:%d/%s", proto, host, port, app); ff_amf_write_string(&p, "connect"); - ff_amf_write_number(&p, 1.0); + 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, app); + ff_amf_write_string(&p, rt->app); - snprintf(ver, sizeof(ver), "%s %d,%d,%d,%d", RTMP_CLIENT_PLATFORM, RTMP_CLIENT_VER1, - RTMP_CLIENT_VER2, RTMP_CLIENT_VER3, RTMP_CLIENT_VER4); + 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, ver); + 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, tcurl); - 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_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); + } 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; - ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size, rt->prev_pkt[1]); + ret = ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size, + rt->prev_pkt[1]); + ff_rtmp_packet_destroy(&pkt); + + return ret; } /** - * Generates 'createStream' call and sends it to the server. It should make + * 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 void gen_create_stream(URLContext *s, RTMPContext *rt) +static int gen_create_stream(URLContext *s, RTMPContext *rt) { RTMPPacket pkt; uint8_t *p; + int ret; - av_log(LOG_CONTEXT, AV_LOG_DEBUG, "Creating stream...\n"); - ff_rtmp_packet_create(&pkt, RTMP_VIDEO_CHANNEL, RTMP_PT_INVOKE, 0, 25); + 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, 3.0); + 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); - ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size, rt->prev_pkt[1]); + ret = ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size, + rt->prev_pkt[1]); ff_rtmp_packet_destroy(&pkt); + + return ret; } /** - * Generates 'play' call and sends it to the server, then pings the server + * 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) +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; - 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_number(&p, ++rt->nb_invokes); ff_amf_write_null(&p); ff_amf_write_string(&p, rt->playpath); + ff_amf_write_number(&p, rt->live); - ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size, rt->prev_pkt[1]); + ret = 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); + 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; - bytestream_put_be16(&p, 3); - bytestream_put_be32(&p, 1); - bytestream_put_be32(&p, 256); //TODO: what is a good value here? + 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"); - ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size, rt->prev_pkt[1]); + ret = ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size, + rt->prev_pkt[1]); ff_rtmp_packet_destroy(&pkt); + + return ret; } /** - * Generates ping reply and sends it to the server. + * Generate ping reply and send it to the server. */ -static void gen_pong(URLContext *s, RTMPContext *rt, RTMPPacket *ppkt) +static int gen_pong(URLContext *s, RTMPContext *rt, RTMPPacket *ppkt) { RTMPPacket pkt; uint8_t *p; + int ret; + + if ((ret = ff_rtmp_packet_create(&pkt, RTMP_NETWORK_CHANNEL, RTMP_PT_PING, + ppkt->timestamp + 1, 6)) < 0) + return ret; - ff_rtmp_packet_create(&pkt, RTMP_NETWORK_CHANNEL, RTMP_PT_PING, ppkt->timestamp + 1, 6); p = pkt.data; bytestream_put_be16(&p, 7); - bytestream_put_be32(&p, AV_RB32(ppkt->data+2) + 1); - ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size, rt->prev_pkt[1]); + 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; } //TODO: Move HMAC code somewhere. Eventually. @@ -208,7 +595,7 @@ static void gen_pong(URLContext *s, RTMPContext *rt, RTMPPacket *ppkt) #define HMAC_OPAD_VAL 0x5C /** - * Calculates HMAC-SHA2 digest for RTMP handshake packets. + * Calculate HMAC-SHA2 digest for RTMP handshake packets. * * @param src input buffer * @param len input buffer length (should be 1536) @@ -218,14 +605,16 @@ static void gen_pong(URLContext *s, RTMPContext *rt, RTMPPacket *ppkt) * @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) +static int 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); @@ -254,10 +643,12 @@ static void rtmp_calc_digest(const uint8_t *src, int len, int gap, av_sha_final(sha, dst); av_free(sha); + + return 0; } /** - * Puts HMAC-SHA2 digest of packet data (except for the bytes where this digest + * 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) @@ -266,19 +657,23 @@ static void rtmp_calc_digest(const uint8_t *src, int len, int gap, static int rtmp_handshake_imprint_with_digest(uint8_t *buf) { int i, digest_pos = 0; + int ret; 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); + ret = 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; } /** - * Verifies that the received server response has the expected digest value. + * 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 @@ -288,21 +683,25 @@ static int rtmp_validate_digest(uint8_t *buf, int off) { int i, digest_pos = 0; uint8_t digest[32]; + int ret; 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); + ret = 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; } /** - * Performs handshake with the server by means of exchanging pseudorandom data + * Perform handshake with the server by means of exchanging pseudorandom data * signed with HMAC-SHA2 digest. * * @return 0 if handshake succeeds, negative value otherwise @@ -323,66 +722,99 @@ static int rtmp_handshake(URLContext *s, RTMPContext *rt) int i; int server_pos, client_pos; uint8_t digest[32]; + int ret; - av_log(LOG_CONTEXT, AV_LOG_DEBUG, "Handshaking...\n"); + 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); + if (client_pos < 0) + return client_pos; - 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; + 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; } - 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; + + 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(LOG_CONTEXT, AV_LOG_DEBUG, "Server version %d.%d.%d.%d\n", + av_log(s, AV_LOG_DEBUG, "Server version %d.%d.%d.%d\n", serverdata[5], serverdata[6], serverdata[7], serverdata[8]); - server_pos = rtmp_validate_digest(serverdata + 1, 772); - if (!server_pos) { - server_pos = rtmp_validate_digest(serverdata + 1, 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) { - av_log(LOG_CONTEXT, AV_LOG_ERROR, "Server response validating failed\n"); - return -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); + } } - } - 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; + ret = rtmp_calc_digest(tosend + 1 + client_pos, 32, 0, rtmp_server_key, + sizeof(rtmp_server_key), digest); + if (ret < 0) + return ret; + + ret = rtmp_calc_digest(clientdata, RTMP_HANDSHAKE_PACKET_SIZE - 32, 0, + digest, 32, digest); + if (ret < 0) + return ret; + + if (memcmp(digest, 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 = rtmp_calc_digest(serverdata + 1 + server_pos, 32, 0, + rtmp_player_key, sizeof(rtmp_player_key), + digest); + if (ret < 0) + return ret; + + ret = rtmp_calc_digest(tosend, RTMP_HANDSHAKE_PACKET_SIZE - 32, 0, + digest, 32, + tosend + RTMP_HANDSHAKE_PACKET_SIZE - 32); + if (ret < 0) + return ret; + + // write reply back to the server + if ((ret = ffurl_write(rt->stream, tosend, + RTMP_HANDSHAKE_PACKET_SIZE)) < 0) + return ret; + } else { + if ((ret = ffurl_write(rt->stream, serverdata + 1, + RTMP_HANDSHAKE_PACKET_SIZE)) < 0) + return ret; } - 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); return 0; } /** - * Parses received packet and may perform some action depending on + * 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 @@ -391,25 +823,53 @@ static int rtmp_parse_result(URLContext *s, RTMPContext *rt, RTMPPacket *pkt) { int i, t; const uint8_t *data_end = pkt->data + pkt->data_size; + int ret; + +#ifdef DEBUG + ff_rtmp_packet_dump(s, pkt); +#endif switch (pkt->type) { case RTMP_PT_CHUNK_SIZE: if (pkt->data_size != 4) { - av_log(LOG_CONTEXT, AV_LOG_ERROR, + 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) + 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(LOG_CONTEXT, AV_LOG_ERROR, "Incorrect chunk size %d\n", rt->chunk_size); + av_log(s, 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); + 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); + if ((ret = gen_pong(s, rt, pkt)) < 0) + return ret; + 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_SERVER_BW: + 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(EINVAL); + } + av_log(s, AV_LOG_DEBUG, "Server bandwidth = %d\n", rt->server_bw); break; case RTMP_PT_INVOKE: //TODO: check for the messages sent for wrong state? @@ -418,29 +878,61 @@ static int rtmp_parse_result(URLContext *s, RTMPContext *rt, RTMPPacket *pkt) 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); + 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: - gen_create_stream(s, rt); + 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(LOG_CONTEXT, AV_LOG_WARNING, "Unexpected reply on connect()\n"); + av_log(s, AV_LOG_WARNING, "Unexpected reply on connect()\n"); } else { - rt->main_channel_id = (int) av_int2dbl(AV_RB64(pkt->data + 21)); + 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; } - gen_play(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]; - int t; for (i = 0; i < 2; i++) { t = ff_amf_tag_size(ptr, data_end); @@ -453,23 +945,33 @@ static int rtmp_parse_result(URLContext *s, RTMPContext *rt, RTMPPacket *pkt) 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); + 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; - return 0; - } + 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; } break; + case RTMP_PT_VIDEO: + case RTMP_PT_AUDIO: + /* Audio and Video 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; } /** - * Interacts with the server by receiving and sending RTMP packets until + * 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 @@ -483,35 +985,52 @@ 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; - for(;;) { - RTMPPacket rpkt; + 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) { + 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 -1; + return ret; + } + if (rt->state == STATE_STOPPED) { + ff_rtmp_packet_destroy(&rpkt); + return AVERROR_EOF; } - if (for_header && rt->state == STATE_PLAYING) { + if (for_header && (rt->state == STATE_PLAYING || rt->state == STATE_PUBLISHING)) { ff_rtmp_packet_destroy(&rpkt); return 0; } - if (!rpkt.data_size) { + if (!rpkt.data_size || !rt->is_input) { ff_rtmp_packet_destroy(&rpkt); continue; } if (rpkt.type == RTMP_PT_VIDEO || rpkt.type == RTMP_PT_AUDIO || (rpkt.type == RTMP_PT_NOTIFY && !memcmp("\002\000\012onMetaData", rpkt.data, 13))) { - uint8_t *p; - uint32_t ts = rpkt.timestamp; + ts = rpkt.timestamp; // generate packet header and put data into buffer for FLV demuxer rt->flv_off = 0; @@ -531,27 +1050,53 @@ static int get_packet(URLContext *s, int for_header) 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; + 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); - url_close(rt->stream); - av_free(rt); - return 0; + ffurl_close(rt->stream); + return ret; } /** - * Opens RTMP connection and verifies that the stream can be played. + * 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 @@ -561,94 +1106,173 @@ static int rtmp_close(URLContext *h) */ static int rtmp_open(URLContext *s, const char *uri, int flags) { - RTMPContext *rt; - char proto[8], hostname[256], path[1024], app[128], *fname; + RTMPContext *rt = s->priv_data; + char proto[8], hostname[256], path[1024], *fname; + char *old_app; uint8_t buf[2048]; - int port, is_input; + int port; + AVDictionary *opts = NULL; int ret; - is_input = !(flags & URL_WRONLY); + rt->is_input = !(flags & AVIO_FLAG_WRITE); - rt = av_mallocz(sizeof(RTMPContext)); - if (!rt) - return AVERROR(ENOMEM); - s->priv_data = rt; + av_url_split(proto, sizeof(proto), NULL, 0, hostname, sizeof(hostname), &port, + path, sizeof(path), s->filename); - 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); - if (port < 0) - port = RTMP_DEFAULT_PORT; - snprintf(buf, sizeof(buf), "tcp://%s:%d", hostname, port); + /* 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 { + /* open the tcp connection */ + 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); + 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; } - if (!is_input) { - av_log(LOG_CONTEXT, AV_LOG_ERROR, "RTMP output is not supported yet.\n"); + rt->state = STATE_START; + if ((ret = rtmp_handshake(s, rt)) < 0) goto fail; - } else { - 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(app, "ondemand", 9); + 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 { - char *p = strchr(path + 1, '/'); - if (!p) { - fname = path + 1; - app[0] = '\0'; + // 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 { - char *c = strchr(p + 1, ':'); - fname = strchr(p + 1, '/'); - if (!fname || c < fname) { - fname = p + 1; - av_strlcpy(app, path + 1, p - path); - } else { - fname++; - av_strlcpy(app, path + 1, fname - path - 1); - } + 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"))) { + } + + 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, sizeof(rt->playpath) - 5); + strncat(rt->playpath, fname, PLAYPATH_MAX_LENGTH - 5); + } - av_log(LOG_CONTEXT, AV_LOG_DEBUG, "Proto = %s, path = %s, app = %s, fname = %s\n", - proto, path, app, rt->playpath); - gen_connect(s, rt, proto, hostname, port, app); + 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); + } - do { - ret = get_packet(s, 1); - } while (ret == EAGAIN); - if (ret < 0) + 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 = url_get_max_packet_size(rt->stream); + s->max_packet_size = rt->stream->max_packet_size; s->is_streamed = 1; return 0; fail: + av_dict_free(&opts); rtmp_close(s); - return AVERROR(EIO); + return ret; } static int rtmp_read(URLContext *s, uint8_t *buf, int size) @@ -670,6 +1294,7 @@ static int rtmp_read(URLContext *s, uint8_t *buf, int size) buf += data_left; size -= data_left; rt->flv_off = rt->flv_size; + return data_left; } if ((ret = get_packet(s, 0)) < 0) return ret; @@ -677,16 +1302,208 @@ static int rtmp_read(URLContext *s, uint8_t *buf, int size) return orig_size; } -static int rtmp_write(URLContext *h, uint8_t *buf, int size) +static int rtmp_write(URLContext *s, const uint8_t *buf, int size) { - return 0; + 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; } -URLProtocol rtmp_protocol = { - "rtmp", - rtmp_open, - rtmp_read, - rtmp_write, - NULL, /* seek */ - rtmp_close, +#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_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 rtmp://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 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 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, };