mirror of https://git.ffmpeg.org/ffmpeg.git
1707 lines
55 KiB
C
1707 lines
55 KiB
C
/*
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* RTMP network protocol
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* Copyright (c) 2009 Kostya Shishkov
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*
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* This file is part of FFmpeg.
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*
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* FFmpeg is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Lesser General Public
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* License as published by the Free Software Foundation; either
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* version 2.1 of the License, or (at your option) any later version.
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*
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* FFmpeg is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Lesser General Public License for more details.
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*
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* You should have received a copy of the GNU Lesser General Public
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* License along with FFmpeg; if not, write to the Free Software
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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*/
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/**
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* @file
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* RTMP protocol
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*/
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#include "libavcodec/bytestream.h"
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#include "libavutil/avstring.h"
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#include "libavutil/intfloat.h"
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#include "libavutil/lfg.h"
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#include "libavutil/opt.h"
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#include "libavutil/sha.h"
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#include "avformat.h"
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#include "internal.h"
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#include "network.h"
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#include "flv.h"
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#include "rtmp.h"
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#include "rtmpcrypt.h"
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#include "rtmppkt.h"
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#include "url.h"
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//#define DEBUG
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#define APP_MAX_LENGTH 128
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#define PLAYPATH_MAX_LENGTH 256
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#define TCURL_MAX_LENGTH 512
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#define FLASHVER_MAX_LENGTH 64
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/** RTMP protocol handler state */
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typedef enum {
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STATE_START, ///< client has not done anything yet
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STATE_HANDSHAKED, ///< client has performed handshake
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STATE_RELEASING, ///< client releasing stream before publish it (for output)
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STATE_FCPUBLISH, ///< client FCPublishing stream (for output)
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STATE_CONNECTING, ///< client connected to server successfully
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STATE_READY, ///< client has sent all needed commands and waits for server reply
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STATE_PLAYING, ///< client has started receiving multimedia data from server
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STATE_PUBLISHING, ///< client has started sending multimedia data to server (for output)
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STATE_STOPPED, ///< the broadcast has been stopped
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} ClientState;
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/** protocol handler context */
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typedef struct RTMPContext {
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const AVClass *class;
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URLContext* stream; ///< TCP stream used in interactions with RTMP server
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RTMPPacket prev_pkt[2][RTMP_CHANNELS]; ///< packet history used when reading and sending packets
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int chunk_size; ///< size of the chunks RTMP packets are divided into
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int is_input; ///< input/output flag
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char *playpath; ///< stream identifier to play (with possible "mp4:" prefix)
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int live; ///< 0: recorded, -1: live, -2: both
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char *app; ///< name of application
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char *conn; ///< append arbitrary AMF data to the Connect message
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ClientState state; ///< current state
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int main_channel_id; ///< an additional channel ID which is used for some invocations
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uint8_t* flv_data; ///< buffer with data for demuxer
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int flv_size; ///< current buffer size
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int flv_off; ///< number of bytes read from current buffer
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int flv_nb_packets; ///< number of flv packets published
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RTMPPacket out_pkt; ///< rtmp packet, created from flv a/v or metadata (for output)
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uint32_t client_report_size; ///< number of bytes after which client should report to server
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uint32_t bytes_read; ///< number of bytes read from server
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uint32_t last_bytes_read; ///< number of bytes read last reported to server
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int skip_bytes; ///< number of bytes to skip from the input FLV stream in the next write call
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uint8_t flv_header[11]; ///< partial incoming flv packet header
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int flv_header_bytes; ///< number of initialized bytes in flv_header
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int nb_invokes; ///< keeps track of invoke messages
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int create_stream_invoke; ///< invoke id for the create stream command
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char* tcurl; ///< url of the target stream
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char* flashver; ///< version of the flash plugin
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char* swfurl; ///< url of the swf player
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char* pageurl; ///< url of the web page
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int server_bw; ///< server bandwidth
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int client_buffer_time; ///< client buffer time in ms
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int flush_interval; ///< number of packets flushed in the same request (RTMPT only)
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int encrypted; ///< use an encrypted connection (RTMPE only)
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} RTMPContext;
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#define PLAYER_KEY_OPEN_PART_LEN 30 ///< length of partial key used for first client digest signing
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/** Client key used for digest signing */
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static const uint8_t rtmp_player_key[] = {
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'G', 'e', 'n', 'u', 'i', 'n', 'e', ' ', 'A', 'd', 'o', 'b', 'e', ' ',
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'F', 'l', 'a', 's', 'h', ' ', 'P', 'l', 'a', 'y', 'e', 'r', ' ', '0', '0', '1',
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0xF0, 0xEE, 0xC2, 0x4A, 0x80, 0x68, 0xBE, 0xE8, 0x2E, 0x00, 0xD0, 0xD1, 0x02,
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0x9E, 0x7E, 0x57, 0x6E, 0xEC, 0x5D, 0x2D, 0x29, 0x80, 0x6F, 0xAB, 0x93, 0xB8,
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0xE6, 0x36, 0xCF, 0xEB, 0x31, 0xAE
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};
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#define SERVER_KEY_OPEN_PART_LEN 36 ///< length of partial key used for first server digest signing
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/** Key used for RTMP server digest signing */
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static const uint8_t rtmp_server_key[] = {
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'G', 'e', 'n', 'u', 'i', 'n', 'e', ' ', 'A', 'd', 'o', 'b', 'e', ' ',
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'F', 'l', 'a', 's', 'h', ' ', 'M', 'e', 'd', 'i', 'a', ' ',
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'S', 'e', 'r', 'v', 'e', 'r', ' ', '0', '0', '1',
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0xF0, 0xEE, 0xC2, 0x4A, 0x80, 0x68, 0xBE, 0xE8, 0x2E, 0x00, 0xD0, 0xD1, 0x02,
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0x9E, 0x7E, 0x57, 0x6E, 0xEC, 0x5D, 0x2D, 0x29, 0x80, 0x6F, 0xAB, 0x93, 0xB8,
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0xE6, 0x36, 0xCF, 0xEB, 0x31, 0xAE
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};
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static int rtmp_write_amf_data(URLContext *s, char *param, uint8_t **p)
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{
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char *field, *value;
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char type;
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/* The type must be B for Boolean, N for number, S for string, O for
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* object, or Z for null. For Booleans the data must be either 0 or 1 for
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* FALSE or TRUE, respectively. Likewise for Objects the data must be
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* 0 or 1 to end or begin an object, respectively. Data items in subobjects
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* may be named, by prefixing the type with 'N' and specifying the name
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* before the value (ie. NB:myFlag:1). This option may be used multiple times
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* to construct arbitrary AMF sequences. */
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if (param[0] && param[1] == ':') {
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type = param[0];
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value = param + 2;
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} else if (param[0] == 'N' && param[1] && param[2] == ':') {
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type = param[1];
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field = param + 3;
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value = strchr(field, ':');
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if (!value)
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goto fail;
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*value = '\0';
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value++;
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if (!field || !value)
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goto fail;
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ff_amf_write_field_name(p, field);
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} else {
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goto fail;
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}
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switch (type) {
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case 'B':
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ff_amf_write_bool(p, value[0] != '0');
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break;
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case 'S':
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ff_amf_write_string(p, value);
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break;
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case 'N':
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ff_amf_write_number(p, strtod(value, NULL));
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break;
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case 'Z':
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ff_amf_write_null(p);
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break;
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case 'O':
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if (value[0] != '0')
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ff_amf_write_object_start(p);
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else
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ff_amf_write_object_end(p);
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break;
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default:
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goto fail;
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break;
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}
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return 0;
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fail:
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av_log(s, AV_LOG_ERROR, "Invalid AMF parameter: %s\n", param);
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return AVERROR(EINVAL);
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}
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/**
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* Generate 'connect' call and send it to the server.
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*/
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static int gen_connect(URLContext *s, RTMPContext *rt)
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{
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RTMPPacket pkt;
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uint8_t *p;
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int ret;
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if ((ret = ff_rtmp_packet_create(&pkt, RTMP_SYSTEM_CHANNEL, RTMP_PT_INVOKE,
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0, 4096)) < 0)
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return ret;
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p = pkt.data;
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ff_amf_write_string(&p, "connect");
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ff_amf_write_number(&p, ++rt->nb_invokes);
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ff_amf_write_object_start(&p);
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ff_amf_write_field_name(&p, "app");
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ff_amf_write_string(&p, rt->app);
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if (!rt->is_input) {
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ff_amf_write_field_name(&p, "type");
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ff_amf_write_string(&p, "nonprivate");
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}
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ff_amf_write_field_name(&p, "flashVer");
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ff_amf_write_string(&p, rt->flashver);
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if (rt->swfurl) {
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ff_amf_write_field_name(&p, "swfUrl");
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ff_amf_write_string(&p, rt->swfurl);
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}
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ff_amf_write_field_name(&p, "tcUrl");
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ff_amf_write_string(&p, rt->tcurl);
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if (rt->is_input) {
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ff_amf_write_field_name(&p, "fpad");
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ff_amf_write_bool(&p, 0);
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ff_amf_write_field_name(&p, "capabilities");
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ff_amf_write_number(&p, 15.0);
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/* Tell the server we support all the audio codecs except
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* SUPPORT_SND_INTEL (0x0008) and SUPPORT_SND_UNUSED (0x0010)
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* which are unused in the RTMP protocol implementation. */
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ff_amf_write_field_name(&p, "audioCodecs");
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ff_amf_write_number(&p, 4071.0);
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ff_amf_write_field_name(&p, "videoCodecs");
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ff_amf_write_number(&p, 252.0);
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ff_amf_write_field_name(&p, "videoFunction");
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ff_amf_write_number(&p, 1.0);
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if (rt->pageurl) {
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ff_amf_write_field_name(&p, "pageUrl");
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ff_amf_write_string(&p, rt->pageurl);
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}
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}
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ff_amf_write_object_end(&p);
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if (rt->conn) {
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char *param = rt->conn;
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// Write arbitrary AMF data to the Connect message.
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while (param != NULL) {
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char *sep;
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param += strspn(param, " ");
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if (!*param)
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break;
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sep = strchr(param, ' ');
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if (sep)
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*sep = '\0';
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if ((ret = rtmp_write_amf_data(s, param, &p)) < 0) {
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// Invalid AMF parameter.
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ff_rtmp_packet_destroy(&pkt);
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return ret;
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}
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if (sep)
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param = sep + 1;
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else
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break;
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}
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}
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pkt.data_size = p - pkt.data;
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ret = ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size,
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rt->prev_pkt[1]);
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ff_rtmp_packet_destroy(&pkt);
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return ret;
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}
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/**
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* Generate 'releaseStream' call and send it to the server. It should make
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* the server release some channel for media streams.
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*/
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static int gen_release_stream(URLContext *s, RTMPContext *rt)
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{
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RTMPPacket pkt;
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uint8_t *p;
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int ret;
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if ((ret = ff_rtmp_packet_create(&pkt, RTMP_SYSTEM_CHANNEL, RTMP_PT_INVOKE,
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0, 29 + strlen(rt->playpath))) < 0)
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return ret;
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av_log(s, AV_LOG_DEBUG, "Releasing stream...\n");
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p = pkt.data;
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ff_amf_write_string(&p, "releaseStream");
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ff_amf_write_number(&p, ++rt->nb_invokes);
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ff_amf_write_null(&p);
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ff_amf_write_string(&p, rt->playpath);
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ret = ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size,
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rt->prev_pkt[1]);
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ff_rtmp_packet_destroy(&pkt);
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return ret;
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}
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/**
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* Generate 'FCPublish' call and send it to the server. It should make
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* the server preapare for receiving media streams.
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*/
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static int gen_fcpublish_stream(URLContext *s, RTMPContext *rt)
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{
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RTMPPacket pkt;
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uint8_t *p;
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int ret;
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if ((ret = ff_rtmp_packet_create(&pkt, RTMP_SYSTEM_CHANNEL, RTMP_PT_INVOKE,
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0, 25 + strlen(rt->playpath))) < 0)
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return ret;
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av_log(s, AV_LOG_DEBUG, "FCPublish stream...\n");
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p = pkt.data;
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ff_amf_write_string(&p, "FCPublish");
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ff_amf_write_number(&p, ++rt->nb_invokes);
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ff_amf_write_null(&p);
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ff_amf_write_string(&p, rt->playpath);
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ret = ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size,
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rt->prev_pkt[1]);
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ff_rtmp_packet_destroy(&pkt);
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return ret;
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}
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/**
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* Generate 'FCUnpublish' call and send it to the server. It should make
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* the server destroy stream.
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*/
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static int gen_fcunpublish_stream(URLContext *s, RTMPContext *rt)
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{
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RTMPPacket pkt;
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uint8_t *p;
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int ret;
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if ((ret = ff_rtmp_packet_create(&pkt, RTMP_SYSTEM_CHANNEL, RTMP_PT_INVOKE,
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0, 27 + strlen(rt->playpath))) < 0)
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return ret;
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av_log(s, AV_LOG_DEBUG, "UnPublishing stream...\n");
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p = pkt.data;
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ff_amf_write_string(&p, "FCUnpublish");
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ff_amf_write_number(&p, ++rt->nb_invokes);
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ff_amf_write_null(&p);
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ff_amf_write_string(&p, rt->playpath);
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ret = ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size,
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rt->prev_pkt[1]);
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ff_rtmp_packet_destroy(&pkt);
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return ret;
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}
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/**
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* Generate 'createStream' call and send it to the server. It should make
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* the server allocate some channel for media streams.
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*/
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static int gen_create_stream(URLContext *s, RTMPContext *rt)
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{
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RTMPPacket pkt;
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uint8_t *p;
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int ret;
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av_log(s, AV_LOG_DEBUG, "Creating stream...\n");
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if ((ret = ff_rtmp_packet_create(&pkt, RTMP_SYSTEM_CHANNEL, RTMP_PT_INVOKE,
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0, 25)) < 0)
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return ret;
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p = pkt.data;
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ff_amf_write_string(&p, "createStream");
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ff_amf_write_number(&p, ++rt->nb_invokes);
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ff_amf_write_null(&p);
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rt->create_stream_invoke = rt->nb_invokes;
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ret = ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size,
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rt->prev_pkt[1]);
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ff_rtmp_packet_destroy(&pkt);
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return ret;
|
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}
|
|
|
|
|
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/**
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* Generate 'deleteStream' call and send it to the server. It should make
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* the server remove some channel for media streams.
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*/
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static int gen_delete_stream(URLContext *s, RTMPContext *rt)
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{
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RTMPPacket pkt;
|
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uint8_t *p;
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int ret;
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|
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av_log(s, AV_LOG_DEBUG, "Deleting stream...\n");
|
|
|
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if ((ret = ff_rtmp_packet_create(&pkt, RTMP_SYSTEM_CHANNEL, RTMP_PT_INVOKE,
|
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0, 34)) < 0)
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return ret;
|
|
|
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p = pkt.data;
|
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ff_amf_write_string(&p, "deleteStream");
|
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ff_amf_write_number(&p, ++rt->nb_invokes);
|
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ff_amf_write_null(&p);
|
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ff_amf_write_number(&p, rt->main_channel_id);
|
|
|
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ret = ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size,
|
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rt->prev_pkt[1]);
|
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ff_rtmp_packet_destroy(&pkt);
|
|
|
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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);
|
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|
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ret = ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size,
|
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rt->prev_pkt[1]);
|
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ff_rtmp_packet_destroy(&pkt);
|
|
|
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return ret;
|
|
}
|
|
|
|
/**
|
|
* Generate 'play' call and send it to the server, then ping the server
|
|
* to start actual playing.
|
|
*/
|
|
static int gen_play(URLContext *s, RTMPContext *rt)
|
|
{
|
|
RTMPPacket pkt;
|
|
uint8_t *p;
|
|
int ret;
|
|
|
|
av_log(s, AV_LOG_DEBUG, "Sending play command for '%s'\n", rt->playpath);
|
|
|
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if ((ret = ff_rtmp_packet_create(&pkt, RTMP_VIDEO_CHANNEL, RTMP_PT_INVOKE,
|
|
0, 29 + strlen(rt->playpath))) < 0)
|
|
return ret;
|
|
|
|
pkt.extra = rt->main_channel_id;
|
|
|
|
p = pkt.data;
|
|
ff_amf_write_string(&p, "play");
|
|
ff_amf_write_number(&p, ++rt->nb_invokes);
|
|
ff_amf_write_null(&p);
|
|
ff_amf_write_string(&p, rt->playpath);
|
|
ff_amf_write_number(&p, rt->live);
|
|
|
|
ret = ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size,
|
|
rt->prev_pkt[1]);
|
|
ff_rtmp_packet_destroy(&pkt);
|
|
|
|
return ret;
|
|
}
|
|
|
|
/**
|
|
* Generate 'publish' call and send it to the server.
|
|
*/
|
|
static int gen_publish(URLContext *s, RTMPContext *rt)
|
|
{
|
|
RTMPPacket pkt;
|
|
uint8_t *p;
|
|
int ret;
|
|
|
|
av_log(s, AV_LOG_DEBUG, "Sending publish command for '%s'\n", rt->playpath);
|
|
|
|
if ((ret = ff_rtmp_packet_create(&pkt, RTMP_SOURCE_CHANNEL, RTMP_PT_INVOKE,
|
|
0, 30 + strlen(rt->playpath))) < 0)
|
|
return ret;
|
|
|
|
pkt.extra = rt->main_channel_id;
|
|
|
|
p = pkt.data;
|
|
ff_amf_write_string(&p, "publish");
|
|
ff_amf_write_number(&p, ++rt->nb_invokes);
|
|
ff_amf_write_null(&p);
|
|
ff_amf_write_string(&p, rt->playpath);
|
|
ff_amf_write_string(&p, "live");
|
|
|
|
ret = ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size,
|
|
rt->prev_pkt[1]);
|
|
ff_rtmp_packet_destroy(&pkt);
|
|
|
|
return ret;
|
|
}
|
|
|
|
/**
|
|
* Generate ping reply and send it to the server.
|
|
*/
|
|
static int gen_pong(URLContext *s, RTMPContext *rt, RTMPPacket *ppkt)
|
|
{
|
|
RTMPPacket pkt;
|
|
uint8_t *p;
|
|
int ret;
|
|
|
|
if (ppkt->data_size < 6) {
|
|
av_log(s, AV_LOG_ERROR, "Too short ping packet (%d)\n",
|
|
ppkt->data_size);
|
|
return AVERROR_INVALIDDATA;
|
|
}
|
|
|
|
if ((ret = ff_rtmp_packet_create(&pkt, RTMP_NETWORK_CHANNEL, RTMP_PT_PING,
|
|
ppkt->timestamp + 1, 6)) < 0)
|
|
return ret;
|
|
|
|
p = pkt.data;
|
|
bytestream_put_be16(&p, 7);
|
|
bytestream_put_be32(&p, AV_RB32(ppkt->data+2));
|
|
ret = ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size,
|
|
rt->prev_pkt[1]);
|
|
ff_rtmp_packet_destroy(&pkt);
|
|
|
|
return ret;
|
|
}
|
|
|
|
/**
|
|
* Generate server bandwidth message and send it to the server.
|
|
*/
|
|
static int gen_server_bw(URLContext *s, RTMPContext *rt)
|
|
{
|
|
RTMPPacket pkt;
|
|
uint8_t *p;
|
|
int ret;
|
|
|
|
if ((ret = ff_rtmp_packet_create(&pkt, RTMP_NETWORK_CHANNEL, RTMP_PT_SERVER_BW,
|
|
0, 4)) < 0)
|
|
return ret;
|
|
|
|
p = pkt.data;
|
|
bytestream_put_be32(&p, rt->server_bw);
|
|
ret = ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size,
|
|
rt->prev_pkt[1]);
|
|
ff_rtmp_packet_destroy(&pkt);
|
|
|
|
return ret;
|
|
}
|
|
|
|
/**
|
|
* Generate check bandwidth message and send it to the server.
|
|
*/
|
|
static int gen_check_bw(URLContext *s, RTMPContext *rt)
|
|
{
|
|
RTMPPacket pkt;
|
|
uint8_t *p;
|
|
int ret;
|
|
|
|
if ((ret = ff_rtmp_packet_create(&pkt, RTMP_SYSTEM_CHANNEL, RTMP_PT_INVOKE,
|
|
0, 21)) < 0)
|
|
return ret;
|
|
|
|
p = pkt.data;
|
|
ff_amf_write_string(&p, "_checkbw");
|
|
ff_amf_write_number(&p, ++rt->nb_invokes);
|
|
ff_amf_write_null(&p);
|
|
|
|
ret = ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size,
|
|
rt->prev_pkt[1]);
|
|
ff_rtmp_packet_destroy(&pkt);
|
|
|
|
return ret;
|
|
}
|
|
|
|
/**
|
|
* Generate report on bytes read so far and send it to the server.
|
|
*/
|
|
static int gen_bytes_read(URLContext *s, RTMPContext *rt, uint32_t ts)
|
|
{
|
|
RTMPPacket pkt;
|
|
uint8_t *p;
|
|
int ret;
|
|
|
|
if ((ret = ff_rtmp_packet_create(&pkt, RTMP_NETWORK_CHANNEL, RTMP_PT_BYTES_READ,
|
|
ts, 4)) < 0)
|
|
return ret;
|
|
|
|
p = pkt.data;
|
|
bytestream_put_be32(&p, rt->bytes_read);
|
|
ret = ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size,
|
|
rt->prev_pkt[1]);
|
|
ff_rtmp_packet_destroy(&pkt);
|
|
|
|
return ret;
|
|
}
|
|
|
|
int ff_rtmp_calc_digest(const uint8_t *src, int len, int gap,
|
|
const uint8_t *key, int keylen, uint8_t *dst)
|
|
{
|
|
struct AVSHA *sha;
|
|
uint8_t hmac_buf[64+32] = {0};
|
|
int i;
|
|
|
|
sha = av_mallocz(av_sha_size);
|
|
if (!sha)
|
|
return AVERROR(ENOMEM);
|
|
|
|
if (keylen < 64) {
|
|
memcpy(hmac_buf, key, keylen);
|
|
} else {
|
|
av_sha_init(sha, 256);
|
|
av_sha_update(sha,key, keylen);
|
|
av_sha_final(sha, hmac_buf);
|
|
}
|
|
for (i = 0; i < 64; i++)
|
|
hmac_buf[i] ^= HMAC_IPAD_VAL;
|
|
|
|
av_sha_init(sha, 256);
|
|
av_sha_update(sha, hmac_buf, 64);
|
|
if (gap <= 0) {
|
|
av_sha_update(sha, src, len);
|
|
} else { //skip 32 bytes used for storing digest
|
|
av_sha_update(sha, src, gap);
|
|
av_sha_update(sha, src + gap + 32, len - gap - 32);
|
|
}
|
|
av_sha_final(sha, hmac_buf + 64);
|
|
|
|
for (i = 0; i < 64; i++)
|
|
hmac_buf[i] ^= HMAC_IPAD_VAL ^ HMAC_OPAD_VAL; //reuse XORed key for opad
|
|
av_sha_init(sha, 256);
|
|
av_sha_update(sha, hmac_buf, 64+32);
|
|
av_sha_final(sha, dst);
|
|
|
|
av_free(sha);
|
|
|
|
return 0;
|
|
}
|
|
|
|
int ff_rtmp_calc_digest_pos(const uint8_t *buf, int off, int mod_val,
|
|
int add_val)
|
|
{
|
|
int i, digest_pos = 0;
|
|
|
|
for (i = 0; i < 4; i++)
|
|
digest_pos += buf[i + off];
|
|
digest_pos = digest_pos % mod_val + add_val;
|
|
|
|
return digest_pos;
|
|
}
|
|
|
|
/**
|
|
* Put HMAC-SHA2 digest of packet data (except for the bytes where this digest
|
|
* will be stored) into that packet.
|
|
*
|
|
* @param buf handshake data (1536 bytes)
|
|
* @param encrypted use an encrypted connection (RTMPE)
|
|
* @return offset to the digest inside input data
|
|
*/
|
|
static int rtmp_handshake_imprint_with_digest(uint8_t *buf, int encrypted)
|
|
{
|
|
int ret, digest_pos;
|
|
|
|
if (encrypted)
|
|
digest_pos = ff_rtmp_calc_digest_pos(buf, 772, 728, 776);
|
|
else
|
|
digest_pos = ff_rtmp_calc_digest_pos(buf, 8, 728, 12);
|
|
|
|
ret = ff_rtmp_calc_digest(buf, RTMP_HANDSHAKE_PACKET_SIZE, digest_pos,
|
|
rtmp_player_key, PLAYER_KEY_OPEN_PART_LEN,
|
|
buf + digest_pos);
|
|
if (ret < 0)
|
|
return ret;
|
|
|
|
return digest_pos;
|
|
}
|
|
|
|
/**
|
|
* Verify that the received server response has the expected digest value.
|
|
*
|
|
* @param buf handshake data received from the server (1536 bytes)
|
|
* @param off position to search digest offset from
|
|
* @return 0 if digest is valid, digest position otherwise
|
|
*/
|
|
static int rtmp_validate_digest(uint8_t *buf, int off)
|
|
{
|
|
uint8_t digest[32];
|
|
int ret, digest_pos;
|
|
|
|
digest_pos = ff_rtmp_calc_digest_pos(buf, off, 728, off + 4);
|
|
|
|
ret = ff_rtmp_calc_digest(buf, RTMP_HANDSHAKE_PACKET_SIZE, digest_pos,
|
|
rtmp_server_key, SERVER_KEY_OPEN_PART_LEN,
|
|
digest);
|
|
if (ret < 0)
|
|
return ret;
|
|
|
|
if (!memcmp(digest, buf + digest_pos, 32))
|
|
return digest_pos;
|
|
return 0;
|
|
}
|
|
|
|
/**
|
|
* Perform handshake with the server by means of exchanging pseudorandom data
|
|
* signed with HMAC-SHA2 digest.
|
|
*
|
|
* @return 0 if handshake succeeds, negative value otherwise
|
|
*/
|
|
static int rtmp_handshake(URLContext *s, RTMPContext *rt)
|
|
{
|
|
AVLFG rnd;
|
|
uint8_t tosend [RTMP_HANDSHAKE_PACKET_SIZE+1] = {
|
|
3, // unencrypted data
|
|
0, 0, 0, 0, // client uptime
|
|
RTMP_CLIENT_VER1,
|
|
RTMP_CLIENT_VER2,
|
|
RTMP_CLIENT_VER3,
|
|
RTMP_CLIENT_VER4,
|
|
};
|
|
uint8_t clientdata[RTMP_HANDSHAKE_PACKET_SIZE];
|
|
uint8_t serverdata[RTMP_HANDSHAKE_PACKET_SIZE+1];
|
|
int i;
|
|
int server_pos, client_pos;
|
|
uint8_t digest[32], signature[32];
|
|
int ret, type = 0;
|
|
|
|
av_log(s, AV_LOG_DEBUG, "Handshaking...\n");
|
|
|
|
av_lfg_init(&rnd, 0xDEADC0DE);
|
|
// generate handshake packet - 1536 bytes of pseudorandom data
|
|
for (i = 9; i <= RTMP_HANDSHAKE_PACKET_SIZE; i++)
|
|
tosend[i] = av_lfg_get(&rnd) >> 24;
|
|
|
|
if (rt->encrypted && CONFIG_FFRTMPCRYPT_PROTOCOL) {
|
|
/* When the client wants to use RTMPE, we have to change the command
|
|
* byte to 0x06 which means to use encrypted data and we have to set
|
|
* the flash version to at least 9.0.115.0. */
|
|
tosend[0] = 6;
|
|
tosend[5] = 128;
|
|
tosend[6] = 0;
|
|
tosend[7] = 3;
|
|
tosend[8] = 2;
|
|
|
|
/* Initialize the Diffie-Hellmann context and generate the public key
|
|
* to send to the server. */
|
|
if ((ret = ff_rtmpe_gen_pub_key(rt->stream, tosend + 1)) < 0)
|
|
return ret;
|
|
}
|
|
|
|
client_pos = rtmp_handshake_imprint_with_digest(tosend + 1, rt->encrypted);
|
|
if (client_pos < 0)
|
|
return client_pos;
|
|
|
|
if ((ret = ffurl_write(rt->stream, tosend,
|
|
RTMP_HANDSHAKE_PACKET_SIZE + 1)) < 0) {
|
|
av_log(s, AV_LOG_ERROR, "Cannot write RTMP handshake request\n");
|
|
return ret;
|
|
}
|
|
|
|
if ((ret = ffurl_read_complete(rt->stream, serverdata,
|
|
RTMP_HANDSHAKE_PACKET_SIZE + 1)) < 0) {
|
|
av_log(s, AV_LOG_ERROR, "Cannot read RTMP handshake response\n");
|
|
return ret;
|
|
}
|
|
|
|
if ((ret = ffurl_read_complete(rt->stream, clientdata,
|
|
RTMP_HANDSHAKE_PACKET_SIZE)) < 0) {
|
|
av_log(s, AV_LOG_ERROR, "Cannot read RTMP handshake response\n");
|
|
return ret;
|
|
}
|
|
|
|
av_log(s, AV_LOG_DEBUG, "Type answer %d\n", serverdata[0]);
|
|
av_log(s, AV_LOG_DEBUG, "Server version %d.%d.%d.%d\n",
|
|
serverdata[5], serverdata[6], serverdata[7], serverdata[8]);
|
|
|
|
if (rt->is_input && serverdata[5] >= 3) {
|
|
server_pos = rtmp_validate_digest(serverdata + 1, 772);
|
|
if (server_pos < 0)
|
|
return server_pos;
|
|
|
|
if (!server_pos) {
|
|
type = 1;
|
|
server_pos = rtmp_validate_digest(serverdata + 1, 8);
|
|
if (server_pos < 0)
|
|
return server_pos;
|
|
|
|
if (!server_pos) {
|
|
av_log(s, AV_LOG_ERROR, "Server response validating failed\n");
|
|
return AVERROR(EIO);
|
|
}
|
|
}
|
|
|
|
ret = ff_rtmp_calc_digest(tosend + 1 + client_pos, 32, 0,
|
|
rtmp_server_key, sizeof(rtmp_server_key),
|
|
digest);
|
|
if (ret < 0)
|
|
return ret;
|
|
|
|
ret = ff_rtmp_calc_digest(clientdata, RTMP_HANDSHAKE_PACKET_SIZE - 32,
|
|
0, digest, 32, signature);
|
|
if (ret < 0)
|
|
return ret;
|
|
|
|
if (rt->encrypted && CONFIG_FFRTMPCRYPT_PROTOCOL) {
|
|
/* Compute the shared secret key sent by the server and initialize
|
|
* the RC4 encryption. */
|
|
if ((ret = ff_rtmpe_compute_secret_key(rt->stream, serverdata + 1,
|
|
tosend + 1, type)) < 0)
|
|
return ret;
|
|
|
|
/* Encrypt the signature received by the server. */
|
|
ff_rtmpe_encrypt_sig(rt->stream, signature, digest, serverdata[0]);
|
|
}
|
|
|
|
if (memcmp(signature, clientdata + RTMP_HANDSHAKE_PACKET_SIZE - 32, 32)) {
|
|
av_log(s, AV_LOG_ERROR, "Signature mismatch\n");
|
|
return AVERROR(EIO);
|
|
}
|
|
|
|
for (i = 0; i < RTMP_HANDSHAKE_PACKET_SIZE; i++)
|
|
tosend[i] = av_lfg_get(&rnd) >> 24;
|
|
ret = ff_rtmp_calc_digest(serverdata + 1 + server_pos, 32, 0,
|
|
rtmp_player_key, sizeof(rtmp_player_key),
|
|
digest);
|
|
if (ret < 0)
|
|
return ret;
|
|
|
|
ret = ff_rtmp_calc_digest(tosend, RTMP_HANDSHAKE_PACKET_SIZE - 32, 0,
|
|
digest, 32,
|
|
tosend + RTMP_HANDSHAKE_PACKET_SIZE - 32);
|
|
if (ret < 0)
|
|
return ret;
|
|
|
|
if (rt->encrypted && CONFIG_FFRTMPCRYPT_PROTOCOL) {
|
|
/* Encrypt the signature to be send to the server. */
|
|
ff_rtmpe_encrypt_sig(rt->stream, tosend +
|
|
RTMP_HANDSHAKE_PACKET_SIZE - 32, digest,
|
|
serverdata[0]);
|
|
}
|
|
|
|
// write reply back to the server
|
|
if ((ret = ffurl_write(rt->stream, tosend,
|
|
RTMP_HANDSHAKE_PACKET_SIZE)) < 0)
|
|
return ret;
|
|
|
|
if (rt->encrypted && CONFIG_FFRTMPCRYPT_PROTOCOL) {
|
|
/* Set RC4 keys for encryption and update the keystreams. */
|
|
if ((ret = ff_rtmpe_update_keystream(rt->stream)) < 0)
|
|
return ret;
|
|
}
|
|
} else {
|
|
if (rt->encrypted && CONFIG_FFRTMPCRYPT_PROTOCOL) {
|
|
/* Compute the shared secret key sent by the server and initialize
|
|
* the RC4 encryption. */
|
|
if ((ret = ff_rtmpe_compute_secret_key(rt->stream, serverdata + 1,
|
|
tosend + 1, 1)) < 0)
|
|
return ret;
|
|
|
|
if (serverdata[0] == 9) {
|
|
/* Encrypt the signature received by the server. */
|
|
ff_rtmpe_encrypt_sig(rt->stream, signature, digest,
|
|
serverdata[0]);
|
|
}
|
|
}
|
|
|
|
if ((ret = ffurl_write(rt->stream, serverdata + 1,
|
|
RTMP_HANDSHAKE_PACKET_SIZE)) < 0)
|
|
return ret;
|
|
|
|
if (rt->encrypted && CONFIG_FFRTMPCRYPT_PROTOCOL) {
|
|
/* Set RC4 keys for encryption and update the keystreams. */
|
|
if ((ret = ff_rtmpe_update_keystream(rt->stream)) < 0)
|
|
return ret;
|
|
}
|
|
}
|
|
|
|
return 0;
|
|
}
|
|
|
|
static int handle_chunk_size(URLContext *s, RTMPPacket *pkt)
|
|
{
|
|
RTMPContext *rt = s->priv_data;
|
|
int ret;
|
|
|
|
if (pkt->data_size < 4) {
|
|
av_log(s, AV_LOG_ERROR,
|
|
"Too short chunk size change packet (%d)\n",
|
|
pkt->data_size);
|
|
return AVERROR_INVALIDDATA;
|
|
}
|
|
|
|
if (!rt->is_input) {
|
|
if ((ret = ff_rtmp_packet_write(rt->stream, pkt, rt->chunk_size,
|
|
rt->prev_pkt[1])) < 0)
|
|
return ret;
|
|
}
|
|
|
|
rt->chunk_size = AV_RB32(pkt->data);
|
|
if (rt->chunk_size <= 0) {
|
|
av_log(s, AV_LOG_ERROR, "Incorrect chunk size %d\n", rt->chunk_size);
|
|
return AVERROR_INVALIDDATA;
|
|
}
|
|
av_log(s, AV_LOG_DEBUG, "New chunk size = %d\n", rt->chunk_size);
|
|
|
|
return 0;
|
|
}
|
|
|
|
static int handle_ping(URLContext *s, RTMPPacket *pkt)
|
|
{
|
|
RTMPContext *rt = s->priv_data;
|
|
int t, ret;
|
|
|
|
if (pkt->data_size < 2) {
|
|
av_log(s, AV_LOG_ERROR, "Too short ping packet (%d)\n",
|
|
pkt->data_size);
|
|
return AVERROR_INVALIDDATA;
|
|
}
|
|
|
|
t = AV_RB16(pkt->data);
|
|
if (t == 6) {
|
|
if ((ret = gen_pong(s, rt, pkt)) < 0)
|
|
return ret;
|
|
}
|
|
|
|
return 0;
|
|
}
|
|
|
|
static int handle_client_bw(URLContext *s, RTMPPacket *pkt)
|
|
{
|
|
RTMPContext *rt = s->priv_data;
|
|
|
|
if (pkt->data_size < 4) {
|
|
av_log(s, AV_LOG_ERROR,
|
|
"Client bandwidth report packet is less than 4 bytes long (%d)\n",
|
|
pkt->data_size);
|
|
return AVERROR_INVALIDDATA;
|
|
}
|
|
|
|
rt->client_report_size = AV_RB32(pkt->data);
|
|
if (rt->client_report_size <= 0) {
|
|
av_log(s, AV_LOG_ERROR, "Incorrect client bandwidth %d\n",
|
|
rt->client_report_size);
|
|
return AVERROR_INVALIDDATA;
|
|
|
|
}
|
|
av_log(s, AV_LOG_DEBUG, "Client bandwidth = %d\n", rt->client_report_size);
|
|
rt->client_report_size >>= 1;
|
|
|
|
return 0;
|
|
}
|
|
|
|
static int handle_server_bw(URLContext *s, RTMPPacket *pkt)
|
|
{
|
|
RTMPContext *rt = s->priv_data;
|
|
|
|
if (pkt->data_size < 4) {
|
|
av_log(s, AV_LOG_ERROR,
|
|
"Too short server bandwidth report packet (%d)\n",
|
|
pkt->data_size);
|
|
return AVERROR_INVALIDDATA;
|
|
}
|
|
|
|
rt->server_bw = AV_RB32(pkt->data);
|
|
if (rt->server_bw <= 0) {
|
|
av_log(s, AV_LOG_ERROR, "Incorrect server bandwidth %d\n",
|
|
rt->server_bw);
|
|
return AVERROR_INVALIDDATA;
|
|
}
|
|
av_log(s, AV_LOG_DEBUG, "Server bandwidth = %d\n", rt->server_bw);
|
|
|
|
return 0;
|
|
}
|
|
|
|
static int handle_invoke(URLContext *s, RTMPPacket *pkt)
|
|
{
|
|
RTMPContext *rt = s->priv_data;
|
|
int i, t;
|
|
const uint8_t *data_end = pkt->data + pkt->data_size;
|
|
int ret;
|
|
|
|
//TODO: check for the messages sent for wrong state?
|
|
if (!memcmp(pkt->data, "\002\000\006_error", 9)) {
|
|
uint8_t tmpstr[256];
|
|
|
|
if (!ff_amf_get_field_value(pkt->data + 9, data_end,
|
|
"description", tmpstr, sizeof(tmpstr)))
|
|
av_log(s, AV_LOG_ERROR, "Server error: %s\n",tmpstr);
|
|
return -1;
|
|
} else if (!memcmp(pkt->data, "\002\000\007_result", 10)) {
|
|
switch (rt->state) {
|
|
case STATE_HANDSHAKED:
|
|
if (!rt->is_input) {
|
|
if ((ret = gen_release_stream(s, rt)) < 0)
|
|
return ret;
|
|
if ((ret = gen_fcpublish_stream(s, rt)) < 0)
|
|
return ret;
|
|
rt->state = STATE_RELEASING;
|
|
} else {
|
|
if ((ret = gen_server_bw(s, rt)) < 0)
|
|
return ret;
|
|
rt->state = STATE_CONNECTING;
|
|
}
|
|
if ((ret = gen_create_stream(s, rt)) < 0)
|
|
return ret;
|
|
break;
|
|
case STATE_FCPUBLISH:
|
|
rt->state = STATE_CONNECTING;
|
|
break;
|
|
case STATE_RELEASING:
|
|
rt->state = STATE_FCPUBLISH;
|
|
/* hack for Wowza Media Server, it does not send result for
|
|
* releaseStream and FCPublish calls */
|
|
if (!pkt->data[10]) {
|
|
int pkt_id = av_int2double(AV_RB64(pkt->data + 11));
|
|
if (pkt_id == rt->create_stream_invoke)
|
|
rt->state = STATE_CONNECTING;
|
|
}
|
|
if (rt->state != STATE_CONNECTING)
|
|
break;
|
|
case STATE_CONNECTING:
|
|
//extract a number from the result
|
|
if (pkt->data[10] || pkt->data[19] != 5 || pkt->data[20]) {
|
|
av_log(s, AV_LOG_WARNING, "Unexpected reply on connect()\n");
|
|
} else {
|
|
rt->main_channel_id = av_int2double(AV_RB64(pkt->data + 21));
|
|
}
|
|
if (rt->is_input) {
|
|
if ((ret = gen_play(s, rt)) < 0)
|
|
return ret;
|
|
if ((ret = gen_buffer_time(s, rt)) < 0)
|
|
return ret;
|
|
} else {
|
|
if ((ret = gen_publish(s, rt)) < 0)
|
|
return ret;
|
|
}
|
|
rt->state = STATE_READY;
|
|
break;
|
|
}
|
|
} else if (!memcmp(pkt->data, "\002\000\010onStatus", 11)) {
|
|
const uint8_t* ptr = pkt->data + 11;
|
|
uint8_t tmpstr[256];
|
|
|
|
for (i = 0; i < 2; i++) {
|
|
t = ff_amf_tag_size(ptr, data_end);
|
|
if (t < 0)
|
|
return 1;
|
|
ptr += t;
|
|
}
|
|
t = ff_amf_get_field_value(ptr, data_end,
|
|
"level", tmpstr, sizeof(tmpstr));
|
|
if (!t && !strcmp(tmpstr, "error")) {
|
|
if (!ff_amf_get_field_value(ptr, data_end,
|
|
"description", tmpstr, sizeof(tmpstr)))
|
|
av_log(s, AV_LOG_ERROR, "Server error: %s\n",tmpstr);
|
|
return -1;
|
|
}
|
|
t = ff_amf_get_field_value(ptr, data_end,
|
|
"code", tmpstr, sizeof(tmpstr));
|
|
if (!t && !strcmp(tmpstr, "NetStream.Play.Start")) rt->state = STATE_PLAYING;
|
|
if (!t && !strcmp(tmpstr, "NetStream.Play.Stop")) rt->state = STATE_STOPPED;
|
|
if (!t && !strcmp(tmpstr, "NetStream.Play.UnpublishNotify")) rt->state = STATE_STOPPED;
|
|
if (!t && !strcmp(tmpstr, "NetStream.Publish.Start")) rt->state = STATE_PUBLISHING;
|
|
} else if (!memcmp(pkt->data, "\002\000\010onBWDone", 11)) {
|
|
if ((ret = gen_check_bw(s, rt)) < 0)
|
|
return ret;
|
|
}
|
|
|
|
return 0;
|
|
}
|
|
|
|
/**
|
|
* Parse received packet and possibly perform some action depending on
|
|
* the packet contents.
|
|
* @return 0 for no errors, negative values for serious errors which prevent
|
|
* further communications, positive values for uncritical errors
|
|
*/
|
|
static int rtmp_parse_result(URLContext *s, RTMPContext *rt, RTMPPacket *pkt)
|
|
{
|
|
int ret;
|
|
|
|
#ifdef DEBUG
|
|
ff_rtmp_packet_dump(s, pkt);
|
|
#endif
|
|
|
|
switch (pkt->type) {
|
|
case RTMP_PT_CHUNK_SIZE:
|
|
if ((ret = handle_chunk_size(s, pkt)) < 0)
|
|
return ret;
|
|
break;
|
|
case RTMP_PT_PING:
|
|
if ((ret = handle_ping(s, pkt)) < 0)
|
|
return ret;
|
|
break;
|
|
case RTMP_PT_CLIENT_BW:
|
|
if ((ret = handle_client_bw(s, pkt)) < 0)
|
|
return ret;
|
|
break;
|
|
case RTMP_PT_SERVER_BW:
|
|
if ((ret = handle_server_bw(s, pkt)) < 0)
|
|
return ret;
|
|
break;
|
|
case RTMP_PT_INVOKE:
|
|
if ((ret = handle_invoke(s, pkt)) < 0)
|
|
return ret;
|
|
break;
|
|
case RTMP_PT_VIDEO:
|
|
case RTMP_PT_AUDIO:
|
|
/* 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;
|
|
}
|
|
|
|
/**
|
|
* Interact with the server by receiving and sending RTMP packets until
|
|
* there is some significant data (media data or expected status notification).
|
|
*
|
|
* @param s reading context
|
|
* @param for_header non-zero value tells function to work until it
|
|
* gets notification from the server that playing has been started,
|
|
* otherwise function will work until some media data is received (or
|
|
* an error happens)
|
|
* @return 0 for successful operation, negative value in case of error
|
|
*/
|
|
static int get_packet(URLContext *s, int for_header)
|
|
{
|
|
RTMPContext *rt = s->priv_data;
|
|
int ret;
|
|
uint8_t *p;
|
|
const uint8_t *next;
|
|
uint32_t data_size;
|
|
uint32_t ts, cts, pts=0;
|
|
|
|
if (rt->state == STATE_STOPPED)
|
|
return AVERROR_EOF;
|
|
|
|
for (;;) {
|
|
RTMPPacket rpkt = { 0 };
|
|
if ((ret = ff_rtmp_packet_read(rt->stream, &rpkt,
|
|
rt->chunk_size, rt->prev_pkt[0])) <= 0) {
|
|
if (ret == 0) {
|
|
return AVERROR(EAGAIN);
|
|
} else {
|
|
return AVERROR(EIO);
|
|
}
|
|
}
|
|
rt->bytes_read += ret;
|
|
if (rt->bytes_read - rt->last_bytes_read > rt->client_report_size) {
|
|
av_log(s, AV_LOG_DEBUG, "Sending bytes read report\n");
|
|
if ((ret = gen_bytes_read(s, rt, rpkt.timestamp + 1)) < 0)
|
|
return ret;
|
|
rt->last_bytes_read = rt->bytes_read;
|
|
}
|
|
|
|
ret = rtmp_parse_result(s, rt, &rpkt);
|
|
if (ret < 0) {//serious error in current packet
|
|
ff_rtmp_packet_destroy(&rpkt);
|
|
return ret;
|
|
}
|
|
if (rt->state == STATE_STOPPED) {
|
|
ff_rtmp_packet_destroy(&rpkt);
|
|
return AVERROR_EOF;
|
|
}
|
|
if (for_header && (rt->state == STATE_PLAYING || rt->state == STATE_PUBLISHING)) {
|
|
ff_rtmp_packet_destroy(&rpkt);
|
|
return 0;
|
|
}
|
|
if (!rpkt.data_size || !rt->is_input) {
|
|
ff_rtmp_packet_destroy(&rpkt);
|
|
continue;
|
|
}
|
|
if (rpkt.type == RTMP_PT_VIDEO || rpkt.type == RTMP_PT_AUDIO ||
|
|
(rpkt.type == RTMP_PT_NOTIFY && !memcmp("\002\000\012onMetaData", rpkt.data, 13))) {
|
|
ts = rpkt.timestamp;
|
|
|
|
// generate packet header and put data into buffer for FLV demuxer
|
|
rt->flv_off = 0;
|
|
rt->flv_size = rpkt.data_size + 15;
|
|
rt->flv_data = p = av_realloc(rt->flv_data, rt->flv_size);
|
|
bytestream_put_byte(&p, rpkt.type);
|
|
bytestream_put_be24(&p, rpkt.data_size);
|
|
bytestream_put_be24(&p, ts);
|
|
bytestream_put_byte(&p, ts >> 24);
|
|
bytestream_put_be24(&p, 0);
|
|
bytestream_put_buffer(&p, rpkt.data, rpkt.data_size);
|
|
bytestream_put_be32(&p, 0);
|
|
ff_rtmp_packet_destroy(&rpkt);
|
|
return 0;
|
|
} else if (rpkt.type == RTMP_PT_METADATA) {
|
|
// we got raw FLV data, make it available for FLV demuxer
|
|
rt->flv_off = 0;
|
|
rt->flv_size = rpkt.data_size;
|
|
rt->flv_data = av_realloc(rt->flv_data, rt->flv_size);
|
|
/* rewrite timestamps */
|
|
next = rpkt.data;
|
|
ts = rpkt.timestamp;
|
|
while (next - rpkt.data < rpkt.data_size - 11) {
|
|
next++;
|
|
data_size = bytestream_get_be24(&next);
|
|
p=next;
|
|
cts = bytestream_get_be24(&next);
|
|
cts |= bytestream_get_byte(&next) << 24;
|
|
if (pts==0)
|
|
pts=cts;
|
|
ts += cts - pts;
|
|
pts = cts;
|
|
bytestream_put_be24(&p, ts);
|
|
bytestream_put_byte(&p, ts >> 24);
|
|
next += data_size + 3 + 4;
|
|
}
|
|
memcpy(rt->flv_data, rpkt.data, rpkt.data_size);
|
|
ff_rtmp_packet_destroy(&rpkt);
|
|
return 0;
|
|
}
|
|
ff_rtmp_packet_destroy(&rpkt);
|
|
}
|
|
}
|
|
|
|
static int rtmp_close(URLContext *h)
|
|
{
|
|
RTMPContext *rt = h->priv_data;
|
|
int ret = 0;
|
|
|
|
if (!rt->is_input) {
|
|
rt->flv_data = NULL;
|
|
if (rt->out_pkt.data_size)
|
|
ff_rtmp_packet_destroy(&rt->out_pkt);
|
|
if (rt->state > STATE_FCPUBLISH)
|
|
ret = gen_fcunpublish_stream(h, rt);
|
|
}
|
|
if (rt->state > STATE_HANDSHAKED)
|
|
ret = gen_delete_stream(h, rt);
|
|
|
|
av_freep(&rt->flv_data);
|
|
ffurl_close(rt->stream);
|
|
return ret;
|
|
}
|
|
|
|
/**
|
|
* Open RTMP connection and verify that the stream can be played.
|
|
*
|
|
* URL syntax: rtmp://server[:port][/app][/playpath]
|
|
* where 'app' is first one or two directories in the path
|
|
* (e.g. /ondemand/, /flash/live/, etc.)
|
|
* and 'playpath' is a file name (the rest of the path,
|
|
* may be prefixed with "mp4:")
|
|
*/
|
|
static int rtmp_open(URLContext *s, const char *uri, int flags)
|
|
{
|
|
RTMPContext *rt = s->priv_data;
|
|
char proto[8], hostname[256], path[1024], *fname;
|
|
char *old_app;
|
|
uint8_t buf[2048];
|
|
int port;
|
|
AVDictionary *opts = NULL;
|
|
int ret;
|
|
|
|
rt->is_input = !(flags & AVIO_FLAG_WRITE);
|
|
|
|
av_url_split(proto, sizeof(proto), NULL, 0, hostname, sizeof(hostname), &port,
|
|
path, sizeof(path), s->filename);
|
|
|
|
if (!strcmp(proto, "rtmpt") || !strcmp(proto, "rtmpts")) {
|
|
if (!strcmp(proto, "rtmpts"))
|
|
av_dict_set(&opts, "ffrtmphttp_tls", "1", 1);
|
|
|
|
/* open the http tunneling connection */
|
|
ff_url_join(buf, sizeof(buf), "ffrtmphttp", NULL, hostname, port, NULL);
|
|
} else if (!strcmp(proto, "rtmps")) {
|
|
/* open the tls connection */
|
|
if (port < 0)
|
|
port = RTMPS_DEFAULT_PORT;
|
|
ff_url_join(buf, sizeof(buf), "tls", NULL, hostname, port, NULL);
|
|
} else if (!strcmp(proto, "rtmpe") || (!strcmp(proto, "rtmpte"))) {
|
|
if (!strcmp(proto, "rtmpte"))
|
|
av_dict_set(&opts, "ffrtmpcrypt_tunneling", "1", 1);
|
|
|
|
/* open the encrypted connection */
|
|
ff_url_join(buf, sizeof(buf), "ffrtmpcrypt", NULL, hostname, port, NULL);
|
|
rt->encrypted = 1;
|
|
} else {
|
|
/* open the tcp connection */
|
|
if (port < 0)
|
|
port = RTMP_DEFAULT_PORT;
|
|
ff_url_join(buf, sizeof(buf), "tcp", NULL, hostname, port, NULL);
|
|
}
|
|
|
|
if ((ret = ffurl_open(&rt->stream, buf, AVIO_FLAG_READ_WRITE,
|
|
&s->interrupt_callback, &opts)) < 0) {
|
|
av_log(s , AV_LOG_ERROR, "Cannot open connection %s\n", buf);
|
|
goto fail;
|
|
}
|
|
|
|
rt->state = STATE_START;
|
|
if ((ret = rtmp_handshake(s, rt)) < 0)
|
|
goto fail;
|
|
|
|
rt->chunk_size = 128;
|
|
rt->state = STATE_HANDSHAKED;
|
|
|
|
// Keep the application name when it has been defined by the user.
|
|
old_app = rt->app;
|
|
|
|
rt->app = av_malloc(APP_MAX_LENGTH);
|
|
if (!rt->app) {
|
|
ret = AVERROR(ENOMEM);
|
|
goto fail;
|
|
}
|
|
|
|
//extract "app" part from path
|
|
if (!strncmp(path, "/ondemand/", 10)) {
|
|
fname = path + 10;
|
|
memcpy(rt->app, "ondemand", 9);
|
|
} else {
|
|
char *next = *path ? path + 1 : path;
|
|
char *p = strchr(next, '/');
|
|
if (!p) {
|
|
fname = next;
|
|
rt->app[0] = '\0';
|
|
} else {
|
|
// make sure we do not mismatch a playpath for an application instance
|
|
char *c = strchr(p + 1, ':');
|
|
fname = strchr(p + 1, '/');
|
|
if (!fname || (c && c < fname)) {
|
|
fname = p + 1;
|
|
av_strlcpy(rt->app, path + 1, p - path);
|
|
} else {
|
|
fname++;
|
|
av_strlcpy(rt->app, path + 1, fname - path - 1);
|
|
}
|
|
}
|
|
}
|
|
|
|
if (old_app) {
|
|
// The name of application has been defined by the user, override it.
|
|
av_free(rt->app);
|
|
rt->app = old_app;
|
|
}
|
|
|
|
if (!rt->playpath) {
|
|
int len = strlen(fname);
|
|
|
|
rt->playpath = av_malloc(PLAYPATH_MAX_LENGTH);
|
|
if (!rt->playpath) {
|
|
ret = AVERROR(ENOMEM);
|
|
goto fail;
|
|
}
|
|
|
|
if (!strchr(fname, ':') && len >= 4 &&
|
|
(!strcmp(fname + len - 4, ".f4v") ||
|
|
!strcmp(fname + len - 4, ".mp4"))) {
|
|
memcpy(rt->playpath, "mp4:", 5);
|
|
} else if (len >= 4 && !strcmp(fname + len - 4, ".flv")) {
|
|
fname[len - 4] = '\0';
|
|
} else {
|
|
rt->playpath[0] = 0;
|
|
}
|
|
strncat(rt->playpath, fname, PLAYPATH_MAX_LENGTH - 5);
|
|
}
|
|
|
|
if (!rt->tcurl) {
|
|
rt->tcurl = av_malloc(TCURL_MAX_LENGTH);
|
|
if (!rt->tcurl) {
|
|
ret = AVERROR(ENOMEM);
|
|
goto fail;
|
|
}
|
|
ff_url_join(rt->tcurl, TCURL_MAX_LENGTH, proto, NULL, hostname,
|
|
port, "/%s", rt->app);
|
|
}
|
|
|
|
if (!rt->flashver) {
|
|
rt->flashver = av_malloc(FLASHVER_MAX_LENGTH);
|
|
if (!rt->flashver) {
|
|
ret = AVERROR(ENOMEM);
|
|
goto fail;
|
|
}
|
|
if (rt->is_input) {
|
|
snprintf(rt->flashver, FLASHVER_MAX_LENGTH, "%s %d,%d,%d,%d",
|
|
RTMP_CLIENT_PLATFORM, RTMP_CLIENT_VER1, RTMP_CLIENT_VER2,
|
|
RTMP_CLIENT_VER3, RTMP_CLIENT_VER4);
|
|
} else {
|
|
snprintf(rt->flashver, FLASHVER_MAX_LENGTH,
|
|
"FMLE/3.0 (compatible; %s)", LIBAVFORMAT_IDENT);
|
|
}
|
|
}
|
|
|
|
rt->client_report_size = 1048576;
|
|
rt->bytes_read = 0;
|
|
rt->last_bytes_read = 0;
|
|
rt->server_bw = 2500000;
|
|
|
|
av_log(s, AV_LOG_DEBUG, "Proto = %s, path = %s, app = %s, fname = %s\n",
|
|
proto, path, rt->app, rt->playpath);
|
|
if ((ret = gen_connect(s, rt)) < 0)
|
|
goto fail;
|
|
|
|
do {
|
|
ret = get_packet(s, 1);
|
|
} while (ret == EAGAIN);
|
|
if (ret < 0)
|
|
goto fail;
|
|
|
|
if (rt->is_input) {
|
|
// generate FLV header for demuxer
|
|
rt->flv_size = 13;
|
|
rt->flv_data = av_realloc(rt->flv_data, rt->flv_size);
|
|
rt->flv_off = 0;
|
|
memcpy(rt->flv_data, "FLV\1\5\0\0\0\011\0\0\0\0", rt->flv_size);
|
|
} else {
|
|
rt->flv_size = 0;
|
|
rt->flv_data = NULL;
|
|
rt->flv_off = 0;
|
|
rt->skip_bytes = 13;
|
|
}
|
|
|
|
s->max_packet_size = rt->stream->max_packet_size;
|
|
s->is_streamed = 1;
|
|
return 0;
|
|
|
|
fail:
|
|
av_dict_free(&opts);
|
|
rtmp_close(s);
|
|
return ret;
|
|
}
|
|
|
|
static int rtmp_read(URLContext *s, uint8_t *buf, int size)
|
|
{
|
|
RTMPContext *rt = s->priv_data;
|
|
int orig_size = size;
|
|
int ret;
|
|
|
|
while (size > 0) {
|
|
int data_left = rt->flv_size - rt->flv_off;
|
|
|
|
if (data_left >= size) {
|
|
memcpy(buf, rt->flv_data + rt->flv_off, size);
|
|
rt->flv_off += size;
|
|
return orig_size;
|
|
}
|
|
if (data_left > 0) {
|
|
memcpy(buf, rt->flv_data + rt->flv_off, data_left);
|
|
buf += data_left;
|
|
size -= data_left;
|
|
rt->flv_off = rt->flv_size;
|
|
return data_left;
|
|
}
|
|
if ((ret = get_packet(s, 0)) < 0)
|
|
return ret;
|
|
}
|
|
return orig_size;
|
|
}
|
|
|
|
static int rtmp_write(URLContext *s, const uint8_t *buf, int size)
|
|
{
|
|
RTMPContext *rt = s->priv_data;
|
|
int size_temp = size;
|
|
int pktsize, pkttype;
|
|
uint32_t ts;
|
|
const uint8_t *buf_temp = buf;
|
|
uint8_t c;
|
|
int ret;
|
|
|
|
do {
|
|
if (rt->skip_bytes) {
|
|
int skip = FFMIN(rt->skip_bytes, size_temp);
|
|
buf_temp += skip;
|
|
size_temp -= skip;
|
|
rt->skip_bytes -= skip;
|
|
continue;
|
|
}
|
|
|
|
if (rt->flv_header_bytes < 11) {
|
|
const uint8_t *header = rt->flv_header;
|
|
int copy = FFMIN(11 - rt->flv_header_bytes, size_temp);
|
|
bytestream_get_buffer(&buf_temp, rt->flv_header + rt->flv_header_bytes, copy);
|
|
rt->flv_header_bytes += copy;
|
|
size_temp -= copy;
|
|
if (rt->flv_header_bytes < 11)
|
|
break;
|
|
|
|
pkttype = bytestream_get_byte(&header);
|
|
pktsize = bytestream_get_be24(&header);
|
|
ts = bytestream_get_be24(&header);
|
|
ts |= bytestream_get_byte(&header) << 24;
|
|
bytestream_get_be24(&header);
|
|
rt->flv_size = pktsize;
|
|
|
|
//force 12bytes header
|
|
if (((pkttype == RTMP_PT_VIDEO || pkttype == RTMP_PT_AUDIO) && ts == 0) ||
|
|
pkttype == RTMP_PT_NOTIFY) {
|
|
if (pkttype == RTMP_PT_NOTIFY)
|
|
pktsize += 16;
|
|
rt->prev_pkt[1][RTMP_SOURCE_CHANNEL].channel_id = 0;
|
|
}
|
|
|
|
//this can be a big packet, it's better to send it right here
|
|
if ((ret = ff_rtmp_packet_create(&rt->out_pkt, RTMP_SOURCE_CHANNEL,
|
|
pkttype, ts, pktsize)) < 0)
|
|
return ret;
|
|
|
|
rt->out_pkt.extra = rt->main_channel_id;
|
|
rt->flv_data = rt->out_pkt.data;
|
|
|
|
if (pkttype == RTMP_PT_NOTIFY)
|
|
ff_amf_write_string(&rt->flv_data, "@setDataFrame");
|
|
}
|
|
|
|
if (rt->flv_size - rt->flv_off > size_temp) {
|
|
bytestream_get_buffer(&buf_temp, rt->flv_data + rt->flv_off, size_temp);
|
|
rt->flv_off += size_temp;
|
|
size_temp = 0;
|
|
} else {
|
|
bytestream_get_buffer(&buf_temp, rt->flv_data + rt->flv_off, rt->flv_size - rt->flv_off);
|
|
size_temp -= rt->flv_size - rt->flv_off;
|
|
rt->flv_off += rt->flv_size - rt->flv_off;
|
|
}
|
|
|
|
if (rt->flv_off == rt->flv_size) {
|
|
rt->skip_bytes = 4;
|
|
|
|
if ((ret = ff_rtmp_packet_write(rt->stream, &rt->out_pkt,
|
|
rt->chunk_size, rt->prev_pkt[1])) < 0)
|
|
return ret;
|
|
ff_rtmp_packet_destroy(&rt->out_pkt);
|
|
rt->flv_size = 0;
|
|
rt->flv_off = 0;
|
|
rt->flv_header_bytes = 0;
|
|
rt->flv_nb_packets++;
|
|
}
|
|
} while (buf_temp - buf < size);
|
|
|
|
if (rt->flv_nb_packets < rt->flush_interval)
|
|
return size;
|
|
rt->flv_nb_packets = 0;
|
|
|
|
/* set stream into nonblocking mode */
|
|
rt->stream->flags |= AVIO_FLAG_NONBLOCK;
|
|
|
|
/* try to read one byte from the stream */
|
|
ret = ffurl_read(rt->stream, &c, 1);
|
|
|
|
/* switch the stream back into blocking mode */
|
|
rt->stream->flags &= ~AVIO_FLAG_NONBLOCK;
|
|
|
|
if (ret == AVERROR(EAGAIN)) {
|
|
/* no incoming data to handle */
|
|
return size;
|
|
} else if (ret < 0) {
|
|
return ret;
|
|
} else if (ret == 1) {
|
|
RTMPPacket rpkt = { 0 };
|
|
|
|
if ((ret = ff_rtmp_packet_read_internal(rt->stream, &rpkt,
|
|
rt->chunk_size,
|
|
rt->prev_pkt[0], c)) <= 0)
|
|
return ret;
|
|
|
|
if ((ret = rtmp_parse_result(s, rt, &rpkt)) < 0)
|
|
return ret;
|
|
|
|
ff_rtmp_packet_destroy(&rpkt);
|
|
}
|
|
|
|
return size;
|
|
}
|
|
|
|
#define OFFSET(x) offsetof(RTMPContext, x)
|
|
#define DEC AV_OPT_FLAG_DECODING_PARAM
|
|
#define ENC AV_OPT_FLAG_ENCODING_PARAM
|
|
|
|
static const AVOption rtmp_options[] = {
|
|
{"rtmp_app", "Name of application to connect to on the RTMP server", OFFSET(app), AV_OPT_TYPE_STRING, {.str = NULL }, 0, 0, DEC|ENC},
|
|
{"rtmp_buffer", "Set buffer time in milliseconds. The default is 3000.", OFFSET(client_buffer_time), AV_OPT_TYPE_INT, {3000}, 0, INT_MAX, DEC|ENC},
|
|
{"rtmp_conn", "Append arbitrary AMF data to the Connect message", OFFSET(conn), AV_OPT_TYPE_STRING, {.str = NULL }, 0, 0, DEC|ENC},
|
|
{"rtmp_flashver", "Version of the Flash plugin used to run the SWF player.", OFFSET(flashver), AV_OPT_TYPE_STRING, {.str = NULL }, 0, 0, DEC|ENC},
|
|
{"rtmp_flush_interval", "Number of packets flushed in the same request (RTMPT only).", OFFSET(flush_interval), AV_OPT_TYPE_INT, {10}, 0, INT_MAX, ENC},
|
|
{"rtmp_live", "Specify that the media is a live stream.", OFFSET(live), AV_OPT_TYPE_INT, {-2}, INT_MIN, INT_MAX, DEC, "rtmp_live"},
|
|
{"any", "both", 0, AV_OPT_TYPE_CONST, {-2}, 0, 0, DEC, "rtmp_live"},
|
|
{"live", "live stream", 0, AV_OPT_TYPE_CONST, {-1}, 0, 0, DEC, "rtmp_live"},
|
|
{"recorded", "recorded stream", 0, AV_OPT_TYPE_CONST, {0}, 0, 0, DEC, "rtmp_live"},
|
|
{"rtmp_pageurl", "URL of the web page in which the media was embedded. By default no value will be sent.", OFFSET(pageurl), AV_OPT_TYPE_STRING, {.str = NULL }, 0, 0, DEC},
|
|
{"rtmp_playpath", "Stream identifier to play or to publish", OFFSET(playpath), AV_OPT_TYPE_STRING, {.str = NULL }, 0, 0, DEC|ENC},
|
|
{"rtmp_swfurl", "URL of the SWF player. By default no value will be sent", OFFSET(swfurl), AV_OPT_TYPE_STRING, {.str = NULL }, 0, 0, DEC|ENC},
|
|
{"rtmp_tcurl", "URL of the target stream. Defaults to proto://host[:port]/app.", OFFSET(tcurl), AV_OPT_TYPE_STRING, {.str = NULL }, 0, 0, DEC|ENC},
|
|
{ NULL },
|
|
};
|
|
|
|
static const AVClass rtmp_class = {
|
|
.class_name = "rtmp",
|
|
.item_name = av_default_item_name,
|
|
.option = rtmp_options,
|
|
.version = LIBAVUTIL_VERSION_INT,
|
|
};
|
|
|
|
URLProtocol ff_rtmp_protocol = {
|
|
.name = "rtmp",
|
|
.url_open = rtmp_open,
|
|
.url_read = rtmp_read,
|
|
.url_write = rtmp_write,
|
|
.url_close = rtmp_close,
|
|
.priv_data_size = sizeof(RTMPContext),
|
|
.flags = URL_PROTOCOL_FLAG_NETWORK,
|
|
.priv_data_class= &rtmp_class,
|
|
};
|
|
|
|
static const AVClass rtmpe_class = {
|
|
.class_name = "rtmpe",
|
|
.item_name = av_default_item_name,
|
|
.option = rtmp_options,
|
|
.version = LIBAVUTIL_VERSION_INT,
|
|
};
|
|
|
|
URLProtocol ff_rtmpe_protocol = {
|
|
.name = "rtmpe",
|
|
.url_open = rtmp_open,
|
|
.url_read = rtmp_read,
|
|
.url_write = rtmp_write,
|
|
.url_close = rtmp_close,
|
|
.priv_data_size = sizeof(RTMPContext),
|
|
.flags = URL_PROTOCOL_FLAG_NETWORK,
|
|
.priv_data_class = &rtmpe_class,
|
|
};
|
|
|
|
static const AVClass rtmps_class = {
|
|
.class_name = "rtmps",
|
|
.item_name = av_default_item_name,
|
|
.option = rtmp_options,
|
|
.version = LIBAVUTIL_VERSION_INT,
|
|
};
|
|
|
|
URLProtocol ff_rtmps_protocol = {
|
|
.name = "rtmps",
|
|
.url_open = rtmp_open,
|
|
.url_read = rtmp_read,
|
|
.url_write = rtmp_write,
|
|
.url_close = rtmp_close,
|
|
.priv_data_size = sizeof(RTMPContext),
|
|
.flags = URL_PROTOCOL_FLAG_NETWORK,
|
|
.priv_data_class = &rtmps_class,
|
|
};
|
|
|
|
static const AVClass rtmpt_class = {
|
|
.class_name = "rtmpt",
|
|
.item_name = av_default_item_name,
|
|
.option = rtmp_options,
|
|
.version = LIBAVUTIL_VERSION_INT,
|
|
};
|
|
|
|
URLProtocol ff_rtmpt_protocol = {
|
|
.name = "rtmpt",
|
|
.url_open = rtmp_open,
|
|
.url_read = rtmp_read,
|
|
.url_write = rtmp_write,
|
|
.url_close = rtmp_close,
|
|
.priv_data_size = sizeof(RTMPContext),
|
|
.flags = URL_PROTOCOL_FLAG_NETWORK,
|
|
.priv_data_class = &rtmpt_class,
|
|
};
|
|
|
|
static const AVClass rtmpte_class = {
|
|
.class_name = "rtmpte",
|
|
.item_name = av_default_item_name,
|
|
.option = rtmp_options,
|
|
.version = LIBAVUTIL_VERSION_INT,
|
|
};
|
|
|
|
URLProtocol ff_rtmpte_protocol = {
|
|
.name = "rtmpte",
|
|
.url_open = rtmp_open,
|
|
.url_read = rtmp_read,
|
|
.url_write = rtmp_write,
|
|
.url_close = rtmp_close,
|
|
.priv_data_size = sizeof(RTMPContext),
|
|
.flags = URL_PROTOCOL_FLAG_NETWORK,
|
|
.priv_data_class = &rtmpte_class,
|
|
};
|
|
|
|
static const AVClass rtmpts_class = {
|
|
.class_name = "rtmpts",
|
|
.item_name = av_default_item_name,
|
|
.option = rtmp_options,
|
|
.version = LIBAVUTIL_VERSION_INT,
|
|
};
|
|
|
|
URLProtocol ff_rtmpts_protocol = {
|
|
.name = "rtmpts",
|
|
.url_open = rtmp_open,
|
|
.url_read = rtmp_read,
|
|
.url_write = rtmp_write,
|
|
.url_close = rtmp_close,
|
|
.priv_data_size = sizeof(RTMPContext),
|
|
.flags = URL_PROTOCOL_FLAG_NETWORK,
|
|
.priv_data_class = &rtmpts_class,
|
|
};
|