mirror of
https://git.videolan.org/git/ffmpeg.git
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3146 lines
105 KiB
C
3146 lines
105 KiB
C
/*
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* RTMP network protocol
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* Copyright (c) 2009 Konstantin 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/base64.h"
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#include "libavutil/intfloat.h"
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#include "libavutil/lfg.h"
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#include "libavutil/md5.h"
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#include "libavutil/opt.h"
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#include "libavutil/random_seed.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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#if CONFIG_ZLIB
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#include <zlib.h>
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#endif
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#define APP_MAX_LENGTH 1024
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#define PLAYPATH_MAX_LENGTH 512
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#define TCURL_MAX_LENGTH 1024
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#define FLASHVER_MAX_LENGTH 64
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#define RTMP_PKTDATA_DEFAULT_SIZE 4096
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#define RTMP_HEADER 11
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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_FCPUBLISH, ///< client FCPublishing stream (for output)
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STATE_PLAYING, ///< client has started receiving multimedia data from server
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STATE_SEEKING, ///< client has started the seek operation. Back on STATE_PLAYING when the time comes
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STATE_PUBLISHING, ///< client has started sending multimedia data to server (for output)
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STATE_RECEIVING, ///< received a publish command (for input)
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STATE_SENDING, ///< received a play command (for output)
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STATE_STOPPED, ///< the broadcast has been stopped
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} ClientState;
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typedef struct TrackedMethod {
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char *name;
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int id;
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} TrackedMethod;
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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]; ///< packet history used when reading and sending packets ([0] for reading, [1] for writing)
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int nb_prev_pkt[2]; ///< number of elements in prev_pkt
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int in_chunk_size; ///< size of the chunks incoming RTMP packets are divided into
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int out_chunk_size; ///< size of the chunks outgoing 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 stream_id; ///< ID assigned by the server for the stream
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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 receive_report_size; ///< number of bytes after which we should report the number of received bytes to the peer
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uint64_t bytes_read; ///< number of bytes read from server
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uint64_t last_bytes_read; ///< number of bytes read last reported to server
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uint32_t last_timestamp; ///< last timestamp received in a packet
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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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int has_audio; ///< presence of audio data
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int has_video; ///< presence of video data
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int received_metadata; ///< Indicates if we have received metadata about the streams
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uint8_t flv_header[RTMP_HEADER]; ///< 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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char* tcurl; ///< url of the target stream
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char* flashver; ///< version of the flash plugin
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char* swfhash; ///< SHA256 hash of the decompressed SWF file (32 bytes)
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int swfhash_len; ///< length of the SHA256 hash
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int swfsize; ///< size of the decompressed SWF file
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char* swfurl; ///< url of the swf player
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char* swfverify; ///< URL to player swf file, compute hash/size automatically
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char swfverification[42]; ///< hash of the SWF verification
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char* pageurl; ///< url of the web page
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char* subscribe; ///< name of live stream to subscribe
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int max_sent_unacked; ///< max unacked sent bytes
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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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TrackedMethod*tracked_methods; ///< tracked methods buffer
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int nb_tracked_methods; ///< number of tracked methods
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int tracked_methods_size; ///< size of the tracked methods buffer
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int listen; ///< listen mode flag
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int listen_timeout; ///< listen timeout to wait for new connections
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int nb_streamid; ///< The next stream id to return on createStream calls
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double duration; ///< Duration of the stream in seconds as returned by the server (only valid if non-zero)
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char username[50];
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char password[50];
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char auth_params[500];
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int do_reconnect;
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int auth_tried;
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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 handle_chunk_size(URLContext *s, RTMPPacket *pkt);
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static int handle_window_ack_size(URLContext *s, RTMPPacket *pkt);
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static int handle_set_peer_bw(URLContext *s, RTMPPacket *pkt);
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static int add_tracked_method(RTMPContext *rt, const char *name, int id)
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{
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int err;
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if (rt->nb_tracked_methods + 1 > rt->tracked_methods_size) {
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rt->tracked_methods_size = (rt->nb_tracked_methods + 1) * 2;
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if ((err = av_reallocp(&rt->tracked_methods, rt->tracked_methods_size *
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sizeof(*rt->tracked_methods))) < 0) {
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rt->nb_tracked_methods = 0;
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rt->tracked_methods_size = 0;
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return err;
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}
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}
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rt->tracked_methods[rt->nb_tracked_methods].name = av_strdup(name);
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if (!rt->tracked_methods[rt->nb_tracked_methods].name)
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return AVERROR(ENOMEM);
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rt->tracked_methods[rt->nb_tracked_methods].id = id;
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rt->nb_tracked_methods++;
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return 0;
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}
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static void del_tracked_method(RTMPContext *rt, int index)
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{
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memmove(&rt->tracked_methods[index], &rt->tracked_methods[index + 1],
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sizeof(*rt->tracked_methods) * (rt->nb_tracked_methods - index - 1));
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rt->nb_tracked_methods--;
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}
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static int find_tracked_method(URLContext *s, RTMPPacket *pkt, int offset,
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char **tracked_method)
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{
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RTMPContext *rt = s->priv_data;
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GetByteContext gbc;
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double pkt_id;
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int ret;
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int i;
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bytestream2_init(&gbc, pkt->data + offset, pkt->size - offset);
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if ((ret = ff_amf_read_number(&gbc, &pkt_id)) < 0)
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return ret;
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for (i = 0; i < rt->nb_tracked_methods; i++) {
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if (rt->tracked_methods[i].id != pkt_id)
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continue;
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*tracked_method = rt->tracked_methods[i].name;
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del_tracked_method(rt, i);
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break;
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}
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return 0;
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}
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static void free_tracked_methods(RTMPContext *rt)
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{
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int i;
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for (i = 0; i < rt->nb_tracked_methods; i ++)
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av_freep(&rt->tracked_methods[i].name);
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av_freep(&rt->tracked_methods);
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rt->tracked_methods_size = 0;
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rt->nb_tracked_methods = 0;
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}
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static int rtmp_send_packet(RTMPContext *rt, RTMPPacket *pkt, int track)
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{
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int ret;
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if (pkt->type == RTMP_PT_INVOKE && track) {
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GetByteContext gbc;
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char name[128];
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double pkt_id;
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int len;
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bytestream2_init(&gbc, pkt->data, pkt->size);
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if ((ret = ff_amf_read_string(&gbc, name, sizeof(name), &len)) < 0)
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goto fail;
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if ((ret = ff_amf_read_number(&gbc, &pkt_id)) < 0)
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goto fail;
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if ((ret = add_tracked_method(rt, name, pkt_id)) < 0)
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goto fail;
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}
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ret = ff_rtmp_packet_write(rt->stream, pkt, rt->out_chunk_size,
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&rt->prev_pkt[1], &rt->nb_prev_pkt[1]);
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fail:
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ff_rtmp_packet_destroy(pkt);
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return ret;
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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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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 + APP_MAX_LENGTH)) < 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_string2(&p, rt->app, rt->auth_params);
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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 || rt->swfverify) {
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ff_amf_write_field_name(&p, "swfUrl");
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if (rt->swfurl)
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ff_amf_write_string(&p, rt->swfurl);
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else
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ff_amf_write_string(&p, rt->swfverify);
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}
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ff_amf_write_field_name(&p, "tcUrl");
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ff_amf_write_string2(&p, rt->tcurl, rt->auth_params);
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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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|
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if (rt->conn) {
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char *param = rt->conn;
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|
|
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// Write arbitrary AMF data to the Connect message.
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while (param) {
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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.size = p - pkt.data;
|
|
|
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return rtmp_send_packet(rt, &pkt, 1);
|
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}
|
|
|
|
|
|
#define RTMP_CTRL_ABORT_MESSAGE (2)
|
|
|
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static int read_connect(URLContext *s, RTMPContext *rt)
|
|
{
|
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RTMPPacket pkt = { 0 };
|
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uint8_t *p;
|
|
const uint8_t *cp;
|
|
int ret;
|
|
char command[64];
|
|
int stringlen;
|
|
double seqnum;
|
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uint8_t tmpstr[256];
|
|
GetByteContext gbc;
|
|
|
|
// handle RTMP Protocol Control Messages
|
|
for (;;) {
|
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if ((ret = ff_rtmp_packet_read(rt->stream, &pkt, rt->in_chunk_size,
|
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&rt->prev_pkt[0], &rt->nb_prev_pkt[0])) < 0)
|
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return ret;
|
|
#ifdef DEBUG
|
|
ff_rtmp_packet_dump(s, &pkt);
|
|
#endif
|
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if (pkt.type == RTMP_PT_CHUNK_SIZE) {
|
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if ((ret = handle_chunk_size(s, &pkt)) < 0) {
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ff_rtmp_packet_destroy(&pkt);
|
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return ret;
|
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}
|
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} else if (pkt.type == RTMP_CTRL_ABORT_MESSAGE) {
|
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av_log(s, AV_LOG_ERROR, "received abort message\n");
|
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ff_rtmp_packet_destroy(&pkt);
|
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return AVERROR_UNKNOWN;
|
|
} else if (pkt.type == RTMP_PT_BYTES_READ) {
|
|
av_log(s, AV_LOG_TRACE, "received acknowledgement\n");
|
|
} else if (pkt.type == RTMP_PT_WINDOW_ACK_SIZE) {
|
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if ((ret = handle_window_ack_size(s, &pkt)) < 0) {
|
|
ff_rtmp_packet_destroy(&pkt);
|
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return ret;
|
|
}
|
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} else if (pkt.type == RTMP_PT_SET_PEER_BW) {
|
|
if ((ret = handle_set_peer_bw(s, &pkt)) < 0) {
|
|
ff_rtmp_packet_destroy(&pkt);
|
|
return ret;
|
|
}
|
|
} else if (pkt.type == RTMP_PT_INVOKE) {
|
|
// received RTMP Command Message
|
|
break;
|
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} else {
|
|
av_log(s, AV_LOG_ERROR, "Unknown control message type (%d)\n", pkt.type);
|
|
}
|
|
ff_rtmp_packet_destroy(&pkt);
|
|
}
|
|
|
|
cp = pkt.data;
|
|
bytestream2_init(&gbc, cp, pkt.size);
|
|
if (ff_amf_read_string(&gbc, command, sizeof(command), &stringlen)) {
|
|
av_log(s, AV_LOG_ERROR, "Unable to read command string\n");
|
|
ff_rtmp_packet_destroy(&pkt);
|
|
return AVERROR_INVALIDDATA;
|
|
}
|
|
if (strcmp(command, "connect")) {
|
|
av_log(s, AV_LOG_ERROR, "Expecting connect, got %s\n", command);
|
|
ff_rtmp_packet_destroy(&pkt);
|
|
return AVERROR_INVALIDDATA;
|
|
}
|
|
ret = ff_amf_read_number(&gbc, &seqnum);
|
|
if (ret)
|
|
av_log(s, AV_LOG_WARNING, "SeqNum not found\n");
|
|
/* Here one could parse an AMF Object with data as flashVers and others. */
|
|
ret = ff_amf_get_field_value(gbc.buffer,
|
|
gbc.buffer + bytestream2_get_bytes_left(&gbc),
|
|
"app", tmpstr, sizeof(tmpstr));
|
|
if (ret)
|
|
av_log(s, AV_LOG_WARNING, "App field not found in connect\n");
|
|
if (!ret && strcmp(tmpstr, rt->app))
|
|
av_log(s, AV_LOG_WARNING, "App field don't match up: %s <-> %s\n",
|
|
tmpstr, rt->app);
|
|
ff_rtmp_packet_destroy(&pkt);
|
|
|
|
// Send Window Acknowledgement Size (as defined in specification)
|
|
if ((ret = ff_rtmp_packet_create(&pkt, RTMP_NETWORK_CHANNEL,
|
|
RTMP_PT_WINDOW_ACK_SIZE, 0, 4)) < 0)
|
|
return ret;
|
|
p = pkt.data;
|
|
// Inform the peer about how often we want acknowledgements about what
|
|
// we send. (We don't check for the acknowledgements currently.)
|
|
bytestream_put_be32(&p, rt->max_sent_unacked);
|
|
pkt.size = p - pkt.data;
|
|
ret = ff_rtmp_packet_write(rt->stream, &pkt, rt->out_chunk_size,
|
|
&rt->prev_pkt[1], &rt->nb_prev_pkt[1]);
|
|
ff_rtmp_packet_destroy(&pkt);
|
|
if (ret < 0)
|
|
return ret;
|
|
// Set Peer Bandwidth
|
|
if ((ret = ff_rtmp_packet_create(&pkt, RTMP_NETWORK_CHANNEL,
|
|
RTMP_PT_SET_PEER_BW, 0, 5)) < 0)
|
|
return ret;
|
|
p = pkt.data;
|
|
// Tell the peer to only send this many bytes unless it gets acknowledgements.
|
|
// This could be any arbitrary value we want here.
|
|
bytestream_put_be32(&p, rt->max_sent_unacked);
|
|
bytestream_put_byte(&p, 2); // dynamic
|
|
pkt.size = p - pkt.data;
|
|
ret = ff_rtmp_packet_write(rt->stream, &pkt, rt->out_chunk_size,
|
|
&rt->prev_pkt[1], &rt->nb_prev_pkt[1]);
|
|
ff_rtmp_packet_destroy(&pkt);
|
|
if (ret < 0)
|
|
return ret;
|
|
|
|
// User control
|
|
if ((ret = ff_rtmp_packet_create(&pkt, RTMP_NETWORK_CHANNEL,
|
|
RTMP_PT_USER_CONTROL, 0, 6)) < 0)
|
|
return ret;
|
|
|
|
p = pkt.data;
|
|
bytestream_put_be16(&p, 0); // 0 -> Stream Begin
|
|
bytestream_put_be32(&p, 0); // Stream 0
|
|
ret = ff_rtmp_packet_write(rt->stream, &pkt, rt->out_chunk_size,
|
|
&rt->prev_pkt[1], &rt->nb_prev_pkt[1]);
|
|
ff_rtmp_packet_destroy(&pkt);
|
|
if (ret < 0)
|
|
return ret;
|
|
|
|
// Chunk size
|
|
if ((ret = ff_rtmp_packet_create(&pkt, RTMP_NETWORK_CHANNEL,
|
|
RTMP_PT_CHUNK_SIZE, 0, 4)) < 0)
|
|
return ret;
|
|
|
|
p = pkt.data;
|
|
bytestream_put_be32(&p, rt->out_chunk_size);
|
|
ret = ff_rtmp_packet_write(rt->stream, &pkt, rt->out_chunk_size,
|
|
&rt->prev_pkt[1], &rt->nb_prev_pkt[1]);
|
|
ff_rtmp_packet_destroy(&pkt);
|
|
if (ret < 0)
|
|
return ret;
|
|
|
|
// Send _result NetConnection.Connect.Success to connect
|
|
if ((ret = ff_rtmp_packet_create(&pkt, RTMP_SYSTEM_CHANNEL,
|
|
RTMP_PT_INVOKE, 0,
|
|
RTMP_PKTDATA_DEFAULT_SIZE)) < 0)
|
|
return ret;
|
|
|
|
p = pkt.data;
|
|
ff_amf_write_string(&p, "_result");
|
|
ff_amf_write_number(&p, seqnum);
|
|
|
|
ff_amf_write_object_start(&p);
|
|
ff_amf_write_field_name(&p, "fmsVer");
|
|
ff_amf_write_string(&p, "FMS/3,0,1,123");
|
|
ff_amf_write_field_name(&p, "capabilities");
|
|
ff_amf_write_number(&p, 31);
|
|
ff_amf_write_object_end(&p);
|
|
|
|
ff_amf_write_object_start(&p);
|
|
ff_amf_write_field_name(&p, "level");
|
|
ff_amf_write_string(&p, "status");
|
|
ff_amf_write_field_name(&p, "code");
|
|
ff_amf_write_string(&p, "NetConnection.Connect.Success");
|
|
ff_amf_write_field_name(&p, "description");
|
|
ff_amf_write_string(&p, "Connection succeeded.");
|
|
ff_amf_write_field_name(&p, "objectEncoding");
|
|
ff_amf_write_number(&p, 0);
|
|
ff_amf_write_object_end(&p);
|
|
|
|
pkt.size = p - pkt.data;
|
|
ret = ff_rtmp_packet_write(rt->stream, &pkt, rt->out_chunk_size,
|
|
&rt->prev_pkt[1], &rt->nb_prev_pkt[1]);
|
|
ff_rtmp_packet_destroy(&pkt);
|
|
if (ret < 0)
|
|
return ret;
|
|
|
|
if ((ret = ff_rtmp_packet_create(&pkt, RTMP_SYSTEM_CHANNEL,
|
|
RTMP_PT_INVOKE, 0, 30)) < 0)
|
|
return ret;
|
|
p = pkt.data;
|
|
ff_amf_write_string(&p, "onBWDone");
|
|
ff_amf_write_number(&p, 0);
|
|
ff_amf_write_null(&p);
|
|
ff_amf_write_number(&p, 8192);
|
|
pkt.size = p - pkt.data;
|
|
ret = ff_rtmp_packet_write(rt->stream, &pkt, rt->out_chunk_size,
|
|
&rt->prev_pkt[1], &rt->nb_prev_pkt[1]);
|
|
ff_rtmp_packet_destroy(&pkt);
|
|
|
|
return ret;
|
|
}
|
|
|
|
/**
|
|
* 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);
|
|
|
|
return rtmp_send_packet(rt, &pkt, 1);
|
|
}
|
|
|
|
/**
|
|
* Generate 'FCPublish' call and send it to the server. It should make
|
|
* the server prepare 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);
|
|
|
|
return rtmp_send_packet(rt, &pkt, 1);
|
|
}
|
|
|
|
/**
|
|
* 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);
|
|
|
|
return rtmp_send_packet(rt, &pkt, 0);
|
|
}
|
|
|
|
/**
|
|
* Generate 'createStream' call and send it to the server. It should make
|
|
* the server allocate some channel for media streams.
|
|
*/
|
|
static int gen_create_stream(URLContext *s, RTMPContext *rt)
|
|
{
|
|
RTMPPacket pkt;
|
|
uint8_t *p;
|
|
int ret;
|
|
|
|
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, ++rt->nb_invokes);
|
|
ff_amf_write_null(&p);
|
|
|
|
return rtmp_send_packet(rt, &pkt, 1);
|
|
}
|
|
|
|
|
|
/**
|
|
* 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->stream_id);
|
|
|
|
return rtmp_send_packet(rt, &pkt, 0);
|
|
}
|
|
|
|
/**
|
|
* Generate 'getStreamLength' call and send it to the server. If the server
|
|
* knows the duration of the selected stream, it will reply with the duration
|
|
* in seconds.
|
|
*/
|
|
static int gen_get_stream_length(URLContext *s, RTMPContext *rt)
|
|
{
|
|
RTMPPacket pkt;
|
|
uint8_t *p;
|
|
int ret;
|
|
|
|
if ((ret = ff_rtmp_packet_create(&pkt, RTMP_SOURCE_CHANNEL, RTMP_PT_INVOKE,
|
|
0, 31 + strlen(rt->playpath))) < 0)
|
|
return ret;
|
|
|
|
p = pkt.data;
|
|
ff_amf_write_string(&p, "getStreamLength");
|
|
ff_amf_write_number(&p, ++rt->nb_invokes);
|
|
ff_amf_write_null(&p);
|
|
ff_amf_write_string(&p, rt->playpath);
|
|
|
|
return rtmp_send_packet(rt, &pkt, 1);
|
|
}
|
|
|
|
/**
|
|
* 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_USER_CONTROL,
|
|
1, 10)) < 0)
|
|
return ret;
|
|
|
|
p = pkt.data;
|
|
bytestream_put_be16(&p, 3); // SetBuffer Length
|
|
bytestream_put_be32(&p, rt->stream_id);
|
|
bytestream_put_be32(&p, rt->client_buffer_time);
|
|
|
|
return rtmp_send_packet(rt, &pkt, 0);
|
|
}
|
|
|
|
/**
|
|
* 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);
|
|
|
|
if ((ret = ff_rtmp_packet_create(&pkt, RTMP_SOURCE_CHANNEL, RTMP_PT_INVOKE,
|
|
0, 29 + strlen(rt->playpath))) < 0)
|
|
return ret;
|
|
|
|
pkt.extra = rt->stream_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 * 1000);
|
|
|
|
return rtmp_send_packet(rt, &pkt, 1);
|
|
}
|
|
|
|
static int gen_seek(URLContext *s, RTMPContext *rt, int64_t timestamp)
|
|
{
|
|
RTMPPacket pkt;
|
|
uint8_t *p;
|
|
int ret;
|
|
|
|
av_log(s, AV_LOG_DEBUG, "Sending seek command for timestamp %"PRId64"\n",
|
|
timestamp);
|
|
|
|
if ((ret = ff_rtmp_packet_create(&pkt, 3, RTMP_PT_INVOKE, 0, 26)) < 0)
|
|
return ret;
|
|
|
|
pkt.extra = rt->stream_id;
|
|
|
|
p = pkt.data;
|
|
ff_amf_write_string(&p, "seek");
|
|
ff_amf_write_number(&p, 0); //no tracking back responses
|
|
ff_amf_write_null(&p); //as usual, the first null param
|
|
ff_amf_write_number(&p, timestamp); //where we want to jump
|
|
|
|
return rtmp_send_packet(rt, &pkt, 1);
|
|
}
|
|
|
|
/**
|
|
* Generate a pause packet that either pauses or unpauses the current stream.
|
|
*/
|
|
static int gen_pause(URLContext *s, RTMPContext *rt, int pause, uint32_t timestamp)
|
|
{
|
|
RTMPPacket pkt;
|
|
uint8_t *p;
|
|
int ret;
|
|
|
|
av_log(s, AV_LOG_DEBUG, "Sending pause command for timestamp %d\n",
|
|
timestamp);
|
|
|
|
if ((ret = ff_rtmp_packet_create(&pkt, 3, RTMP_PT_INVOKE, 0, 29)) < 0)
|
|
return ret;
|
|
|
|
pkt.extra = rt->stream_id;
|
|
|
|
p = pkt.data;
|
|
ff_amf_write_string(&p, "pause");
|
|
ff_amf_write_number(&p, 0); //no tracking back responses
|
|
ff_amf_write_null(&p); //as usual, the first null param
|
|
ff_amf_write_bool(&p, pause); // pause or unpause
|
|
ff_amf_write_number(&p, timestamp); //where we pause the stream
|
|
|
|
return rtmp_send_packet(rt, &pkt, 1);
|
|
}
|
|
|
|
/**
|
|
* 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->stream_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");
|
|
|
|
return rtmp_send_packet(rt, &pkt, 1);
|
|
}
|
|
|
|
/**
|
|
* 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->size < 6) {
|
|
av_log(s, AV_LOG_ERROR, "Too short ping packet (%d)\n",
|
|
ppkt->size);
|
|
return AVERROR_INVALIDDATA;
|
|
}
|
|
|
|
if ((ret = ff_rtmp_packet_create(&pkt, RTMP_NETWORK_CHANNEL,RTMP_PT_USER_CONTROL,
|
|
ppkt->timestamp + 1, 6)) < 0)
|
|
return ret;
|
|
|
|
p = pkt.data;
|
|
bytestream_put_be16(&p, 7); // PingResponse
|
|
bytestream_put_be32(&p, AV_RB32(ppkt->data+2));
|
|
|
|
return rtmp_send_packet(rt, &pkt, 0);
|
|
}
|
|
|
|
/**
|
|
* Generate SWF verification message and send it to the server.
|
|
*/
|
|
static int gen_swf_verification(URLContext *s, RTMPContext *rt)
|
|
{
|
|
RTMPPacket pkt;
|
|
uint8_t *p;
|
|
int ret;
|
|
|
|
av_log(s, AV_LOG_DEBUG, "Sending SWF verification...\n");
|
|
if ((ret = ff_rtmp_packet_create(&pkt, RTMP_NETWORK_CHANNEL, RTMP_PT_USER_CONTROL,
|
|
0, 44)) < 0)
|
|
return ret;
|
|
|
|
p = pkt.data;
|
|
bytestream_put_be16(&p, 27);
|
|
memcpy(p, rt->swfverification, 42);
|
|
|
|
return rtmp_send_packet(rt, &pkt, 0);
|
|
}
|
|
|
|
/**
|
|
* Generate window acknowledgement size message and send it to the server.
|
|
*/
|
|
static int gen_window_ack_size(URLContext *s, RTMPContext *rt)
|
|
{
|
|
RTMPPacket pkt;
|
|
uint8_t *p;
|
|
int ret;
|
|
|
|
if ((ret = ff_rtmp_packet_create(&pkt, RTMP_NETWORK_CHANNEL, RTMP_PT_WINDOW_ACK_SIZE,
|
|
0, 4)) < 0)
|
|
return ret;
|
|
|
|
p = pkt.data;
|
|
bytestream_put_be32(&p, rt->max_sent_unacked);
|
|
|
|
return rtmp_send_packet(rt, &pkt, 0);
|
|
}
|
|
|
|
/**
|
|
* 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);
|
|
|
|
return rtmp_send_packet(rt, &pkt, 1);
|
|
}
|
|
|
|
/**
|
|
* 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);
|
|
|
|
return rtmp_send_packet(rt, &pkt, 0);
|
|
}
|
|
|
|
static int gen_fcsubscribe_stream(URLContext *s, RTMPContext *rt,
|
|
const char *subscribe)
|
|
{
|
|
RTMPPacket pkt;
|
|
uint8_t *p;
|
|
int ret;
|
|
|
|
if ((ret = ff_rtmp_packet_create(&pkt, RTMP_SYSTEM_CHANNEL, RTMP_PT_INVOKE,
|
|
0, 27 + strlen(subscribe))) < 0)
|
|
return ret;
|
|
|
|
p = pkt.data;
|
|
ff_amf_write_string(&p, "FCSubscribe");
|
|
ff_amf_write_number(&p, ++rt->nb_invokes);
|
|
ff_amf_write_null(&p);
|
|
ff_amf_write_string(&p, subscribe);
|
|
|
|
return rtmp_send_packet(rt, &pkt, 1);
|
|
}
|
|
|
|
/**
|
|
* 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;
|
|
}
|
|
|
|
static int rtmp_calc_swf_verification(URLContext *s, RTMPContext *rt,
|
|
uint8_t *buf)
|
|
{
|
|
uint8_t *p;
|
|
int ret;
|
|
|
|
if (rt->swfhash_len != 32) {
|
|
av_log(s, AV_LOG_ERROR,
|
|
"Hash of the decompressed SWF file is not 32 bytes long.\n");
|
|
return AVERROR(EINVAL);
|
|
}
|
|
|
|
p = &rt->swfverification[0];
|
|
bytestream_put_byte(&p, 1);
|
|
bytestream_put_byte(&p, 1);
|
|
bytestream_put_be32(&p, rt->swfsize);
|
|
bytestream_put_be32(&p, rt->swfsize);
|
|
|
|
if ((ret = ff_rtmp_calc_digest(rt->swfhash, 32, 0, buf, 32, p)) < 0)
|
|
return ret;
|
|
|
|
return 0;
|
|
}
|
|
|
|
#if CONFIG_ZLIB
|
|
static int rtmp_uncompress_swfplayer(uint8_t *in_data, int64_t in_size,
|
|
uint8_t **out_data, int64_t *out_size)
|
|
{
|
|
z_stream zs = { 0 };
|
|
void *ptr;
|
|
int size;
|
|
int ret = 0;
|
|
|
|
zs.avail_in = in_size;
|
|
zs.next_in = in_data;
|
|
ret = inflateInit(&zs);
|
|
if (ret != Z_OK)
|
|
return AVERROR_UNKNOWN;
|
|
|
|
do {
|
|
uint8_t tmp_buf[16384];
|
|
|
|
zs.avail_out = sizeof(tmp_buf);
|
|
zs.next_out = tmp_buf;
|
|
|
|
ret = inflate(&zs, Z_NO_FLUSH);
|
|
if (ret != Z_OK && ret != Z_STREAM_END) {
|
|
ret = AVERROR_UNKNOWN;
|
|
goto fail;
|
|
}
|
|
|
|
size = sizeof(tmp_buf) - zs.avail_out;
|
|
if (!(ptr = av_realloc(*out_data, *out_size + size))) {
|
|
ret = AVERROR(ENOMEM);
|
|
goto fail;
|
|
}
|
|
*out_data = ptr;
|
|
|
|
memcpy(*out_data + *out_size, tmp_buf, size);
|
|
*out_size += size;
|
|
} while (zs.avail_out == 0);
|
|
|
|
fail:
|
|
inflateEnd(&zs);
|
|
return ret;
|
|
}
|
|
#endif
|
|
|
|
static int rtmp_calc_swfhash(URLContext *s)
|
|
{
|
|
RTMPContext *rt = s->priv_data;
|
|
uint8_t *in_data = NULL, *out_data = NULL, *swfdata;
|
|
int64_t in_size;
|
|
URLContext *stream;
|
|
char swfhash[32];
|
|
int swfsize;
|
|
int ret = 0;
|
|
|
|
/* Get the SWF player file. */
|
|
if ((ret = ffurl_open_whitelist(&stream, rt->swfverify, AVIO_FLAG_READ,
|
|
&s->interrupt_callback, NULL,
|
|
s->protocol_whitelist, s->protocol_blacklist, s)) < 0) {
|
|
av_log(s, AV_LOG_ERROR, "Cannot open connection %s.\n", rt->swfverify);
|
|
goto fail;
|
|
}
|
|
|
|
if ((in_size = ffurl_seek(stream, 0, AVSEEK_SIZE)) < 0) {
|
|
ret = AVERROR(EIO);
|
|
goto fail;
|
|
}
|
|
|
|
if (!(in_data = av_malloc(in_size))) {
|
|
ret = AVERROR(ENOMEM);
|
|
goto fail;
|
|
}
|
|
|
|
if ((ret = ffurl_read_complete(stream, in_data, in_size)) < 0)
|
|
goto fail;
|
|
|
|
if (in_size < 3) {
|
|
ret = AVERROR_INVALIDDATA;
|
|
goto fail;
|
|
}
|
|
|
|
if (!memcmp(in_data, "CWS", 3)) {
|
|
#if CONFIG_ZLIB
|
|
int64_t out_size;
|
|
/* Decompress the SWF player file using Zlib. */
|
|
if (!(out_data = av_malloc(8))) {
|
|
ret = AVERROR(ENOMEM);
|
|
goto fail;
|
|
}
|
|
*in_data = 'F'; // magic stuff
|
|
memcpy(out_data, in_data, 8);
|
|
out_size = 8;
|
|
|
|
if ((ret = rtmp_uncompress_swfplayer(in_data + 8, in_size - 8,
|
|
&out_data, &out_size)) < 0)
|
|
goto fail;
|
|
swfsize = out_size;
|
|
swfdata = out_data;
|
|
#else
|
|
av_log(s, AV_LOG_ERROR,
|
|
"Zlib is required for decompressing the SWF player file.\n");
|
|
ret = AVERROR(EINVAL);
|
|
goto fail;
|
|
#endif
|
|
} else {
|
|
swfsize = in_size;
|
|
swfdata = in_data;
|
|
}
|
|
|
|
/* Compute the SHA256 hash of the SWF player file. */
|
|
if ((ret = ff_rtmp_calc_digest(swfdata, swfsize, 0,
|
|
"Genuine Adobe Flash Player 001", 30,
|
|
swfhash)) < 0)
|
|
goto fail;
|
|
|
|
/* Set SWFVerification parameters. */
|
|
av_opt_set_bin(rt, "rtmp_swfhash", swfhash, 32, 0);
|
|
rt->swfsize = swfsize;
|
|
|
|
fail:
|
|
av_freep(&in_data);
|
|
av_freep(&out_data);
|
|
ffurl_close(stream);
|
|
return ret;
|
|
}
|
|
|
|
/**
|
|
* 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 (CONFIG_FFRTMPCRYPT_PROTOCOL && rt->encrypted) {
|
|
/* 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);
|
|
}
|
|
}
|
|
|
|
/* Generate SWFVerification token (SHA256 HMAC hash of decompressed SWF,
|
|
* key are the last 32 bytes of the server handshake. */
|
|
if (rt->swfsize) {
|
|
if ((ret = rtmp_calc_swf_verification(s, rt, serverdata + 1 +
|
|
RTMP_HANDSHAKE_PACKET_SIZE - 32)) < 0)
|
|
return ret;
|
|
}
|
|
|
|
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 (CONFIG_FFRTMPCRYPT_PROTOCOL && rt->encrypted) {
|
|
/* 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 (CONFIG_FFRTMPCRYPT_PROTOCOL && rt->encrypted) {
|
|
/* 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 (CONFIG_FFRTMPCRYPT_PROTOCOL && rt->encrypted) {
|
|
/* Set RC4 keys for encryption and update the keystreams. */
|
|
if ((ret = ff_rtmpe_update_keystream(rt->stream)) < 0)
|
|
return ret;
|
|
}
|
|
} else {
|
|
if (CONFIG_FFRTMPCRYPT_PROTOCOL && rt->encrypted) {
|
|
/* 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 (CONFIG_FFRTMPCRYPT_PROTOCOL && rt->encrypted) {
|
|
/* Set RC4 keys for encryption and update the keystreams. */
|
|
if ((ret = ff_rtmpe_update_keystream(rt->stream)) < 0)
|
|
return ret;
|
|
}
|
|
}
|
|
|
|
return 0;
|
|
}
|
|
|
|
static int rtmp_receive_hs_packet(RTMPContext* rt, uint32_t *first_int,
|
|
uint32_t *second_int, char *arraydata,
|
|
int size)
|
|
{
|
|
int inoutsize;
|
|
|
|
inoutsize = ffurl_read_complete(rt->stream, arraydata,
|
|
RTMP_HANDSHAKE_PACKET_SIZE);
|
|
if (inoutsize <= 0)
|
|
return AVERROR(EIO);
|
|
if (inoutsize != RTMP_HANDSHAKE_PACKET_SIZE) {
|
|
av_log(rt, AV_LOG_ERROR, "Erroneous Message size %d"
|
|
" not following standard\n", (int)inoutsize);
|
|
return AVERROR(EINVAL);
|
|
}
|
|
|
|
*first_int = AV_RB32(arraydata);
|
|
*second_int = AV_RB32(arraydata + 4);
|
|
return 0;
|
|
}
|
|
|
|
static int rtmp_send_hs_packet(RTMPContext* rt, uint32_t first_int,
|
|
uint32_t second_int, char *arraydata, int size)
|
|
{
|
|
int inoutsize;
|
|
|
|
AV_WB32(arraydata, first_int);
|
|
AV_WB32(arraydata + 4, second_int);
|
|
inoutsize = ffurl_write(rt->stream, arraydata,
|
|
RTMP_HANDSHAKE_PACKET_SIZE);
|
|
if (inoutsize != RTMP_HANDSHAKE_PACKET_SIZE) {
|
|
av_log(rt, AV_LOG_ERROR, "Unable to write answer\n");
|
|
return AVERROR(EIO);
|
|
}
|
|
|
|
return 0;
|
|
}
|
|
|
|
/**
|
|
* rtmp handshake server side
|
|
*/
|
|
static int rtmp_server_handshake(URLContext *s, RTMPContext *rt)
|
|
{
|
|
uint8_t buffer[RTMP_HANDSHAKE_PACKET_SIZE];
|
|
uint32_t hs_epoch;
|
|
uint32_t hs_my_epoch;
|
|
uint8_t hs_c1[RTMP_HANDSHAKE_PACKET_SIZE];
|
|
uint8_t hs_s1[RTMP_HANDSHAKE_PACKET_SIZE];
|
|
uint32_t zeroes;
|
|
uint32_t temp = 0;
|
|
int randomidx = 0;
|
|
int inoutsize = 0;
|
|
int ret;
|
|
|
|
inoutsize = ffurl_read_complete(rt->stream, buffer, 1); // Receive C0
|
|
if (inoutsize <= 0) {
|
|
av_log(s, AV_LOG_ERROR, "Unable to read handshake\n");
|
|
return AVERROR(EIO);
|
|
}
|
|
// Check Version
|
|
if (buffer[0] != 3) {
|
|
av_log(s, AV_LOG_ERROR, "RTMP protocol version mismatch\n");
|
|
return AVERROR(EIO);
|
|
}
|
|
if (ffurl_write(rt->stream, buffer, 1) <= 0) { // Send S0
|
|
av_log(s, AV_LOG_ERROR,
|
|
"Unable to write answer - RTMP S0\n");
|
|
return AVERROR(EIO);
|
|
}
|
|
/* Receive C1 */
|
|
ret = rtmp_receive_hs_packet(rt, &hs_epoch, &zeroes, hs_c1,
|
|
RTMP_HANDSHAKE_PACKET_SIZE);
|
|
if (ret) {
|
|
av_log(s, AV_LOG_ERROR, "RTMP Handshake C1 Error\n");
|
|
return ret;
|
|
}
|
|
/* Send S1 */
|
|
/* By now same epoch will be sent */
|
|
hs_my_epoch = hs_epoch;
|
|
/* Generate random */
|
|
for (randomidx = 8; randomidx < (RTMP_HANDSHAKE_PACKET_SIZE);
|
|
randomidx += 4)
|
|
AV_WB32(hs_s1 + randomidx, av_get_random_seed());
|
|
|
|
ret = rtmp_send_hs_packet(rt, hs_my_epoch, 0, hs_s1,
|
|
RTMP_HANDSHAKE_PACKET_SIZE);
|
|
if (ret) {
|
|
av_log(s, AV_LOG_ERROR, "RTMP Handshake S1 Error\n");
|
|
return ret;
|
|
}
|
|
/* Send S2 */
|
|
ret = rtmp_send_hs_packet(rt, hs_epoch, 0, hs_c1,
|
|
RTMP_HANDSHAKE_PACKET_SIZE);
|
|
if (ret) {
|
|
av_log(s, AV_LOG_ERROR, "RTMP Handshake S2 Error\n");
|
|
return ret;
|
|
}
|
|
/* Receive C2 */
|
|
ret = rtmp_receive_hs_packet(rt, &temp, &zeroes, buffer,
|
|
RTMP_HANDSHAKE_PACKET_SIZE);
|
|
if (ret) {
|
|
av_log(s, AV_LOG_ERROR, "RTMP Handshake C2 Error\n");
|
|
return ret;
|
|
}
|
|
if (temp != hs_my_epoch)
|
|
av_log(s, AV_LOG_WARNING,
|
|
"Erroneous C2 Message epoch does not match up with C1 epoch\n");
|
|
if (memcmp(buffer + 8, hs_s1 + 8,
|
|
RTMP_HANDSHAKE_PACKET_SIZE - 8))
|
|
av_log(s, AV_LOG_WARNING,
|
|
"Erroneous C2 Message random does not match up\n");
|
|
|
|
return 0;
|
|
}
|
|
|
|
static int handle_chunk_size(URLContext *s, RTMPPacket *pkt)
|
|
{
|
|
RTMPContext *rt = s->priv_data;
|
|
int ret;
|
|
|
|
if (pkt->size < 4) {
|
|
av_log(s, AV_LOG_ERROR,
|
|
"Too short chunk size change packet (%d)\n",
|
|
pkt->size);
|
|
return AVERROR_INVALIDDATA;
|
|
}
|
|
|
|
if (!rt->is_input) {
|
|
/* Send the same chunk size change packet back to the server,
|
|
* setting the outgoing chunk size to the same as the incoming one. */
|
|
if ((ret = ff_rtmp_packet_write(rt->stream, pkt, rt->out_chunk_size,
|
|
&rt->prev_pkt[1], &rt->nb_prev_pkt[1])) < 0)
|
|
return ret;
|
|
rt->out_chunk_size = AV_RB32(pkt->data);
|
|
}
|
|
|
|
rt->in_chunk_size = AV_RB32(pkt->data);
|
|
if (rt->in_chunk_size <= 0) {
|
|
av_log(s, AV_LOG_ERROR, "Incorrect chunk size %d\n",
|
|
rt->in_chunk_size);
|
|
return AVERROR_INVALIDDATA;
|
|
}
|
|
av_log(s, AV_LOG_DEBUG, "New incoming chunk size = %d\n",
|
|
rt->in_chunk_size);
|
|
|
|
return 0;
|
|
}
|
|
|
|
static int handle_user_control(URLContext *s, RTMPPacket *pkt)
|
|
{
|
|
RTMPContext *rt = s->priv_data;
|
|
int t, ret;
|
|
|
|
if (pkt->size < 2) {
|
|
av_log(s, AV_LOG_ERROR, "Too short user control packet (%d)\n",
|
|
pkt->size);
|
|
return AVERROR_INVALIDDATA;
|
|
}
|
|
|
|
t = AV_RB16(pkt->data);
|
|
if (t == 6) { // PingRequest
|
|
if ((ret = gen_pong(s, rt, pkt)) < 0)
|
|
return ret;
|
|
} else if (t == 26) {
|
|
if (rt->swfsize) {
|
|
if ((ret = gen_swf_verification(s, rt)) < 0)
|
|
return ret;
|
|
} else {
|
|
av_log(s, AV_LOG_WARNING, "Ignoring SWFVerification request.\n");
|
|
}
|
|
}
|
|
|
|
return 0;
|
|
}
|
|
|
|
static int handle_set_peer_bw(URLContext *s, RTMPPacket *pkt)
|
|
{
|
|
RTMPContext *rt = s->priv_data;
|
|
|
|
if (pkt->size < 4) {
|
|
av_log(s, AV_LOG_ERROR,
|
|
"Peer bandwidth packet is less than 4 bytes long (%d)\n",
|
|
pkt->size);
|
|
return AVERROR_INVALIDDATA;
|
|
}
|
|
|
|
// We currently don't check how much the peer has acknowledged of
|
|
// what we have sent. To do that properly, we should call
|
|
// gen_window_ack_size here, to tell the peer that we want an
|
|
// acknowledgement with (at least) that interval.
|
|
rt->max_sent_unacked = AV_RB32(pkt->data);
|
|
if (rt->max_sent_unacked <= 0) {
|
|
av_log(s, AV_LOG_ERROR, "Incorrect set peer bandwidth %d\n",
|
|
rt->max_sent_unacked);
|
|
return AVERROR_INVALIDDATA;
|
|
|
|
}
|
|
av_log(s, AV_LOG_DEBUG, "Max sent, unacked = %d\n", rt->max_sent_unacked);
|
|
|
|
return 0;
|
|
}
|
|
|
|
static int handle_window_ack_size(URLContext *s, RTMPPacket *pkt)
|
|
{
|
|
RTMPContext *rt = s->priv_data;
|
|
|
|
if (pkt->size < 4) {
|
|
av_log(s, AV_LOG_ERROR,
|
|
"Too short window acknowledgement size packet (%d)\n",
|
|
pkt->size);
|
|
return AVERROR_INVALIDDATA;
|
|
}
|
|
|
|
rt->receive_report_size = AV_RB32(pkt->data);
|
|
if (rt->receive_report_size <= 0) {
|
|
av_log(s, AV_LOG_ERROR, "Incorrect window acknowledgement size %d\n",
|
|
rt->receive_report_size);
|
|
return AVERROR_INVALIDDATA;
|
|
}
|
|
av_log(s, AV_LOG_DEBUG, "Window acknowledgement size = %d\n", rt->receive_report_size);
|
|
// Send an Acknowledgement packet after receiving half the maximum
|
|
// size, to make sure the peer can keep on sending without waiting
|
|
// for acknowledgements.
|
|
rt->receive_report_size >>= 1;
|
|
|
|
return 0;
|
|
}
|
|
|
|
static int do_adobe_auth(RTMPContext *rt, const char *user, const char *salt,
|
|
const char *opaque, const char *challenge)
|
|
{
|
|
uint8_t hash[16];
|
|
char hashstr[AV_BASE64_SIZE(sizeof(hash))], challenge2[10];
|
|
struct AVMD5 *md5 = av_md5_alloc();
|
|
if (!md5)
|
|
return AVERROR(ENOMEM);
|
|
|
|
snprintf(challenge2, sizeof(challenge2), "%08x", av_get_random_seed());
|
|
|
|
av_md5_init(md5);
|
|
av_md5_update(md5, user, strlen(user));
|
|
av_md5_update(md5, salt, strlen(salt));
|
|
av_md5_update(md5, rt->password, strlen(rt->password));
|
|
av_md5_final(md5, hash);
|
|
av_base64_encode(hashstr, sizeof(hashstr), hash,
|
|
sizeof(hash));
|
|
av_md5_init(md5);
|
|
av_md5_update(md5, hashstr, strlen(hashstr));
|
|
if (opaque)
|
|
av_md5_update(md5, opaque, strlen(opaque));
|
|
else if (challenge)
|
|
av_md5_update(md5, challenge, strlen(challenge));
|
|
av_md5_update(md5, challenge2, strlen(challenge2));
|
|
av_md5_final(md5, hash);
|
|
av_base64_encode(hashstr, sizeof(hashstr), hash,
|
|
sizeof(hash));
|
|
snprintf(rt->auth_params, sizeof(rt->auth_params),
|
|
"?authmod=%s&user=%s&challenge=%s&response=%s",
|
|
"adobe", user, challenge2, hashstr);
|
|
if (opaque)
|
|
av_strlcatf(rt->auth_params, sizeof(rt->auth_params),
|
|
"&opaque=%s", opaque);
|
|
|
|
av_free(md5);
|
|
return 0;
|
|
}
|
|
|
|
static int do_llnw_auth(RTMPContext *rt, const char *user, const char *nonce)
|
|
{
|
|
uint8_t hash[16];
|
|
char hashstr1[33], hashstr2[33];
|
|
const char *realm = "live";
|
|
const char *method = "publish";
|
|
const char *qop = "auth";
|
|
const char *nc = "00000001";
|
|
char cnonce[10];
|
|
struct AVMD5 *md5 = av_md5_alloc();
|
|
if (!md5)
|
|
return AVERROR(ENOMEM);
|
|
|
|
snprintf(cnonce, sizeof(cnonce), "%08x", av_get_random_seed());
|
|
|
|
av_md5_init(md5);
|
|
av_md5_update(md5, user, strlen(user));
|
|
av_md5_update(md5, ":", 1);
|
|
av_md5_update(md5, realm, strlen(realm));
|
|
av_md5_update(md5, ":", 1);
|
|
av_md5_update(md5, rt->password, strlen(rt->password));
|
|
av_md5_final(md5, hash);
|
|
ff_data_to_hex(hashstr1, hash, 16, 1);
|
|
hashstr1[32] = '\0';
|
|
|
|
av_md5_init(md5);
|
|
av_md5_update(md5, method, strlen(method));
|
|
av_md5_update(md5, ":/", 2);
|
|
av_md5_update(md5, rt->app, strlen(rt->app));
|
|
if (!strchr(rt->app, '/'))
|
|
av_md5_update(md5, "/_definst_", strlen("/_definst_"));
|
|
av_md5_final(md5, hash);
|
|
ff_data_to_hex(hashstr2, hash, 16, 1);
|
|
hashstr2[32] = '\0';
|
|
|
|
av_md5_init(md5);
|
|
av_md5_update(md5, hashstr1, strlen(hashstr1));
|
|
av_md5_update(md5, ":", 1);
|
|
if (nonce)
|
|
av_md5_update(md5, nonce, strlen(nonce));
|
|
av_md5_update(md5, ":", 1);
|
|
av_md5_update(md5, nc, strlen(nc));
|
|
av_md5_update(md5, ":", 1);
|
|
av_md5_update(md5, cnonce, strlen(cnonce));
|
|
av_md5_update(md5, ":", 1);
|
|
av_md5_update(md5, qop, strlen(qop));
|
|
av_md5_update(md5, ":", 1);
|
|
av_md5_update(md5, hashstr2, strlen(hashstr2));
|
|
av_md5_final(md5, hash);
|
|
ff_data_to_hex(hashstr1, hash, 16, 1);
|
|
|
|
snprintf(rt->auth_params, sizeof(rt->auth_params),
|
|
"?authmod=%s&user=%s&nonce=%s&cnonce=%s&nc=%s&response=%s",
|
|
"llnw", user, nonce, cnonce, nc, hashstr1);
|
|
|
|
av_free(md5);
|
|
return 0;
|
|
}
|
|
|
|
static int handle_connect_error(URLContext *s, const char *desc)
|
|
{
|
|
RTMPContext *rt = s->priv_data;
|
|
char buf[300], *ptr, authmod[15];
|
|
int i = 0, ret = 0;
|
|
const char *user = "", *salt = "", *opaque = NULL,
|
|
*challenge = NULL, *cptr = NULL, *nonce = NULL;
|
|
|
|
if (!(cptr = strstr(desc, "authmod=adobe")) &&
|
|
!(cptr = strstr(desc, "authmod=llnw"))) {
|
|
av_log(s, AV_LOG_ERROR,
|
|
"Unknown connect error (unsupported authentication method?)\n");
|
|
return AVERROR_UNKNOWN;
|
|
}
|
|
cptr += strlen("authmod=");
|
|
while (*cptr && *cptr != ' ' && i < sizeof(authmod) - 1)
|
|
authmod[i++] = *cptr++;
|
|
authmod[i] = '\0';
|
|
|
|
if (!rt->username[0] || !rt->password[0]) {
|
|
av_log(s, AV_LOG_ERROR, "No credentials set\n");
|
|
return AVERROR_UNKNOWN;
|
|
}
|
|
|
|
if (strstr(desc, "?reason=authfailed")) {
|
|
av_log(s, AV_LOG_ERROR, "Incorrect username/password\n");
|
|
return AVERROR_UNKNOWN;
|
|
} else if (strstr(desc, "?reason=nosuchuser")) {
|
|
av_log(s, AV_LOG_ERROR, "Incorrect username\n");
|
|
return AVERROR_UNKNOWN;
|
|
}
|
|
|
|
if (rt->auth_tried) {
|
|
av_log(s, AV_LOG_ERROR, "Authentication failed\n");
|
|
return AVERROR_UNKNOWN;
|
|
}
|
|
|
|
rt->auth_params[0] = '\0';
|
|
|
|
if (strstr(desc, "code=403 need auth")) {
|
|
snprintf(rt->auth_params, sizeof(rt->auth_params),
|
|
"?authmod=%s&user=%s", authmod, rt->username);
|
|
return 0;
|
|
}
|
|
|
|
if (!(cptr = strstr(desc, "?reason=needauth"))) {
|
|
av_log(s, AV_LOG_ERROR, "No auth parameters found\n");
|
|
return AVERROR_UNKNOWN;
|
|
}
|
|
|
|
av_strlcpy(buf, cptr + 1, sizeof(buf));
|
|
ptr = buf;
|
|
|
|
while (ptr) {
|
|
char *next = strchr(ptr, '&');
|
|
char *value = strchr(ptr, '=');
|
|
if (next)
|
|
*next++ = '\0';
|
|
if (value) {
|
|
*value++ = '\0';
|
|
if (!strcmp(ptr, "user")) {
|
|
user = value;
|
|
} else if (!strcmp(ptr, "salt")) {
|
|
salt = value;
|
|
} else if (!strcmp(ptr, "opaque")) {
|
|
opaque = value;
|
|
} else if (!strcmp(ptr, "challenge")) {
|
|
challenge = value;
|
|
} else if (!strcmp(ptr, "nonce")) {
|
|
nonce = value;
|
|
} else {
|
|
av_log(s, AV_LOG_INFO, "Ignoring unsupported var %s\n", ptr);
|
|
}
|
|
} else {
|
|
av_log(s, AV_LOG_WARNING, "Variable %s has NULL value\n", ptr);
|
|
}
|
|
ptr = next;
|
|
}
|
|
|
|
if (!strcmp(authmod, "adobe")) {
|
|
if ((ret = do_adobe_auth(rt, user, salt, opaque, challenge)) < 0)
|
|
return ret;
|
|
} else {
|
|
if ((ret = do_llnw_auth(rt, user, nonce)) < 0)
|
|
return ret;
|
|
}
|
|
|
|
rt->auth_tried = 1;
|
|
return 0;
|
|
}
|
|
|
|
static int handle_invoke_error(URLContext *s, RTMPPacket *pkt)
|
|
{
|
|
RTMPContext *rt = s->priv_data;
|
|
const uint8_t *data_end = pkt->data + pkt->size;
|
|
char *tracked_method = NULL;
|
|
int level = AV_LOG_ERROR;
|
|
uint8_t tmpstr[256];
|
|
int ret;
|
|
|
|
if ((ret = find_tracked_method(s, pkt, 9, &tracked_method)) < 0)
|
|
return ret;
|
|
|
|
if (!ff_amf_get_field_value(pkt->data + 9, data_end,
|
|
"description", tmpstr, sizeof(tmpstr))) {
|
|
if (tracked_method && (!strcmp(tracked_method, "_checkbw") ||
|
|
!strcmp(tracked_method, "releaseStream") ||
|
|
!strcmp(tracked_method, "FCSubscribe") ||
|
|
!strcmp(tracked_method, "FCPublish"))) {
|
|
/* Gracefully ignore Adobe-specific historical artifact errors. */
|
|
level = AV_LOG_WARNING;
|
|
ret = 0;
|
|
} else if (tracked_method && !strcmp(tracked_method, "getStreamLength")) {
|
|
level = rt->live ? AV_LOG_DEBUG : AV_LOG_WARNING;
|
|
ret = 0;
|
|
} else if (tracked_method && !strcmp(tracked_method, "connect")) {
|
|
ret = handle_connect_error(s, tmpstr);
|
|
if (!ret) {
|
|
rt->do_reconnect = 1;
|
|
level = AV_LOG_VERBOSE;
|
|
}
|
|
} else
|
|
ret = AVERROR_UNKNOWN;
|
|
av_log(s, level, "Server error: %s\n", tmpstr);
|
|
}
|
|
|
|
av_free(tracked_method);
|
|
return ret;
|
|
}
|
|
|
|
static int write_begin(URLContext *s)
|
|
{
|
|
RTMPContext *rt = s->priv_data;
|
|
PutByteContext pbc;
|
|
RTMPPacket spkt = { 0 };
|
|
int ret;
|
|
|
|
// Send Stream Begin 1
|
|
if ((ret = ff_rtmp_packet_create(&spkt, RTMP_NETWORK_CHANNEL,
|
|
RTMP_PT_USER_CONTROL, 0, 6)) < 0) {
|
|
av_log(s, AV_LOG_ERROR, "Unable to create response packet\n");
|
|
return ret;
|
|
}
|
|
|
|
bytestream2_init_writer(&pbc, spkt.data, spkt.size);
|
|
bytestream2_put_be16(&pbc, 0); // 0 -> Stream Begin
|
|
bytestream2_put_be32(&pbc, rt->nb_streamid);
|
|
|
|
ret = ff_rtmp_packet_write(rt->stream, &spkt, rt->out_chunk_size,
|
|
&rt->prev_pkt[1], &rt->nb_prev_pkt[1]);
|
|
|
|
ff_rtmp_packet_destroy(&spkt);
|
|
|
|
return ret;
|
|
}
|
|
|
|
static int write_status(URLContext *s, RTMPPacket *pkt,
|
|
const char *status, const char *filename)
|
|
{
|
|
RTMPContext *rt = s->priv_data;
|
|
RTMPPacket spkt = { 0 };
|
|
char statusmsg[128];
|
|
uint8_t *pp;
|
|
int ret;
|
|
|
|
if ((ret = ff_rtmp_packet_create(&spkt, RTMP_SYSTEM_CHANNEL,
|
|
RTMP_PT_INVOKE, 0,
|
|
RTMP_PKTDATA_DEFAULT_SIZE)) < 0) {
|
|
av_log(s, AV_LOG_ERROR, "Unable to create response packet\n");
|
|
return ret;
|
|
}
|
|
|
|
pp = spkt.data;
|
|
spkt.extra = pkt->extra;
|
|
ff_amf_write_string(&pp, "onStatus");
|
|
ff_amf_write_number(&pp, 0);
|
|
ff_amf_write_null(&pp);
|
|
|
|
ff_amf_write_object_start(&pp);
|
|
ff_amf_write_field_name(&pp, "level");
|
|
ff_amf_write_string(&pp, "status");
|
|
ff_amf_write_field_name(&pp, "code");
|
|
ff_amf_write_string(&pp, status);
|
|
ff_amf_write_field_name(&pp, "description");
|
|
snprintf(statusmsg, sizeof(statusmsg),
|
|
"%s is now published", filename);
|
|
ff_amf_write_string(&pp, statusmsg);
|
|
ff_amf_write_field_name(&pp, "details");
|
|
ff_amf_write_string(&pp, filename);
|
|
ff_amf_write_object_end(&pp);
|
|
|
|
spkt.size = pp - spkt.data;
|
|
ret = ff_rtmp_packet_write(rt->stream, &spkt, rt->out_chunk_size,
|
|
&rt->prev_pkt[1], &rt->nb_prev_pkt[1]);
|
|
ff_rtmp_packet_destroy(&spkt);
|
|
|
|
return ret;
|
|
}
|
|
|
|
static int send_invoke_response(URLContext *s, RTMPPacket *pkt)
|
|
{
|
|
RTMPContext *rt = s->priv_data;
|
|
double seqnum;
|
|
char filename[128];
|
|
char command[64];
|
|
int stringlen;
|
|
char *pchar;
|
|
const uint8_t *p = pkt->data;
|
|
uint8_t *pp = NULL;
|
|
RTMPPacket spkt = { 0 };
|
|
GetByteContext gbc;
|
|
int ret;
|
|
|
|
bytestream2_init(&gbc, p, pkt->size);
|
|
if (ff_amf_read_string(&gbc, command, sizeof(command),
|
|
&stringlen)) {
|
|
av_log(s, AV_LOG_ERROR, "Error in PT_INVOKE\n");
|
|
return AVERROR_INVALIDDATA;
|
|
}
|
|
|
|
ret = ff_amf_read_number(&gbc, &seqnum);
|
|
if (ret)
|
|
return ret;
|
|
ret = ff_amf_read_null(&gbc);
|
|
if (ret)
|
|
return ret;
|
|
if (!strcmp(command, "FCPublish") ||
|
|
!strcmp(command, "publish")) {
|
|
ret = ff_amf_read_string(&gbc, filename,
|
|
sizeof(filename), &stringlen);
|
|
if (ret) {
|
|
if (ret == AVERROR(EINVAL))
|
|
av_log(s, AV_LOG_ERROR, "Unable to parse stream name - name too long?\n");
|
|
else
|
|
av_log(s, AV_LOG_ERROR, "Unable to parse stream name\n");
|
|
return ret;
|
|
}
|
|
// check with url
|
|
if (s->filename) {
|
|
pchar = strrchr(s->filename, '/');
|
|
if (!pchar) {
|
|
av_log(s, AV_LOG_WARNING,
|
|
"Unable to find / in url %s, bad format\n",
|
|
s->filename);
|
|
pchar = s->filename;
|
|
}
|
|
pchar++;
|
|
if (strcmp(pchar, filename))
|
|
av_log(s, AV_LOG_WARNING, "Unexpected stream %s, expecting"
|
|
" %s\n", filename, pchar);
|
|
}
|
|
rt->state = STATE_RECEIVING;
|
|
}
|
|
|
|
if (!strcmp(command, "FCPublish")) {
|
|
if ((ret = ff_rtmp_packet_create(&spkt, RTMP_SYSTEM_CHANNEL,
|
|
RTMP_PT_INVOKE, 0,
|
|
RTMP_PKTDATA_DEFAULT_SIZE)) < 0) {
|
|
av_log(s, AV_LOG_ERROR, "Unable to create response packet\n");
|
|
return ret;
|
|
}
|
|
pp = spkt.data;
|
|
ff_amf_write_string(&pp, "onFCPublish");
|
|
} else if (!strcmp(command, "publish")) {
|
|
ret = write_begin(s);
|
|
if (ret < 0)
|
|
return ret;
|
|
|
|
// Send onStatus(NetStream.Publish.Start)
|
|
return write_status(s, pkt, "NetStream.Publish.Start",
|
|
filename);
|
|
} else if (!strcmp(command, "play")) {
|
|
ret = write_begin(s);
|
|
if (ret < 0)
|
|
return ret;
|
|
rt->state = STATE_SENDING;
|
|
return write_status(s, pkt, "NetStream.Play.Start",
|
|
filename);
|
|
} else {
|
|
if ((ret = ff_rtmp_packet_create(&spkt, RTMP_SYSTEM_CHANNEL,
|
|
RTMP_PT_INVOKE, 0,
|
|
RTMP_PKTDATA_DEFAULT_SIZE)) < 0) {
|
|
av_log(s, AV_LOG_ERROR, "Unable to create response packet\n");
|
|
return ret;
|
|
}
|
|
pp = spkt.data;
|
|
ff_amf_write_string(&pp, "_result");
|
|
ff_amf_write_number(&pp, seqnum);
|
|
ff_amf_write_null(&pp);
|
|
if (!strcmp(command, "createStream")) {
|
|
rt->nb_streamid++;
|
|
if (rt->nb_streamid == 0 || rt->nb_streamid == 2)
|
|
rt->nb_streamid++; /* Values 0 and 2 are reserved */
|
|
ff_amf_write_number(&pp, rt->nb_streamid);
|
|
/* By now we don't control which streams are removed in
|
|
* deleteStream. There is no stream creation control
|
|
* if a client creates more than 2^32 - 2 streams. */
|
|
}
|
|
}
|
|
spkt.size = pp - spkt.data;
|
|
ret = ff_rtmp_packet_write(rt->stream, &spkt, rt->out_chunk_size,
|
|
&rt->prev_pkt[1], &rt->nb_prev_pkt[1]);
|
|
ff_rtmp_packet_destroy(&spkt);
|
|
return ret;
|
|
}
|
|
|
|
/**
|
|
* Read the AMF_NUMBER response ("_result") to a function call
|
|
* (e.g. createStream()). This response should be made up of the AMF_STRING
|
|
* "result", a NULL object and then the response encoded as AMF_NUMBER. On a
|
|
* successful response, we will return set the value to number (otherwise number
|
|
* will not be changed).
|
|
*
|
|
* @return 0 if reading the value succeeds, negative value otherwise
|
|
*/
|
|
static int read_number_result(RTMPPacket *pkt, double *number)
|
|
{
|
|
// We only need to fit "_result" in this.
|
|
uint8_t strbuffer[8];
|
|
int stringlen;
|
|
double numbuffer;
|
|
GetByteContext gbc;
|
|
|
|
bytestream2_init(&gbc, pkt->data, pkt->size);
|
|
|
|
// Value 1/4: "_result" as AMF_STRING
|
|
if (ff_amf_read_string(&gbc, strbuffer, sizeof(strbuffer), &stringlen))
|
|
return AVERROR_INVALIDDATA;
|
|
if (strcmp(strbuffer, "_result"))
|
|
return AVERROR_INVALIDDATA;
|
|
// Value 2/4: The callee reference number
|
|
if (ff_amf_read_number(&gbc, &numbuffer))
|
|
return AVERROR_INVALIDDATA;
|
|
// Value 3/4: Null
|
|
if (ff_amf_read_null(&gbc))
|
|
return AVERROR_INVALIDDATA;
|
|
// Value 4/4: The response as AMF_NUMBER
|
|
if (ff_amf_read_number(&gbc, &numbuffer))
|
|
return AVERROR_INVALIDDATA;
|
|
else
|
|
*number = numbuffer;
|
|
|
|
return 0;
|
|
}
|
|
|
|
static int handle_invoke_result(URLContext *s, RTMPPacket *pkt)
|
|
{
|
|
RTMPContext *rt = s->priv_data;
|
|
char *tracked_method = NULL;
|
|
int ret = 0;
|
|
|
|
if ((ret = find_tracked_method(s, pkt, 10, &tracked_method)) < 0)
|
|
return ret;
|
|
|
|
if (!tracked_method) {
|
|
/* Ignore this reply when the current method is not tracked. */
|
|
return ret;
|
|
}
|
|
|
|
if (!strcmp(tracked_method, "connect")) {
|
|
if (!rt->is_input) {
|
|
if ((ret = gen_release_stream(s, rt)) < 0)
|
|
goto fail;
|
|
|
|
if ((ret = gen_fcpublish_stream(s, rt)) < 0)
|
|
goto fail;
|
|
} else {
|
|
if ((ret = gen_window_ack_size(s, rt)) < 0)
|
|
goto fail;
|
|
}
|
|
|
|
if ((ret = gen_create_stream(s, rt)) < 0)
|
|
goto fail;
|
|
|
|
if (rt->is_input) {
|
|
/* Send the FCSubscribe command when the name of live
|
|
* stream is defined by the user or if it's a live stream. */
|
|
if (rt->subscribe) {
|
|
if ((ret = gen_fcsubscribe_stream(s, rt, rt->subscribe)) < 0)
|
|
goto fail;
|
|
} else if (rt->live == -1) {
|
|
if ((ret = gen_fcsubscribe_stream(s, rt, rt->playpath)) < 0)
|
|
goto fail;
|
|
}
|
|
}
|
|
} else if (!strcmp(tracked_method, "createStream")) {
|
|
double stream_id;
|
|
if (read_number_result(pkt, &stream_id)) {
|
|
av_log(s, AV_LOG_WARNING, "Unexpected reply on connect()\n");
|
|
} else {
|
|
rt->stream_id = stream_id;
|
|
}
|
|
|
|
if (!rt->is_input) {
|
|
if ((ret = gen_publish(s, rt)) < 0)
|
|
goto fail;
|
|
} else {
|
|
if (rt->live != -1) {
|
|
if ((ret = gen_get_stream_length(s, rt)) < 0)
|
|
goto fail;
|
|
}
|
|
if ((ret = gen_play(s, rt)) < 0)
|
|
goto fail;
|
|
if ((ret = gen_buffer_time(s, rt)) < 0)
|
|
goto fail;
|
|
}
|
|
} else if (!strcmp(tracked_method, "getStreamLength")) {
|
|
if (read_number_result(pkt, &rt->duration)) {
|
|
av_log(s, AV_LOG_WARNING, "Unexpected reply on getStreamLength()\n");
|
|
}
|
|
}
|
|
|
|
fail:
|
|
av_free(tracked_method);
|
|
return ret;
|
|
}
|
|
|
|
static int handle_invoke_status(URLContext *s, RTMPPacket *pkt)
|
|
{
|
|
RTMPContext *rt = s->priv_data;
|
|
const uint8_t *data_end = pkt->data + pkt->size;
|
|
const uint8_t *ptr = pkt->data + RTMP_HEADER;
|
|
uint8_t tmpstr[256];
|
|
int i, t;
|
|
|
|
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")) {
|
|
t = ff_amf_get_field_value(ptr, data_end,
|
|
"description", tmpstr, sizeof(tmpstr));
|
|
if (t || !tmpstr[0])
|
|
t = ff_amf_get_field_value(ptr, data_end, "code",
|
|
tmpstr, sizeof(tmpstr));
|
|
if (!t)
|
|
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;
|
|
if (!t && !strcmp(tmpstr, "NetStream.Seek.Notify")) rt->state = STATE_PLAYING;
|
|
|
|
return 0;
|
|
}
|
|
|
|
static int handle_invoke(URLContext *s, RTMPPacket *pkt)
|
|
{
|
|
RTMPContext *rt = s->priv_data;
|
|
int ret = 0;
|
|
|
|
//TODO: check for the messages sent for wrong state?
|
|
if (ff_amf_match_string(pkt->data, pkt->size, "_error")) {
|
|
if ((ret = handle_invoke_error(s, pkt)) < 0)
|
|
return ret;
|
|
} else if (ff_amf_match_string(pkt->data, pkt->size, "_result")) {
|
|
if ((ret = handle_invoke_result(s, pkt)) < 0)
|
|
return ret;
|
|
} else if (ff_amf_match_string(pkt->data, pkt->size, "onStatus")) {
|
|
if ((ret = handle_invoke_status(s, pkt)) < 0)
|
|
return ret;
|
|
} else if (ff_amf_match_string(pkt->data, pkt->size, "onBWDone")) {
|
|
if ((ret = gen_check_bw(s, rt)) < 0)
|
|
return ret;
|
|
} else if (ff_amf_match_string(pkt->data, pkt->size, "releaseStream") ||
|
|
ff_amf_match_string(pkt->data, pkt->size, "FCPublish") ||
|
|
ff_amf_match_string(pkt->data, pkt->size, "publish") ||
|
|
ff_amf_match_string(pkt->data, pkt->size, "play") ||
|
|
ff_amf_match_string(pkt->data, pkt->size, "_checkbw") ||
|
|
ff_amf_match_string(pkt->data, pkt->size, "createStream")) {
|
|
if ((ret = send_invoke_response(s, pkt)) < 0)
|
|
return ret;
|
|
}
|
|
|
|
return ret;
|
|
}
|
|
|
|
static int update_offset(RTMPContext *rt, int size)
|
|
{
|
|
int old_flv_size;
|
|
|
|
// generate packet header and put data into buffer for FLV demuxer
|
|
if (rt->flv_off < rt->flv_size) {
|
|
// There is old unread data in the buffer, thus append at the end
|
|
old_flv_size = rt->flv_size;
|
|
rt->flv_size += size;
|
|
} else {
|
|
// All data has been read, write the new data at the start of the buffer
|
|
old_flv_size = 0;
|
|
rt->flv_size = size;
|
|
rt->flv_off = 0;
|
|
}
|
|
|
|
return old_flv_size;
|
|
}
|
|
|
|
static int append_flv_data(RTMPContext *rt, RTMPPacket *pkt, int skip)
|
|
{
|
|
int old_flv_size, ret;
|
|
PutByteContext pbc;
|
|
const uint8_t *data = pkt->data + skip;
|
|
const int size = pkt->size - skip;
|
|
uint32_t ts = pkt->timestamp;
|
|
|
|
if (pkt->type == RTMP_PT_AUDIO) {
|
|
rt->has_audio = 1;
|
|
} else if (pkt->type == RTMP_PT_VIDEO) {
|
|
rt->has_video = 1;
|
|
}
|
|
|
|
old_flv_size = update_offset(rt, size + 15);
|
|
|
|
if ((ret = av_reallocp(&rt->flv_data, rt->flv_size)) < 0) {
|
|
rt->flv_size = rt->flv_off = 0;
|
|
return ret;
|
|
}
|
|
bytestream2_init_writer(&pbc, rt->flv_data, rt->flv_size);
|
|
bytestream2_skip_p(&pbc, old_flv_size);
|
|
bytestream2_put_byte(&pbc, pkt->type);
|
|
bytestream2_put_be24(&pbc, size);
|
|
bytestream2_put_be24(&pbc, ts);
|
|
bytestream2_put_byte(&pbc, ts >> 24);
|
|
bytestream2_put_be24(&pbc, 0);
|
|
bytestream2_put_buffer(&pbc, data, size);
|
|
bytestream2_put_be32(&pbc, size + RTMP_HEADER);
|
|
|
|
return 0;
|
|
}
|
|
|
|
static int handle_notify(URLContext *s, RTMPPacket *pkt)
|
|
{
|
|
RTMPContext *rt = s->priv_data;
|
|
uint8_t commandbuffer[64];
|
|
char statusmsg[128];
|
|
int stringlen, ret, skip = 0;
|
|
GetByteContext gbc;
|
|
|
|
bytestream2_init(&gbc, pkt->data, pkt->size);
|
|
if (ff_amf_read_string(&gbc, commandbuffer, sizeof(commandbuffer),
|
|
&stringlen))
|
|
return AVERROR_INVALIDDATA;
|
|
|
|
if (!strcmp(commandbuffer, "onMetaData")) {
|
|
// metadata properties should be stored in a mixed array
|
|
if (bytestream2_get_byte(&gbc) == AMF_DATA_TYPE_MIXEDARRAY) {
|
|
// We have found a metaData Array so flv can determine the streams
|
|
// from this.
|
|
rt->received_metadata = 1;
|
|
// skip 32-bit max array index
|
|
bytestream2_skip(&gbc, 4);
|
|
while (bytestream2_get_bytes_left(&gbc) > 3) {
|
|
if (ff_amf_get_string(&gbc, statusmsg, sizeof(statusmsg),
|
|
&stringlen))
|
|
return AVERROR_INVALIDDATA;
|
|
// We do not care about the content of the property (yet).
|
|
stringlen = ff_amf_tag_size(gbc.buffer, gbc.buffer_end);
|
|
if (stringlen < 0)
|
|
return AVERROR_INVALIDDATA;
|
|
bytestream2_skip(&gbc, stringlen);
|
|
|
|
// The presence of the following properties indicates that the
|
|
// respective streams are present.
|
|
if (!strcmp(statusmsg, "videocodecid")) {
|
|
rt->has_video = 1;
|
|
}
|
|
if (!strcmp(statusmsg, "audiocodecid")) {
|
|
rt->has_audio = 1;
|
|
}
|
|
}
|
|
if (bytestream2_get_be24(&gbc) != AMF_END_OF_OBJECT)
|
|
return AVERROR_INVALIDDATA;
|
|
}
|
|
}
|
|
|
|
// Skip the @setDataFrame string and validate it is a notification
|
|
if (!strcmp(commandbuffer, "@setDataFrame")) {
|
|
skip = gbc.buffer - pkt->data;
|
|
ret = ff_amf_read_string(&gbc, statusmsg,
|
|
sizeof(statusmsg), &stringlen);
|
|
if (ret < 0)
|
|
return AVERROR_INVALIDDATA;
|
|
}
|
|
|
|
return append_flv_data(rt, pkt, skip);
|
|
}
|
|
|
|
/**
|
|
* 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_BYTES_READ:
|
|
av_log(s, AV_LOG_TRACE, "received bytes read report\n");
|
|
break;
|
|
case RTMP_PT_CHUNK_SIZE:
|
|
if ((ret = handle_chunk_size(s, pkt)) < 0)
|
|
return ret;
|
|
break;
|
|
case RTMP_PT_USER_CONTROL:
|
|
if ((ret = handle_user_control(s, pkt)) < 0)
|
|
return ret;
|
|
break;
|
|
case RTMP_PT_SET_PEER_BW:
|
|
if ((ret = handle_set_peer_bw(s, pkt)) < 0)
|
|
return ret;
|
|
break;
|
|
case RTMP_PT_WINDOW_ACK_SIZE:
|
|
if ((ret = handle_window_ack_size(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:
|
|
case RTMP_PT_METADATA:
|
|
case RTMP_PT_NOTIFY:
|
|
/* Audio, Video and Metadata 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;
|
|
}
|
|
|
|
static int handle_metadata(RTMPContext *rt, RTMPPacket *pkt)
|
|
{
|
|
int ret, old_flv_size, type;
|
|
const uint8_t *next;
|
|
uint8_t *p;
|
|
uint32_t size;
|
|
uint32_t ts, cts, pts = 0;
|
|
|
|
old_flv_size = update_offset(rt, pkt->size);
|
|
|
|
if ((ret = av_reallocp(&rt->flv_data, rt->flv_size)) < 0) {
|
|
rt->flv_size = rt->flv_off = 0;
|
|
return ret;
|
|
}
|
|
|
|
next = pkt->data;
|
|
p = rt->flv_data + old_flv_size;
|
|
|
|
/* copy data while rewriting timestamps */
|
|
ts = pkt->timestamp;
|
|
|
|
while (next - pkt->data < pkt->size - RTMP_HEADER) {
|
|
type = bytestream_get_byte(&next);
|
|
size = bytestream_get_be24(&next);
|
|
cts = bytestream_get_be24(&next);
|
|
cts |= bytestream_get_byte(&next) << 24;
|
|
if (!pts)
|
|
pts = cts;
|
|
ts += cts - pts;
|
|
pts = cts;
|
|
if (size + 3 + 4 > pkt->data + pkt->size - next)
|
|
break;
|
|
bytestream_put_byte(&p, type);
|
|
bytestream_put_be24(&p, size);
|
|
bytestream_put_be24(&p, ts);
|
|
bytestream_put_byte(&p, ts >> 24);
|
|
memcpy(p, next, size + 3 + 4);
|
|
p += size + 3;
|
|
bytestream_put_be32(&p, size + RTMP_HEADER);
|
|
next += size + 3 + 4;
|
|
}
|
|
if (p != rt->flv_data + rt->flv_size) {
|
|
av_log(NULL, AV_LOG_WARNING, "Incomplete flv packets in "
|
|
"RTMP_PT_METADATA packet\n");
|
|
rt->flv_size = p - rt->flv_data;
|
|
}
|
|
|
|
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;
|
|
|
|
if (rt->state == STATE_STOPPED)
|
|
return AVERROR_EOF;
|
|
|
|
for (;;) {
|
|
RTMPPacket rpkt = { 0 };
|
|
if ((ret = ff_rtmp_packet_read(rt->stream, &rpkt,
|
|
rt->in_chunk_size, &rt->prev_pkt[0],
|
|
&rt->nb_prev_pkt[0])) <= 0) {
|
|
if (ret == 0) {
|
|
return AVERROR(EAGAIN);
|
|
} else {
|
|
return AVERROR(EIO);
|
|
}
|
|
}
|
|
|
|
// Track timestamp for later use
|
|
rt->last_timestamp = rpkt.timestamp;
|
|
|
|
rt->bytes_read += ret;
|
|
if (rt->bytes_read - rt->last_bytes_read > rt->receive_report_size) {
|
|
av_log(s, AV_LOG_DEBUG, "Sending bytes read report\n");
|
|
if ((ret = gen_bytes_read(s, rt, rpkt.timestamp + 1)) < 0) {
|
|
ff_rtmp_packet_destroy(&rpkt);
|
|
return ret;
|
|
}
|
|
rt->last_bytes_read = rt->bytes_read;
|
|
}
|
|
|
|
ret = rtmp_parse_result(s, rt, &rpkt);
|
|
|
|
// At this point we must check if we are in the seek state and continue
|
|
// with the next packet. handle_invoke will get us out of this state
|
|
// when the right message is encountered
|
|
if (rt->state == STATE_SEEKING) {
|
|
ff_rtmp_packet_destroy(&rpkt);
|
|
// We continue, let the natural flow of things happen:
|
|
// AVERROR(EAGAIN) or handle_invoke gets us out of here
|
|
continue;
|
|
}
|
|
|
|
if (ret < 0) {//serious error in current packet
|
|
ff_rtmp_packet_destroy(&rpkt);
|
|
return ret;
|
|
}
|
|
if (rt->do_reconnect && for_header) {
|
|
ff_rtmp_packet_destroy(&rpkt);
|
|
return 0;
|
|
}
|
|
if (rt->state == STATE_STOPPED) {
|
|
ff_rtmp_packet_destroy(&rpkt);
|
|
return AVERROR_EOF;
|
|
}
|
|
if (for_header && (rt->state == STATE_PLAYING ||
|
|
rt->state == STATE_PUBLISHING ||
|
|
rt->state == STATE_SENDING ||
|
|
rt->state == STATE_RECEIVING)) {
|
|
ff_rtmp_packet_destroy(&rpkt);
|
|
return 0;
|
|
}
|
|
if (!rpkt.size || !rt->is_input) {
|
|
ff_rtmp_packet_destroy(&rpkt);
|
|
continue;
|
|
}
|
|
if (rpkt.type == RTMP_PT_VIDEO || rpkt.type == RTMP_PT_AUDIO) {
|
|
ret = append_flv_data(rt, &rpkt, 0);
|
|
ff_rtmp_packet_destroy(&rpkt);
|
|
return ret;
|
|
} else if (rpkt.type == RTMP_PT_NOTIFY) {
|
|
ret = handle_notify(s, &rpkt);
|
|
ff_rtmp_packet_destroy(&rpkt);
|
|
return ret;
|
|
} else if (rpkt.type == RTMP_PT_METADATA) {
|
|
ret = handle_metadata(rt, &rpkt);
|
|
ff_rtmp_packet_destroy(&rpkt);
|
|
return ret;
|
|
}
|
|
ff_rtmp_packet_destroy(&rpkt);
|
|
}
|
|
}
|
|
|
|
static int rtmp_close(URLContext *h)
|
|
{
|
|
RTMPContext *rt = h->priv_data;
|
|
int ret = 0, i, j;
|
|
|
|
if (!rt->is_input) {
|
|
rt->flv_data = NULL;
|
|
if (rt->out_pkt.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);
|
|
for (i = 0; i < 2; i++) {
|
|
for (j = 0; j < rt->nb_prev_pkt[i]; j++)
|
|
ff_rtmp_packet_destroy(&rt->prev_pkt[i][j]);
|
|
av_freep(&rt->prev_pkt[i]);
|
|
}
|
|
|
|
free_tracked_methods(rt);
|
|
av_freep(&rt->flv_data);
|
|
ffurl_close(rt->stream);
|
|
return ret;
|
|
}
|
|
|
|
/**
|
|
* Insert a fake onMetadata packet into the FLV stream to notify the FLV
|
|
* demuxer about the duration of the stream.
|
|
*
|
|
* This should only be done if there was no real onMetadata packet sent by the
|
|
* server at the start of the stream and if we were able to retrieve a valid
|
|
* duration via a getStreamLength call.
|
|
*
|
|
* @return 0 for successful operation, negative value in case of error
|
|
*/
|
|
static int inject_fake_duration_metadata(RTMPContext *rt)
|
|
{
|
|
// We need to insert the metadata packet directly after the FLV
|
|
// header, i.e. we need to move all other already read data by the
|
|
// size of our fake metadata packet.
|
|
|
|
uint8_t* p;
|
|
// Keep old flv_data pointer
|
|
uint8_t* old_flv_data = rt->flv_data;
|
|
// Allocate a new flv_data pointer with enough space for the additional package
|
|
if (!(rt->flv_data = av_malloc(rt->flv_size + 55))) {
|
|
rt->flv_data = old_flv_data;
|
|
return AVERROR(ENOMEM);
|
|
}
|
|
|
|
// Copy FLV header
|
|
memcpy(rt->flv_data, old_flv_data, 13);
|
|
// Copy remaining packets
|
|
memcpy(rt->flv_data + 13 + 55, old_flv_data + 13, rt->flv_size - 13);
|
|
// Increase the size by the injected packet
|
|
rt->flv_size += 55;
|
|
// Delete the old FLV data
|
|
av_freep(&old_flv_data);
|
|
|
|
p = rt->flv_data + 13;
|
|
bytestream_put_byte(&p, FLV_TAG_TYPE_META);
|
|
bytestream_put_be24(&p, 40); // size of data part (sum of all parts below)
|
|
bytestream_put_be24(&p, 0); // timestamp
|
|
bytestream_put_be32(&p, 0); // reserved
|
|
|
|
// first event name as a string
|
|
bytestream_put_byte(&p, AMF_DATA_TYPE_STRING);
|
|
// "onMetaData" as AMF string
|
|
bytestream_put_be16(&p, 10);
|
|
bytestream_put_buffer(&p, "onMetaData", 10);
|
|
|
|
// mixed array (hash) with size and string/type/data tuples
|
|
bytestream_put_byte(&p, AMF_DATA_TYPE_MIXEDARRAY);
|
|
bytestream_put_be32(&p, 1); // metadata_count
|
|
|
|
// "duration" as AMF string
|
|
bytestream_put_be16(&p, 8);
|
|
bytestream_put_buffer(&p, "duration", 8);
|
|
bytestream_put_byte(&p, AMF_DATA_TYPE_NUMBER);
|
|
bytestream_put_be64(&p, av_double2int(rt->duration));
|
|
|
|
// Finalise object
|
|
bytestream_put_be16(&p, 0); // Empty string
|
|
bytestream_put_byte(&p, AMF_END_OF_OBJECT);
|
|
bytestream_put_be32(&p, 40 + RTMP_HEADER); // size of data part (sum of all parts above)
|
|
|
|
return 0;
|
|
}
|
|
|
|
/**
|
|
* Open RTMP connection and verify that the stream can be played.
|
|
*
|
|
* URL syntax: rtmp://server[:port][/app][/playpath]
|
|
* where 'app' is first one or two directories in the path
|
|
* (e.g. /ondemand/, /flash/live/, etc.)
|
|
* and 'playpath' is a file name (the rest of the path,
|
|
* may be prefixed with "mp4:")
|
|
*/
|
|
static int rtmp_open(URLContext *s, const char *uri, int flags, AVDictionary **opts)
|
|
{
|
|
RTMPContext *rt = s->priv_data;
|
|
char proto[8], hostname[256], path[1024], auth[100], *fname;
|
|
char *old_app, *qmark, *n, fname_buffer[1024];
|
|
uint8_t buf[2048];
|
|
int port;
|
|
int ret;
|
|
|
|
if (rt->listen_timeout > 0)
|
|
rt->listen = 1;
|
|
|
|
rt->is_input = !(flags & AVIO_FLAG_WRITE);
|
|
|
|
av_url_split(proto, sizeof(proto), auth, sizeof(auth),
|
|
hostname, sizeof(hostname), &port,
|
|
path, sizeof(path), s->filename);
|
|
|
|
n = strchr(path, ' ');
|
|
if (n) {
|
|
av_log(s, AV_LOG_WARNING,
|
|
"Detected librtmp style URL parameters, these aren't supported "
|
|
"by the libavformat internal RTMP handler currently enabled. "
|
|
"See the documentation for the correct way to pass parameters.\n");
|
|
*n = '\0'; // Trim not supported part
|
|
}
|
|
|
|
if (auth[0]) {
|
|
char *ptr = strchr(auth, ':');
|
|
if (ptr) {
|
|
*ptr = '\0';
|
|
av_strlcpy(rt->username, auth, sizeof(rt->username));
|
|
av_strlcpy(rt->password, ptr + 1, sizeof(rt->password));
|
|
}
|
|
}
|
|
|
|
if (rt->listen && strcmp(proto, "rtmp")) {
|
|
av_log(s, AV_LOG_ERROR, "rtmp_listen not available for %s\n",
|
|
proto);
|
|
return AVERROR(EINVAL);
|
|
}
|
|
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;
|
|
if (rt->listen)
|
|
ff_url_join(buf, sizeof(buf), "tcp", NULL, hostname, port,
|
|
"?listen&listen_timeout=%d",
|
|
rt->listen_timeout * 1000);
|
|
else
|
|
ff_url_join(buf, sizeof(buf), "tcp", NULL, hostname, port, NULL);
|
|
}
|
|
|
|
reconnect:
|
|
if ((ret = ffurl_open_whitelist(&rt->stream, buf, AVIO_FLAG_READ_WRITE,
|
|
&s->interrupt_callback, opts,
|
|
s->protocol_whitelist, s->protocol_blacklist, s)) < 0) {
|
|
av_log(s , AV_LOG_ERROR, "Cannot open connection %s\n", buf);
|
|
goto fail;
|
|
}
|
|
|
|
if (rt->swfverify) {
|
|
if ((ret = rtmp_calc_swfhash(s)) < 0)
|
|
goto fail;
|
|
}
|
|
|
|
rt->state = STATE_START;
|
|
if (!rt->listen && (ret = rtmp_handshake(s, rt)) < 0)
|
|
goto fail;
|
|
if (rt->listen && (ret = rtmp_server_handshake(s, rt)) < 0)
|
|
goto fail;
|
|
|
|
rt->out_chunk_size = 128;
|
|
rt->in_chunk_size = 128; // Probably overwritten later
|
|
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
|
|
qmark = strchr(path, '?');
|
|
if (qmark && strstr(qmark, "slist=")) {
|
|
char* amp;
|
|
// After slist we have the playpath, the full path is used as app
|
|
av_strlcpy(rt->app, path + 1, APP_MAX_LENGTH);
|
|
fname = strstr(path, "slist=") + 6;
|
|
// Strip any further query parameters from fname
|
|
amp = strchr(fname, '&');
|
|
if (amp) {
|
|
av_strlcpy(fname_buffer, fname, FFMIN(amp - fname + 1,
|
|
sizeof(fname_buffer)));
|
|
fname = fname_buffer;
|
|
}
|
|
} else 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) {
|
|
if (old_app) {
|
|
// If name of application has been defined by the user, assume that
|
|
// playpath is provided in the URL
|
|
fname = next;
|
|
} else {
|
|
fname = NULL;
|
|
av_strlcpy(rt->app, next, APP_MAX_LENGTH);
|
|
}
|
|
} 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, FFMIN(p - path, APP_MAX_LENGTH));
|
|
} else {
|
|
fname++;
|
|
av_strlcpy(rt->app, path + 1, FFMIN(fname - path - 1, APP_MAX_LENGTH));
|
|
}
|
|
}
|
|
}
|
|
|
|
if (old_app) {
|
|
// The name of application has been defined by the user, override it.
|
|
if (strlen(old_app) >= APP_MAX_LENGTH) {
|
|
ret = AVERROR(EINVAL);
|
|
goto fail;
|
|
}
|
|
av_free(rt->app);
|
|
rt->app = old_app;
|
|
}
|
|
|
|
if (!rt->playpath) {
|
|
rt->playpath = av_malloc(PLAYPATH_MAX_LENGTH);
|
|
if (!rt->playpath) {
|
|
ret = AVERROR(ENOMEM);
|
|
goto fail;
|
|
}
|
|
|
|
if (fname) {
|
|
int len = strlen(fname);
|
|
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';
|
|
rt->playpath[0] = 0;
|
|
}
|
|
av_strlcat(rt->playpath, fname, PLAYPATH_MAX_LENGTH);
|
|
} else {
|
|
rt->playpath[0] = '\0';
|
|
}
|
|
}
|
|
|
|
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->receive_report_size = 1048576;
|
|
rt->bytes_read = 0;
|
|
rt->has_audio = 0;
|
|
rt->has_video = 0;
|
|
rt->received_metadata = 0;
|
|
rt->last_bytes_read = 0;
|
|
rt->max_sent_unacked = 2500000;
|
|
rt->duration = 0;
|
|
|
|
av_log(s, AV_LOG_DEBUG, "Proto = %s, path = %s, app = %s, fname = %s\n",
|
|
proto, path, rt->app, rt->playpath);
|
|
if (!rt->listen) {
|
|
if ((ret = gen_connect(s, rt)) < 0)
|
|
goto fail;
|
|
} else {
|
|
if ((ret = read_connect(s, s->priv_data)) < 0)
|
|
goto fail;
|
|
}
|
|
|
|
do {
|
|
ret = get_packet(s, 1);
|
|
} while (ret == AVERROR(EAGAIN));
|
|
if (ret < 0)
|
|
goto fail;
|
|
|
|
if (rt->do_reconnect) {
|
|
int i;
|
|
ffurl_close(rt->stream);
|
|
rt->stream = NULL;
|
|
rt->do_reconnect = 0;
|
|
rt->nb_invokes = 0;
|
|
for (i = 0; i < 2; i++)
|
|
memset(rt->prev_pkt[i], 0,
|
|
sizeof(**rt->prev_pkt) * rt->nb_prev_pkt[i]);
|
|
free_tracked_methods(rt);
|
|
goto reconnect;
|
|
}
|
|
|
|
if (rt->is_input) {
|
|
// generate FLV header for demuxer
|
|
rt->flv_size = 13;
|
|
if ((ret = av_reallocp(&rt->flv_data, rt->flv_size)) < 0)
|
|
goto fail;
|
|
rt->flv_off = 0;
|
|
memcpy(rt->flv_data, "FLV\1\0\0\0\0\011\0\0\0\0", rt->flv_size);
|
|
|
|
// Read packets until we reach the first A/V packet or read metadata.
|
|
// If there was a metadata package in front of the A/V packets, we can
|
|
// build the FLV header from this. If we do not receive any metadata,
|
|
// the FLV decoder will allocate the needed streams when their first
|
|
// audio or video packet arrives.
|
|
while (!rt->has_audio && !rt->has_video && !rt->received_metadata) {
|
|
if ((ret = get_packet(s, 0)) < 0)
|
|
goto fail;
|
|
}
|
|
|
|
// Either after we have read the metadata or (if there is none) the
|
|
// first packet of an A/V stream, we have a better knowledge about the
|
|
// streams, so set the FLV header accordingly.
|
|
if (rt->has_audio) {
|
|
rt->flv_data[4] |= FLV_HEADER_FLAG_HASAUDIO;
|
|
}
|
|
if (rt->has_video) {
|
|
rt->flv_data[4] |= FLV_HEADER_FLAG_HASVIDEO;
|
|
}
|
|
|
|
// If we received the first packet of an A/V stream and no metadata but
|
|
// the server returned a valid duration, create a fake metadata packet
|
|
// to inform the FLV decoder about the duration.
|
|
if (!rt->received_metadata && rt->duration > 0) {
|
|
if ((ret = inject_fake_duration_metadata(rt)) < 0)
|
|
goto fail;
|
|
}
|
|
} 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 int64_t rtmp_seek(URLContext *s, int stream_index, int64_t timestamp,
|
|
int flags)
|
|
{
|
|
RTMPContext *rt = s->priv_data;
|
|
int ret;
|
|
av_log(s, AV_LOG_DEBUG,
|
|
"Seek on stream index %d at timestamp %"PRId64" with flags %08x\n",
|
|
stream_index, timestamp, flags);
|
|
if ((ret = gen_seek(s, rt, timestamp)) < 0) {
|
|
av_log(s, AV_LOG_ERROR,
|
|
"Unable to send seek command on stream index %d at timestamp "
|
|
"%"PRId64" with flags %08x\n",
|
|
stream_index, timestamp, flags);
|
|
return ret;
|
|
}
|
|
rt->flv_off = rt->flv_size;
|
|
rt->state = STATE_SEEKING;
|
|
return timestamp;
|
|
}
|
|
|
|
static int rtmp_pause(URLContext *s, int pause)
|
|
{
|
|
RTMPContext *rt = s->priv_data;
|
|
int ret;
|
|
av_log(s, AV_LOG_DEBUG, "Pause at timestamp %d\n",
|
|
rt->last_timestamp);
|
|
if ((ret = gen_pause(s, rt, pause, rt->last_timestamp)) < 0) {
|
|
av_log(s, AV_LOG_ERROR, "Unable to send pause command at timestamp %d\n",
|
|
rt->last_timestamp);
|
|
return ret;
|
|
}
|
|
return 0;
|
|
}
|
|
|
|
static int rtmp_write(URLContext *s, const uint8_t *buf, int size)
|
|
{
|
|
RTMPContext *rt = s->priv_data;
|
|
int size_temp = size;
|
|
int pktsize, pkttype, copy;
|
|
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 < RTMP_HEADER) {
|
|
const uint8_t *header = rt->flv_header;
|
|
int channel = RTMP_AUDIO_CHANNEL;
|
|
|
|
copy = FFMIN(RTMP_HEADER - 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 < RTMP_HEADER)
|
|
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;
|
|
|
|
if (pkttype == RTMP_PT_VIDEO)
|
|
channel = RTMP_VIDEO_CHANNEL;
|
|
|
|
if (((pkttype == RTMP_PT_VIDEO || pkttype == RTMP_PT_AUDIO) && ts == 0) ||
|
|
pkttype == RTMP_PT_NOTIFY) {
|
|
if ((ret = ff_rtmp_check_alloc_array(&rt->prev_pkt[1],
|
|
&rt->nb_prev_pkt[1],
|
|
channel)) < 0)
|
|
return ret;
|
|
// Force sending a full 12 bytes header by clearing the
|
|
// channel id, to make it not match a potential earlier
|
|
// packet in the same channel.
|
|
rt->prev_pkt[1][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, channel,
|
|
pkttype, ts, pktsize)) < 0)
|
|
return ret;
|
|
|
|
rt->out_pkt.extra = rt->stream_id;
|
|
rt->flv_data = rt->out_pkt.data;
|
|
}
|
|
|
|
copy = FFMIN(rt->flv_size - rt->flv_off, size_temp);
|
|
bytestream_get_buffer(&buf_temp, rt->flv_data + rt->flv_off, copy);
|
|
rt->flv_off += copy;
|
|
size_temp -= copy;
|
|
|
|
if (rt->flv_off == rt->flv_size) {
|
|
rt->skip_bytes = 4;
|
|
|
|
if (rt->out_pkt.type == RTMP_PT_NOTIFY) {
|
|
// For onMetaData and |RtmpSampleAccess packets, we want
|
|
// @setDataFrame prepended to the packet before it gets sent.
|
|
// However, not all RTMP_PT_NOTIFY packets (e.g., onTextData
|
|
// and onCuePoint).
|
|
uint8_t commandbuffer[64];
|
|
int stringlen = 0;
|
|
GetByteContext gbc;
|
|
|
|
bytestream2_init(&gbc, rt->flv_data, rt->flv_size);
|
|
if (!ff_amf_read_string(&gbc, commandbuffer, sizeof(commandbuffer),
|
|
&stringlen)) {
|
|
if (!strcmp(commandbuffer, "onMetaData") ||
|
|
!strcmp(commandbuffer, "|RtmpSampleAccess")) {
|
|
uint8_t *ptr;
|
|
if ((ret = av_reallocp(&rt->out_pkt.data, rt->out_pkt.size + 16)) < 0) {
|
|
rt->flv_size = rt->flv_off = rt->flv_header_bytes = 0;
|
|
return ret;
|
|
}
|
|
memmove(rt->out_pkt.data + 16, rt->out_pkt.data, rt->out_pkt.size);
|
|
rt->out_pkt.size += 16;
|
|
ptr = rt->out_pkt.data;
|
|
ff_amf_write_string(&ptr, "@setDataFrame");
|
|
}
|
|
}
|
|
}
|
|
|
|
if ((ret = rtmp_send_packet(rt, &rt->out_pkt, 0)) < 0)
|
|
return ret;
|
|
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->in_chunk_size,
|
|
&rt->prev_pkt[0],
|
|
&rt->nb_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, {.i64 = 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, {.i64 = 10}, 0, INT_MAX, ENC},
|
|
{"rtmp_live", "Specify that the media is a live stream.", OFFSET(live), AV_OPT_TYPE_INT, {.i64 = -2}, INT_MIN, INT_MAX, DEC, "rtmp_live"},
|
|
{"any", "both", 0, AV_OPT_TYPE_CONST, {.i64 = -2}, 0, 0, DEC, "rtmp_live"},
|
|
{"live", "live stream", 0, AV_OPT_TYPE_CONST, {.i64 = -1}, 0, 0, DEC, "rtmp_live"},
|
|
{"recorded", "recorded stream", 0, AV_OPT_TYPE_CONST, {.i64 = 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_subscribe", "Name of live stream to subscribe to. Defaults to rtmp_playpath.", OFFSET(subscribe), AV_OPT_TYPE_STRING, {.str = NULL }, 0, 0, DEC},
|
|
{"rtmp_swfhash", "SHA256 hash of the decompressed SWF file (32 bytes).", OFFSET(swfhash), AV_OPT_TYPE_BINARY, .flags = DEC},
|
|
{"rtmp_swfsize", "Size of the decompressed SWF file, required for SWFVerification.", OFFSET(swfsize), AV_OPT_TYPE_INT, {.i64 = 0}, 0, INT_MAX, DEC},
|
|
{"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_swfverify", "URL to player swf file, compute hash/size automatically.", OFFSET(swfverify), AV_OPT_TYPE_STRING, {.str = NULL }, 0, 0, DEC},
|
|
{"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},
|
|
{"rtmp_listen", "Listen for incoming rtmp connections", OFFSET(listen), AV_OPT_TYPE_INT, {.i64 = 0}, INT_MIN, INT_MAX, DEC, "rtmp_listen" },
|
|
{"listen", "Listen for incoming rtmp connections", OFFSET(listen), AV_OPT_TYPE_INT, {.i64 = 0}, INT_MIN, INT_MAX, DEC, "rtmp_listen" },
|
|
{"timeout", "Maximum timeout (in seconds) to wait for incoming connections. -1 is infinite. Implies -rtmp_listen 1", OFFSET(listen_timeout), AV_OPT_TYPE_INT, {.i64 = -1}, INT_MIN, INT_MAX, DEC, "rtmp_listen" },
|
|
{ NULL },
|
|
};
|
|
|
|
#define RTMP_PROTOCOL(flavor) \
|
|
static const AVClass flavor##_class = { \
|
|
.class_name = #flavor, \
|
|
.item_name = av_default_item_name, \
|
|
.option = rtmp_options, \
|
|
.version = LIBAVUTIL_VERSION_INT, \
|
|
}; \
|
|
\
|
|
const URLProtocol ff_##flavor##_protocol = { \
|
|
.name = #flavor, \
|
|
.url_open2 = rtmp_open, \
|
|
.url_read = rtmp_read, \
|
|
.url_read_seek = rtmp_seek, \
|
|
.url_read_pause = rtmp_pause, \
|
|
.url_write = rtmp_write, \
|
|
.url_close = rtmp_close, \
|
|
.priv_data_size = sizeof(RTMPContext), \
|
|
.flags = URL_PROTOCOL_FLAG_NETWORK, \
|
|
.priv_data_class= &flavor##_class, \
|
|
};
|
|
|
|
|
|
RTMP_PROTOCOL(rtmp)
|
|
RTMP_PROTOCOL(rtmpe)
|
|
RTMP_PROTOCOL(rtmps)
|
|
RTMP_PROTOCOL(rtmpt)
|
|
RTMP_PROTOCOL(rtmpte)
|
|
RTMP_PROTOCOL(rtmpts)
|