The _checkbw calls were changed to use transactionId 0 in commit 82613564 so that servers would not return _result/_error about it. While this is the strict interpretation of the spec, there are servers that return _error about it, even if transactionId was 0. The latest version of EvoStream Media Server (the commercial version of crtmpserver) behaves properly as described, i.e. returning an _error normally but not returning anything when using transactionId 0. The latest version of crtmpserver (right now at least) doesn't behave like this though, it returns an error even if transactionId was 0. There are also other servers that return errors even if transactionId is set to 0. Therefore set a proper transaction id so that the invoke can be tracked and the error properly ignored instead. Signed-off-by: Martin Storsjö <martin@martin.st>
1739 lines
56 KiB
C
1739 lines
56 KiB
C
/*
|
|
* RTMP network protocol
|
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* Copyright (c) 2009 Kostya Shishkov
|
|
*
|
|
* This file is part of Libav.
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*
|
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* Libav is free software; you can redistribute it and/or
|
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* modify it under the terms of the GNU Lesser General Public
|
|
* 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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|
*
|
|
* Libav is distributed in the hope that it will be useful,
|
|
* but WITHOUT ANY WARRANTY; without even the implied warranty of
|
|
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
|
|
* Lesser General Public License for more details.
|
|
*
|
|
* You should have received a copy of the GNU Lesser General Public
|
|
* License along with Libav; 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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/**
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* @file
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* RTMP protocol
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*/
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|
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#include "libavcodec/bytestream.h"
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#include "libavutil/avstring.h"
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#include "libavutil/intfloat.h"
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#include "libavutil/lfg.h"
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#include "libavutil/opt.h"
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#include "libavutil/sha.h"
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#include "avformat.h"
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#include "internal.h"
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#include "network.h"
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#include "flv.h"
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#include "rtmp.h"
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#include "rtmpcrypt.h"
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#include "rtmppkt.h"
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#include "url.h"
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//#define DEBUG
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#define APP_MAX_LENGTH 128
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#define PLAYPATH_MAX_LENGTH 256
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#define TCURL_MAX_LENGTH 512
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#define FLASHVER_MAX_LENGTH 64
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|
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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_PUBLISHING, ///< client has started sending multimedia data to server (for output)
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STATE_STOPPED, ///< the broadcast has been stopped
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} ClientState;
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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][RTMP_CHANNELS]; ///< packet history used when reading and sending packets
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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 main_channel_id; ///< an additional channel ID which is used for some invocations
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uint8_t* flv_data; ///< buffer with data for demuxer
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int flv_size; ///< current buffer size
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int flv_off; ///< number of bytes read from current buffer
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int flv_nb_packets; ///< number of flv packets published
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RTMPPacket out_pkt; ///< rtmp packet, created from flv a/v or metadata (for output)
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uint32_t client_report_size; ///< number of bytes after which client should report to server
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uint32_t bytes_read; ///< number of bytes read from server
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uint32_t last_bytes_read; ///< number of bytes read last reported to server
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int skip_bytes; ///< number of bytes to skip from the input FLV stream in the next write call
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uint8_t flv_header[11]; ///< partial incoming flv packet header
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int flv_header_bytes; ///< number of initialized bytes in flv_header
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int nb_invokes; ///< keeps track of invoke messages
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char* tcurl; ///< url of the target stream
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char* flashver; ///< version of the flash plugin
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char* swfurl; ///< url of the swf player
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char* pageurl; ///< url of the web page
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char* subscribe; ///< name of live stream to subscribe
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int server_bw; ///< server bandwidth
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int client_buffer_time; ///< client buffer time in ms
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int flush_interval; ///< number of packets flushed in the same request (RTMPT only)
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int encrypted; ///< use an encrypted connection (RTMPE only)
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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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} RTMPContext;
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|
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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 add_tracked_method(RTMPContext *rt, const char *name, int id)
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{
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void *ptr;
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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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ptr = av_realloc(rt->tracked_methods,
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rt->tracked_methods_size * sizeof(*rt->tracked_methods));
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if (!ptr)
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return AVERROR(ENOMEM);
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rt->tracked_methods = ptr;
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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->data_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_free(rt->tracked_methods[i].name);
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av_free(rt->tracked_methods);
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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->data_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]);
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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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if (!field || !value)
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goto fail;
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ff_amf_write_field_name(p, field);
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} else {
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goto fail;
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}
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switch (type) {
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case 'B':
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ff_amf_write_bool(p, value[0] != '0');
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break;
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case 'S':
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ff_amf_write_string(p, value);
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break;
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case 'N':
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ff_amf_write_number(p, strtod(value, NULL));
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break;
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case 'Z':
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ff_amf_write_null(p);
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break;
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case 'O':
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if (value[0] != '0')
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ff_amf_write_object_start(p);
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else
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ff_amf_write_object_end(p);
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break;
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default:
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goto fail;
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break;
|
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}
|
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return 0;
|
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fail:
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av_log(s, AV_LOG_ERROR, "Invalid AMF parameter: %s\n", param);
|
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return AVERROR(EINVAL);
|
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}
|
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|
|
/**
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|
* Generate 'connect' call and send it to the server.
|
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*/
|
|
static int gen_connect(URLContext *s, RTMPContext *rt)
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|
{
|
|
RTMPPacket pkt;
|
|
uint8_t *p;
|
|
int ret;
|
|
|
|
if ((ret = ff_rtmp_packet_create(&pkt, RTMP_SYSTEM_CHANNEL, RTMP_PT_INVOKE,
|
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0, 4096)) < 0)
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return ret;
|
|
|
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p = pkt.data;
|
|
|
|
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");
|
|
ff_amf_write_string(&p, rt->app);
|
|
|
|
if (!rt->is_input) {
|
|
ff_amf_write_field_name(&p, "type");
|
|
ff_amf_write_string(&p, "nonprivate");
|
|
}
|
|
ff_amf_write_field_name(&p, "flashVer");
|
|
ff_amf_write_string(&p, rt->flashver);
|
|
|
|
if (rt->swfurl) {
|
|
ff_amf_write_field_name(&p, "swfUrl");
|
|
ff_amf_write_string(&p, rt->swfurl);
|
|
}
|
|
|
|
ff_amf_write_field_name(&p, "tcUrl");
|
|
ff_amf_write_string(&p, rt->tcurl);
|
|
if (rt->is_input) {
|
|
ff_amf_write_field_name(&p, "fpad");
|
|
ff_amf_write_bool(&p, 0);
|
|
ff_amf_write_field_name(&p, "capabilities");
|
|
ff_amf_write_number(&p, 15.0);
|
|
|
|
/* Tell the server we support all the audio codecs except
|
|
* SUPPORT_SND_INTEL (0x0008) and SUPPORT_SND_UNUSED (0x0010)
|
|
* which are unused in the RTMP protocol implementation. */
|
|
ff_amf_write_field_name(&p, "audioCodecs");
|
|
ff_amf_write_number(&p, 4071.0);
|
|
ff_amf_write_field_name(&p, "videoCodecs");
|
|
ff_amf_write_number(&p, 252.0);
|
|
ff_amf_write_field_name(&p, "videoFunction");
|
|
ff_amf_write_number(&p, 1.0);
|
|
|
|
if (rt->pageurl) {
|
|
ff_amf_write_field_name(&p, "pageUrl");
|
|
ff_amf_write_string(&p, rt->pageurl);
|
|
}
|
|
}
|
|
ff_amf_write_object_end(&p);
|
|
|
|
if (rt->conn) {
|
|
char *param = rt->conn;
|
|
|
|
// Write arbitrary AMF data to the Connect message.
|
|
while (param != NULL) {
|
|
char *sep;
|
|
param += strspn(param, " ");
|
|
if (!*param)
|
|
break;
|
|
sep = strchr(param, ' ');
|
|
if (sep)
|
|
*sep = '\0';
|
|
if ((ret = rtmp_write_amf_data(s, param, &p)) < 0) {
|
|
// Invalid AMF parameter.
|
|
ff_rtmp_packet_destroy(&pkt);
|
|
return ret;
|
|
}
|
|
|
|
if (sep)
|
|
param = sep + 1;
|
|
else
|
|
break;
|
|
}
|
|
}
|
|
|
|
pkt.data_size = p - pkt.data;
|
|
|
|
return rtmp_send_packet(rt, &pkt, 1);
|
|
}
|
|
|
|
/**
|
|
* 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, 0);
|
|
}
|
|
|
|
/**
|
|
* Generate 'FCPublish' call and send it to the server. It should make
|
|
* the server preapare for receiving media streams.
|
|
*/
|
|
static int gen_fcpublish_stream(URLContext *s, RTMPContext *rt)
|
|
{
|
|
RTMPPacket pkt;
|
|
uint8_t *p;
|
|
int ret;
|
|
|
|
if ((ret = ff_rtmp_packet_create(&pkt, RTMP_SYSTEM_CHANNEL, RTMP_PT_INVOKE,
|
|
0, 25 + strlen(rt->playpath))) < 0)
|
|
return ret;
|
|
|
|
av_log(s, AV_LOG_DEBUG, "FCPublish stream...\n");
|
|
p = pkt.data;
|
|
ff_amf_write_string(&p, "FCPublish");
|
|
ff_amf_write_number(&p, ++rt->nb_invokes);
|
|
ff_amf_write_null(&p);
|
|
ff_amf_write_string(&p, rt->playpath);
|
|
|
|
return rtmp_send_packet(rt, &pkt, 0);
|
|
}
|
|
|
|
/**
|
|
* 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->main_channel_id);
|
|
|
|
return rtmp_send_packet(rt, &pkt, 0);
|
|
}
|
|
|
|
/**
|
|
* Generate client buffer time and send it to the server.
|
|
*/
|
|
static int gen_buffer_time(URLContext *s, RTMPContext *rt)
|
|
{
|
|
RTMPPacket pkt;
|
|
uint8_t *p;
|
|
int ret;
|
|
|
|
if ((ret = ff_rtmp_packet_create(&pkt, RTMP_NETWORK_CHANNEL, RTMP_PT_PING,
|
|
1, 10)) < 0)
|
|
return ret;
|
|
|
|
p = pkt.data;
|
|
bytestream_put_be16(&p, 3);
|
|
bytestream_put_be32(&p, rt->main_channel_id);
|
|
bytestream_put_be32(&p, rt->client_buffer_time);
|
|
|
|
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_VIDEO_CHANNEL, RTMP_PT_INVOKE,
|
|
0, 29 + strlen(rt->playpath))) < 0)
|
|
return ret;
|
|
|
|
pkt.extra = rt->main_channel_id;
|
|
|
|
p = pkt.data;
|
|
ff_amf_write_string(&p, "play");
|
|
ff_amf_write_number(&p, ++rt->nb_invokes);
|
|
ff_amf_write_null(&p);
|
|
ff_amf_write_string(&p, rt->playpath);
|
|
ff_amf_write_number(&p, rt->live);
|
|
|
|
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->main_channel_id;
|
|
|
|
p = pkt.data;
|
|
ff_amf_write_string(&p, "publish");
|
|
ff_amf_write_number(&p, ++rt->nb_invokes);
|
|
ff_amf_write_null(&p);
|
|
ff_amf_write_string(&p, rt->playpath);
|
|
ff_amf_write_string(&p, "live");
|
|
|
|
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->data_size < 6) {
|
|
av_log(s, AV_LOG_ERROR, "Too short ping packet (%d)\n",
|
|
ppkt->data_size);
|
|
return AVERROR_INVALIDDATA;
|
|
}
|
|
|
|
if ((ret = ff_rtmp_packet_create(&pkt, RTMP_NETWORK_CHANNEL, RTMP_PT_PING,
|
|
ppkt->timestamp + 1, 6)) < 0)
|
|
return ret;
|
|
|
|
p = pkt.data;
|
|
bytestream_put_be16(&p, 7);
|
|
bytestream_put_be32(&p, AV_RB32(ppkt->data+2));
|
|
|
|
return rtmp_send_packet(rt, &pkt, 0);
|
|
}
|
|
|
|
/**
|
|
* Generate server bandwidth message and send it to the server.
|
|
*/
|
|
static int gen_server_bw(URLContext *s, RTMPContext *rt)
|
|
{
|
|
RTMPPacket pkt;
|
|
uint8_t *p;
|
|
int ret;
|
|
|
|
if ((ret = ff_rtmp_packet_create(&pkt, RTMP_NETWORK_CHANNEL, RTMP_PT_SERVER_BW,
|
|
0, 4)) < 0)
|
|
return ret;
|
|
|
|
p = pkt.data;
|
|
bytestream_put_be32(&p, rt->server_bw);
|
|
|
|
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, 0);
|
|
}
|
|
|
|
/**
|
|
* 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);
|
|
}
|
|
|
|
int ff_rtmp_calc_digest(const uint8_t *src, int len, int gap,
|
|
const uint8_t *key, int keylen, uint8_t *dst)
|
|
{
|
|
struct AVSHA *sha;
|
|
uint8_t hmac_buf[64+32] = {0};
|
|
int i;
|
|
|
|
sha = av_mallocz(av_sha_size);
|
|
if (!sha)
|
|
return AVERROR(ENOMEM);
|
|
|
|
if (keylen < 64) {
|
|
memcpy(hmac_buf, key, keylen);
|
|
} else {
|
|
av_sha_init(sha, 256);
|
|
av_sha_update(sha,key, keylen);
|
|
av_sha_final(sha, hmac_buf);
|
|
}
|
|
for (i = 0; i < 64; i++)
|
|
hmac_buf[i] ^= HMAC_IPAD_VAL;
|
|
|
|
av_sha_init(sha, 256);
|
|
av_sha_update(sha, hmac_buf, 64);
|
|
if (gap <= 0) {
|
|
av_sha_update(sha, src, len);
|
|
} else { //skip 32 bytes used for storing digest
|
|
av_sha_update(sha, src, gap);
|
|
av_sha_update(sha, src + gap + 32, len - gap - 32);
|
|
}
|
|
av_sha_final(sha, hmac_buf + 64);
|
|
|
|
for (i = 0; i < 64; i++)
|
|
hmac_buf[i] ^= HMAC_IPAD_VAL ^ HMAC_OPAD_VAL; //reuse XORed key for opad
|
|
av_sha_init(sha, 256);
|
|
av_sha_update(sha, hmac_buf, 64+32);
|
|
av_sha_final(sha, dst);
|
|
|
|
av_free(sha);
|
|
|
|
return 0;
|
|
}
|
|
|
|
int ff_rtmp_calc_digest_pos(const uint8_t *buf, int off, int mod_val,
|
|
int add_val)
|
|
{
|
|
int i, digest_pos = 0;
|
|
|
|
for (i = 0; i < 4; i++)
|
|
digest_pos += buf[i + off];
|
|
digest_pos = digest_pos % mod_val + add_val;
|
|
|
|
return digest_pos;
|
|
}
|
|
|
|
/**
|
|
* Put HMAC-SHA2 digest of packet data (except for the bytes where this digest
|
|
* will be stored) into that packet.
|
|
*
|
|
* @param buf handshake data (1536 bytes)
|
|
* @param encrypted use an encrypted connection (RTMPE)
|
|
* @return offset to the digest inside input data
|
|
*/
|
|
static int rtmp_handshake_imprint_with_digest(uint8_t *buf, int encrypted)
|
|
{
|
|
int ret, digest_pos;
|
|
|
|
if (encrypted)
|
|
digest_pos = ff_rtmp_calc_digest_pos(buf, 772, 728, 776);
|
|
else
|
|
digest_pos = ff_rtmp_calc_digest_pos(buf, 8, 728, 12);
|
|
|
|
ret = ff_rtmp_calc_digest(buf, RTMP_HANDSHAKE_PACKET_SIZE, digest_pos,
|
|
rtmp_player_key, PLAYER_KEY_OPEN_PART_LEN,
|
|
buf + digest_pos);
|
|
if (ret < 0)
|
|
return ret;
|
|
|
|
return digest_pos;
|
|
}
|
|
|
|
/**
|
|
* Verify that the received server response has the expected digest value.
|
|
*
|
|
* @param buf handshake data received from the server (1536 bytes)
|
|
* @param off position to search digest offset from
|
|
* @return 0 if digest is valid, digest position otherwise
|
|
*/
|
|
static int rtmp_validate_digest(uint8_t *buf, int off)
|
|
{
|
|
uint8_t digest[32];
|
|
int ret, digest_pos;
|
|
|
|
digest_pos = ff_rtmp_calc_digest_pos(buf, off, 728, off + 4);
|
|
|
|
ret = ff_rtmp_calc_digest(buf, RTMP_HANDSHAKE_PACKET_SIZE, digest_pos,
|
|
rtmp_server_key, SERVER_KEY_OPEN_PART_LEN,
|
|
digest);
|
|
if (ret < 0)
|
|
return ret;
|
|
|
|
if (!memcmp(digest, buf + digest_pos, 32))
|
|
return digest_pos;
|
|
return 0;
|
|
}
|
|
|
|
/**
|
|
* Perform handshake with the server by means of exchanging pseudorandom data
|
|
* signed with HMAC-SHA2 digest.
|
|
*
|
|
* @return 0 if handshake succeeds, negative value otherwise
|
|
*/
|
|
static int rtmp_handshake(URLContext *s, RTMPContext *rt)
|
|
{
|
|
AVLFG rnd;
|
|
uint8_t tosend [RTMP_HANDSHAKE_PACKET_SIZE+1] = {
|
|
3, // unencrypted data
|
|
0, 0, 0, 0, // client uptime
|
|
RTMP_CLIENT_VER1,
|
|
RTMP_CLIENT_VER2,
|
|
RTMP_CLIENT_VER3,
|
|
RTMP_CLIENT_VER4,
|
|
};
|
|
uint8_t clientdata[RTMP_HANDSHAKE_PACKET_SIZE];
|
|
uint8_t serverdata[RTMP_HANDSHAKE_PACKET_SIZE+1];
|
|
int i;
|
|
int server_pos, client_pos;
|
|
uint8_t digest[32], signature[32];
|
|
int ret, type = 0;
|
|
|
|
av_log(s, AV_LOG_DEBUG, "Handshaking...\n");
|
|
|
|
av_lfg_init(&rnd, 0xDEADC0DE);
|
|
// generate handshake packet - 1536 bytes of pseudorandom data
|
|
for (i = 9; i <= RTMP_HANDSHAKE_PACKET_SIZE; i++)
|
|
tosend[i] = av_lfg_get(&rnd) >> 24;
|
|
|
|
if (rt->encrypted && CONFIG_FFRTMPCRYPT_PROTOCOL) {
|
|
/* When the client wants to use RTMPE, we have to change the command
|
|
* byte to 0x06 which means to use encrypted data and we have to set
|
|
* the flash version to at least 9.0.115.0. */
|
|
tosend[0] = 6;
|
|
tosend[5] = 128;
|
|
tosend[6] = 0;
|
|
tosend[7] = 3;
|
|
tosend[8] = 2;
|
|
|
|
/* Initialize the Diffie-Hellmann context and generate the public key
|
|
* to send to the server. */
|
|
if ((ret = ff_rtmpe_gen_pub_key(rt->stream, tosend + 1)) < 0)
|
|
return ret;
|
|
}
|
|
|
|
client_pos = rtmp_handshake_imprint_with_digest(tosend + 1, rt->encrypted);
|
|
if (client_pos < 0)
|
|
return client_pos;
|
|
|
|
if ((ret = ffurl_write(rt->stream, tosend,
|
|
RTMP_HANDSHAKE_PACKET_SIZE + 1)) < 0) {
|
|
av_log(s, AV_LOG_ERROR, "Cannot write RTMP handshake request\n");
|
|
return ret;
|
|
}
|
|
|
|
if ((ret = ffurl_read_complete(rt->stream, serverdata,
|
|
RTMP_HANDSHAKE_PACKET_SIZE + 1)) < 0) {
|
|
av_log(s, AV_LOG_ERROR, "Cannot read RTMP handshake response\n");
|
|
return ret;
|
|
}
|
|
|
|
if ((ret = ffurl_read_complete(rt->stream, clientdata,
|
|
RTMP_HANDSHAKE_PACKET_SIZE)) < 0) {
|
|
av_log(s, AV_LOG_ERROR, "Cannot read RTMP handshake response\n");
|
|
return ret;
|
|
}
|
|
|
|
av_log(s, AV_LOG_DEBUG, "Type answer %d\n", serverdata[0]);
|
|
av_log(s, AV_LOG_DEBUG, "Server version %d.%d.%d.%d\n",
|
|
serverdata[5], serverdata[6], serverdata[7], serverdata[8]);
|
|
|
|
if (rt->is_input && serverdata[5] >= 3) {
|
|
server_pos = rtmp_validate_digest(serverdata + 1, 772);
|
|
if (server_pos < 0)
|
|
return server_pos;
|
|
|
|
if (!server_pos) {
|
|
type = 1;
|
|
server_pos = rtmp_validate_digest(serverdata + 1, 8);
|
|
if (server_pos < 0)
|
|
return server_pos;
|
|
|
|
if (!server_pos) {
|
|
av_log(s, AV_LOG_ERROR, "Server response validating failed\n");
|
|
return AVERROR(EIO);
|
|
}
|
|
}
|
|
|
|
ret = ff_rtmp_calc_digest(tosend + 1 + client_pos, 32, 0,
|
|
rtmp_server_key, sizeof(rtmp_server_key),
|
|
digest);
|
|
if (ret < 0)
|
|
return ret;
|
|
|
|
ret = ff_rtmp_calc_digest(clientdata, RTMP_HANDSHAKE_PACKET_SIZE - 32,
|
|
0, digest, 32, signature);
|
|
if (ret < 0)
|
|
return ret;
|
|
|
|
if (rt->encrypted && CONFIG_FFRTMPCRYPT_PROTOCOL) {
|
|
/* Compute the shared secret key sent by the server and initialize
|
|
* the RC4 encryption. */
|
|
if ((ret = ff_rtmpe_compute_secret_key(rt->stream, serverdata + 1,
|
|
tosend + 1, type)) < 0)
|
|
return ret;
|
|
|
|
/* Encrypt the signature received by the server. */
|
|
ff_rtmpe_encrypt_sig(rt->stream, signature, digest, serverdata[0]);
|
|
}
|
|
|
|
if (memcmp(signature, clientdata + RTMP_HANDSHAKE_PACKET_SIZE - 32, 32)) {
|
|
av_log(s, AV_LOG_ERROR, "Signature mismatch\n");
|
|
return AVERROR(EIO);
|
|
}
|
|
|
|
for (i = 0; i < RTMP_HANDSHAKE_PACKET_SIZE; i++)
|
|
tosend[i] = av_lfg_get(&rnd) >> 24;
|
|
ret = ff_rtmp_calc_digest(serverdata + 1 + server_pos, 32, 0,
|
|
rtmp_player_key, sizeof(rtmp_player_key),
|
|
digest);
|
|
if (ret < 0)
|
|
return ret;
|
|
|
|
ret = ff_rtmp_calc_digest(tosend, RTMP_HANDSHAKE_PACKET_SIZE - 32, 0,
|
|
digest, 32,
|
|
tosend + RTMP_HANDSHAKE_PACKET_SIZE - 32);
|
|
if (ret < 0)
|
|
return ret;
|
|
|
|
if (rt->encrypted && CONFIG_FFRTMPCRYPT_PROTOCOL) {
|
|
/* Encrypt the signature to be send to the server. */
|
|
ff_rtmpe_encrypt_sig(rt->stream, tosend +
|
|
RTMP_HANDSHAKE_PACKET_SIZE - 32, digest,
|
|
serverdata[0]);
|
|
}
|
|
|
|
// write reply back to the server
|
|
if ((ret = ffurl_write(rt->stream, tosend,
|
|
RTMP_HANDSHAKE_PACKET_SIZE)) < 0)
|
|
return ret;
|
|
|
|
if (rt->encrypted && CONFIG_FFRTMPCRYPT_PROTOCOL) {
|
|
/* Set RC4 keys for encryption and update the keystreams. */
|
|
if ((ret = ff_rtmpe_update_keystream(rt->stream)) < 0)
|
|
return ret;
|
|
}
|
|
} else {
|
|
if (rt->encrypted && CONFIG_FFRTMPCRYPT_PROTOCOL) {
|
|
/* Compute the shared secret key sent by the server and initialize
|
|
* the RC4 encryption. */
|
|
if ((ret = ff_rtmpe_compute_secret_key(rt->stream, serverdata + 1,
|
|
tosend + 1, 1)) < 0)
|
|
return ret;
|
|
|
|
if (serverdata[0] == 9) {
|
|
/* Encrypt the signature received by the server. */
|
|
ff_rtmpe_encrypt_sig(rt->stream, signature, digest,
|
|
serverdata[0]);
|
|
}
|
|
}
|
|
|
|
if ((ret = ffurl_write(rt->stream, serverdata + 1,
|
|
RTMP_HANDSHAKE_PACKET_SIZE)) < 0)
|
|
return ret;
|
|
|
|
if (rt->encrypted && CONFIG_FFRTMPCRYPT_PROTOCOL) {
|
|
/* Set RC4 keys for encryption and update the keystreams. */
|
|
if ((ret = ff_rtmpe_update_keystream(rt->stream)) < 0)
|
|
return ret;
|
|
}
|
|
}
|
|
|
|
return 0;
|
|
}
|
|
|
|
static int handle_chunk_size(URLContext *s, RTMPPacket *pkt)
|
|
{
|
|
RTMPContext *rt = s->priv_data;
|
|
int ret;
|
|
|
|
if (pkt->data_size < 4) {
|
|
av_log(s, AV_LOG_ERROR,
|
|
"Too short chunk size change packet (%d)\n",
|
|
pkt->data_size);
|
|
return AVERROR_INVALIDDATA;
|
|
}
|
|
|
|
if (!rt->is_input) {
|
|
/* 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])) < 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_ping(URLContext *s, RTMPPacket *pkt)
|
|
{
|
|
RTMPContext *rt = s->priv_data;
|
|
int t, ret;
|
|
|
|
if (pkt->data_size < 2) {
|
|
av_log(s, AV_LOG_ERROR, "Too short ping packet (%d)\n",
|
|
pkt->data_size);
|
|
return AVERROR_INVALIDDATA;
|
|
}
|
|
|
|
t = AV_RB16(pkt->data);
|
|
if (t == 6) {
|
|
if ((ret = gen_pong(s, rt, pkt)) < 0)
|
|
return ret;
|
|
}
|
|
|
|
return 0;
|
|
}
|
|
|
|
static int handle_client_bw(URLContext *s, RTMPPacket *pkt)
|
|
{
|
|
RTMPContext *rt = s->priv_data;
|
|
|
|
if (pkt->data_size < 4) {
|
|
av_log(s, AV_LOG_ERROR,
|
|
"Client bandwidth report packet is less than 4 bytes long (%d)\n",
|
|
pkt->data_size);
|
|
return AVERROR_INVALIDDATA;
|
|
}
|
|
|
|
rt->client_report_size = AV_RB32(pkt->data);
|
|
if (rt->client_report_size <= 0) {
|
|
av_log(s, AV_LOG_ERROR, "Incorrect client bandwidth %d\n",
|
|
rt->client_report_size);
|
|
return AVERROR_INVALIDDATA;
|
|
|
|
}
|
|
av_log(s, AV_LOG_DEBUG, "Client bandwidth = %d\n", rt->client_report_size);
|
|
rt->client_report_size >>= 1;
|
|
|
|
return 0;
|
|
}
|
|
|
|
static int handle_server_bw(URLContext *s, RTMPPacket *pkt)
|
|
{
|
|
RTMPContext *rt = s->priv_data;
|
|
|
|
if (pkt->data_size < 4) {
|
|
av_log(s, AV_LOG_ERROR,
|
|
"Too short server bandwidth report packet (%d)\n",
|
|
pkt->data_size);
|
|
return AVERROR_INVALIDDATA;
|
|
}
|
|
|
|
rt->server_bw = AV_RB32(pkt->data);
|
|
if (rt->server_bw <= 0) {
|
|
av_log(s, AV_LOG_ERROR, "Incorrect server bandwidth %d\n",
|
|
rt->server_bw);
|
|
return AVERROR_INVALIDDATA;
|
|
}
|
|
av_log(s, AV_LOG_DEBUG, "Server bandwidth = %d\n", rt->server_bw);
|
|
|
|
return 0;
|
|
}
|
|
|
|
static int handle_invoke_error(URLContext *s, RTMPPacket *pkt)
|
|
{
|
|
const uint8_t *data_end = pkt->data + pkt->data_size;
|
|
uint8_t tmpstr[256];
|
|
|
|
if (!ff_amf_get_field_value(pkt->data + 9, data_end,
|
|
"description", tmpstr, sizeof(tmpstr))) {
|
|
av_log(s, AV_LOG_ERROR, "Server error: %s\n", tmpstr);
|
|
return -1;
|
|
}
|
|
|
|
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 (!memcmp(tracked_method, "connect", 7)) {
|
|
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_server_bw(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 (!memcmp(tracked_method, "createStream", 12)) {
|
|
//extract a number from the result
|
|
if (pkt->data[10] || pkt->data[19] != 5 || pkt->data[20]) {
|
|
av_log(s, AV_LOG_WARNING, "Unexpected reply on connect()\n");
|
|
} else {
|
|
rt->main_channel_id = av_int2double(AV_RB64(pkt->data + 21));
|
|
}
|
|
|
|
if (!rt->is_input) {
|
|
if ((ret = gen_publish(s, rt)) < 0)
|
|
goto fail;
|
|
} else {
|
|
if ((ret = gen_play(s, rt)) < 0)
|
|
goto fail;
|
|
if ((ret = gen_buffer_time(s, rt)) < 0)
|
|
goto fail;
|
|
}
|
|
}
|
|
|
|
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->data_size;
|
|
const uint8_t *ptr = pkt->data + 11;
|
|
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")) {
|
|
if (!ff_amf_get_field_value(ptr, data_end,
|
|
"description", tmpstr, sizeof(tmpstr)))
|
|
av_log(s, AV_LOG_ERROR, "Server error: %s\n", tmpstr);
|
|
return -1;
|
|
}
|
|
|
|
t = ff_amf_get_field_value(ptr, data_end, "code", tmpstr, sizeof(tmpstr));
|
|
if (!t && !strcmp(tmpstr, "NetStream.Play.Start")) rt->state = STATE_PLAYING;
|
|
if (!t && !strcmp(tmpstr, "NetStream.Play.Stop")) rt->state = STATE_STOPPED;
|
|
if (!t && !strcmp(tmpstr, "NetStream.Play.UnpublishNotify")) rt->state = STATE_STOPPED;
|
|
if (!t && !strcmp(tmpstr, "NetStream.Publish.Start")) rt->state = STATE_PUBLISHING;
|
|
|
|
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 (!memcmp(pkt->data, "\002\000\006_error", 9)) {
|
|
if ((ret = handle_invoke_error(s, pkt)) < 0)
|
|
return ret;
|
|
} else if (!memcmp(pkt->data, "\002\000\007_result", 10)) {
|
|
if ((ret = handle_invoke_result(s, pkt)) < 0)
|
|
return ret;
|
|
} else if (!memcmp(pkt->data, "\002\000\010onStatus", 11)) {
|
|
if ((ret = handle_invoke_status(s, pkt)) < 0)
|
|
return ret;
|
|
} else if (!memcmp(pkt->data, "\002\000\010onBWDone", 11)) {
|
|
if ((ret = gen_check_bw(s, rt)) < 0)
|
|
return ret;
|
|
}
|
|
|
|
return ret;
|
|
}
|
|
|
|
/**
|
|
* 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_dlog(s, "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_PING:
|
|
if ((ret = handle_ping(s, pkt)) < 0)
|
|
return ret;
|
|
break;
|
|
case RTMP_PT_CLIENT_BW:
|
|
if ((ret = handle_client_bw(s, pkt)) < 0)
|
|
return ret;
|
|
break;
|
|
case RTMP_PT_SERVER_BW:
|
|
if ((ret = handle_server_bw(s, pkt)) < 0)
|
|
return ret;
|
|
break;
|
|
case RTMP_PT_INVOKE:
|
|
if ((ret = handle_invoke(s, pkt)) < 0)
|
|
return ret;
|
|
break;
|
|
case RTMP_PT_VIDEO:
|
|
case RTMP_PT_AUDIO:
|
|
case RTMP_PT_METADATA:
|
|
/* 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;
|
|
}
|
|
|
|
/**
|
|
* Interact with the server by receiving and sending RTMP packets until
|
|
* there is some significant data (media data or expected status notification).
|
|
*
|
|
* @param s reading context
|
|
* @param for_header non-zero value tells function to work until it
|
|
* gets notification from the server that playing has been started,
|
|
* otherwise function will work until some media data is received (or
|
|
* an error happens)
|
|
* @return 0 for successful operation, negative value in case of error
|
|
*/
|
|
static int get_packet(URLContext *s, int for_header)
|
|
{
|
|
RTMPContext *rt = s->priv_data;
|
|
int ret;
|
|
uint8_t *p;
|
|
const uint8_t *next;
|
|
uint32_t data_size;
|
|
uint32_t ts, cts, pts=0;
|
|
|
|
if (rt->state == STATE_STOPPED)
|
|
return AVERROR_EOF;
|
|
|
|
for (;;) {
|
|
RTMPPacket rpkt = { 0 };
|
|
if ((ret = ff_rtmp_packet_read(rt->stream, &rpkt,
|
|
rt->in_chunk_size, rt->prev_pkt[0])) <= 0) {
|
|
if (ret == 0) {
|
|
return AVERROR(EAGAIN);
|
|
} else {
|
|
return AVERROR(EIO);
|
|
}
|
|
}
|
|
rt->bytes_read += ret;
|
|
if (rt->bytes_read > rt->last_bytes_read + rt->client_report_size) {
|
|
av_log(s, AV_LOG_DEBUG, "Sending bytes read report\n");
|
|
if ((ret = gen_bytes_read(s, rt, rpkt.timestamp + 1)) < 0)
|
|
return ret;
|
|
rt->last_bytes_read = rt->bytes_read;
|
|
}
|
|
|
|
ret = rtmp_parse_result(s, rt, &rpkt);
|
|
if (ret < 0) {//serious error in current packet
|
|
ff_rtmp_packet_destroy(&rpkt);
|
|
return ret;
|
|
}
|
|
if (rt->state == STATE_STOPPED) {
|
|
ff_rtmp_packet_destroy(&rpkt);
|
|
return AVERROR_EOF;
|
|
}
|
|
if (for_header && (rt->state == STATE_PLAYING || rt->state == STATE_PUBLISHING)) {
|
|
ff_rtmp_packet_destroy(&rpkt);
|
|
return 0;
|
|
}
|
|
if (!rpkt.data_size || !rt->is_input) {
|
|
ff_rtmp_packet_destroy(&rpkt);
|
|
continue;
|
|
}
|
|
if (rpkt.type == RTMP_PT_VIDEO || rpkt.type == RTMP_PT_AUDIO ||
|
|
(rpkt.type == RTMP_PT_NOTIFY && !memcmp("\002\000\012onMetaData", rpkt.data, 13))) {
|
|
ts = rpkt.timestamp;
|
|
|
|
// generate packet header and put data into buffer for FLV demuxer
|
|
rt->flv_off = 0;
|
|
rt->flv_size = rpkt.data_size + 15;
|
|
rt->flv_data = p = av_realloc(rt->flv_data, rt->flv_size);
|
|
bytestream_put_byte(&p, rpkt.type);
|
|
bytestream_put_be24(&p, rpkt.data_size);
|
|
bytestream_put_be24(&p, ts);
|
|
bytestream_put_byte(&p, ts >> 24);
|
|
bytestream_put_be24(&p, 0);
|
|
bytestream_put_buffer(&p, rpkt.data, rpkt.data_size);
|
|
bytestream_put_be32(&p, 0);
|
|
ff_rtmp_packet_destroy(&rpkt);
|
|
return 0;
|
|
} else if (rpkt.type == RTMP_PT_METADATA) {
|
|
// we got raw FLV data, make it available for FLV demuxer
|
|
rt->flv_off = 0;
|
|
rt->flv_size = rpkt.data_size;
|
|
rt->flv_data = av_realloc(rt->flv_data, rt->flv_size);
|
|
/* rewrite timestamps */
|
|
next = rpkt.data;
|
|
ts = rpkt.timestamp;
|
|
while (next - rpkt.data < rpkt.data_size - 11) {
|
|
next++;
|
|
data_size = bytestream_get_be24(&next);
|
|
p=next;
|
|
cts = bytestream_get_be24(&next);
|
|
cts |= bytestream_get_byte(&next) << 24;
|
|
if (pts==0)
|
|
pts=cts;
|
|
ts += cts - pts;
|
|
pts = cts;
|
|
bytestream_put_be24(&p, ts);
|
|
bytestream_put_byte(&p, ts >> 24);
|
|
next += data_size + 3 + 4;
|
|
}
|
|
memcpy(rt->flv_data, rpkt.data, rpkt.data_size);
|
|
ff_rtmp_packet_destroy(&rpkt);
|
|
return 0;
|
|
}
|
|
ff_rtmp_packet_destroy(&rpkt);
|
|
}
|
|
}
|
|
|
|
static int rtmp_close(URLContext *h)
|
|
{
|
|
RTMPContext *rt = h->priv_data;
|
|
int ret = 0;
|
|
|
|
if (!rt->is_input) {
|
|
rt->flv_data = NULL;
|
|
if (rt->out_pkt.data_size)
|
|
ff_rtmp_packet_destroy(&rt->out_pkt);
|
|
if (rt->state > STATE_FCPUBLISH)
|
|
ret = gen_fcunpublish_stream(h, rt);
|
|
}
|
|
if (rt->state > STATE_HANDSHAKED)
|
|
ret = gen_delete_stream(h, rt);
|
|
|
|
free_tracked_methods(rt);
|
|
av_freep(&rt->flv_data);
|
|
ffurl_close(rt->stream);
|
|
return ret;
|
|
}
|
|
|
|
/**
|
|
* Open RTMP connection and verify that the stream can be played.
|
|
*
|
|
* URL syntax: rtmp://server[:port][/app][/playpath]
|
|
* where 'app' is first one or two directories in the path
|
|
* (e.g. /ondemand/, /flash/live/, etc.)
|
|
* and 'playpath' is a file name (the rest of the path,
|
|
* may be prefixed with "mp4:")
|
|
*/
|
|
static int rtmp_open(URLContext *s, const char *uri, int flags)
|
|
{
|
|
RTMPContext *rt = s->priv_data;
|
|
char proto[8], hostname[256], path[1024], *fname;
|
|
char *old_app;
|
|
uint8_t buf[2048];
|
|
int port;
|
|
AVDictionary *opts = NULL;
|
|
int ret;
|
|
|
|
rt->is_input = !(flags & AVIO_FLAG_WRITE);
|
|
|
|
av_url_split(proto, sizeof(proto), NULL, 0, hostname, sizeof(hostname), &port,
|
|
path, sizeof(path), s->filename);
|
|
|
|
if (!strcmp(proto, "rtmpt") || !strcmp(proto, "rtmpts")) {
|
|
if (!strcmp(proto, "rtmpts"))
|
|
av_dict_set(&opts, "ffrtmphttp_tls", "1", 1);
|
|
|
|
/* open the http tunneling connection */
|
|
ff_url_join(buf, sizeof(buf), "ffrtmphttp", NULL, hostname, port, NULL);
|
|
} else if (!strcmp(proto, "rtmps")) {
|
|
/* open the tls connection */
|
|
if (port < 0)
|
|
port = RTMPS_DEFAULT_PORT;
|
|
ff_url_join(buf, sizeof(buf), "tls", NULL, hostname, port, NULL);
|
|
} else if (!strcmp(proto, "rtmpe") || (!strcmp(proto, "rtmpte"))) {
|
|
if (!strcmp(proto, "rtmpte"))
|
|
av_dict_set(&opts, "ffrtmpcrypt_tunneling", "1", 1);
|
|
|
|
/* open the encrypted connection */
|
|
ff_url_join(buf, sizeof(buf), "ffrtmpcrypt", NULL, hostname, port, NULL);
|
|
rt->encrypted = 1;
|
|
} else {
|
|
/* open the tcp connection */
|
|
if (port < 0)
|
|
port = RTMP_DEFAULT_PORT;
|
|
ff_url_join(buf, sizeof(buf), "tcp", NULL, hostname, port, NULL);
|
|
}
|
|
|
|
if ((ret = ffurl_open(&rt->stream, buf, AVIO_FLAG_READ_WRITE,
|
|
&s->interrupt_callback, &opts)) < 0) {
|
|
av_log(s , AV_LOG_ERROR, "Cannot open connection %s\n", buf);
|
|
goto fail;
|
|
}
|
|
|
|
rt->state = STATE_START;
|
|
if ((ret = rtmp_handshake(s, rt)) < 0)
|
|
goto fail;
|
|
|
|
rt->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
|
|
if (!strncmp(path, "/ondemand/", 10)) {
|
|
fname = path + 10;
|
|
memcpy(rt->app, "ondemand", 9);
|
|
} else {
|
|
char *next = *path ? path + 1 : path;
|
|
char *p = strchr(next, '/');
|
|
if (!p) {
|
|
fname = next;
|
|
rt->app[0] = '\0';
|
|
} else {
|
|
// make sure we do not mismatch a playpath for an application instance
|
|
char *c = strchr(p + 1, ':');
|
|
fname = strchr(p + 1, '/');
|
|
if (!fname || (c && c < fname)) {
|
|
fname = p + 1;
|
|
av_strlcpy(rt->app, path + 1, p - path);
|
|
} else {
|
|
fname++;
|
|
av_strlcpy(rt->app, path + 1, fname - path - 1);
|
|
}
|
|
}
|
|
}
|
|
|
|
if (old_app) {
|
|
// The name of application has been defined by the user, override it.
|
|
av_free(rt->app);
|
|
rt->app = old_app;
|
|
}
|
|
|
|
if (!rt->playpath) {
|
|
int len = strlen(fname);
|
|
|
|
rt->playpath = av_malloc(PLAYPATH_MAX_LENGTH);
|
|
if (!rt->playpath) {
|
|
ret = AVERROR(ENOMEM);
|
|
goto fail;
|
|
}
|
|
|
|
if (!strchr(fname, ':') && len >= 4 &&
|
|
(!strcmp(fname + len - 4, ".f4v") ||
|
|
!strcmp(fname + len - 4, ".mp4"))) {
|
|
memcpy(rt->playpath, "mp4:", 5);
|
|
} else if (len >= 4 && !strcmp(fname + len - 4, ".flv")) {
|
|
fname[len - 4] = '\0';
|
|
} else {
|
|
rt->playpath[0] = 0;
|
|
}
|
|
strncat(rt->playpath, fname, PLAYPATH_MAX_LENGTH - 5);
|
|
}
|
|
|
|
if (!rt->tcurl) {
|
|
rt->tcurl = av_malloc(TCURL_MAX_LENGTH);
|
|
if (!rt->tcurl) {
|
|
ret = AVERROR(ENOMEM);
|
|
goto fail;
|
|
}
|
|
ff_url_join(rt->tcurl, TCURL_MAX_LENGTH, proto, NULL, hostname,
|
|
port, "/%s", rt->app);
|
|
}
|
|
|
|
if (!rt->flashver) {
|
|
rt->flashver = av_malloc(FLASHVER_MAX_LENGTH);
|
|
if (!rt->flashver) {
|
|
ret = AVERROR(ENOMEM);
|
|
goto fail;
|
|
}
|
|
if (rt->is_input) {
|
|
snprintf(rt->flashver, FLASHVER_MAX_LENGTH, "%s %d,%d,%d,%d",
|
|
RTMP_CLIENT_PLATFORM, RTMP_CLIENT_VER1, RTMP_CLIENT_VER2,
|
|
RTMP_CLIENT_VER3, RTMP_CLIENT_VER4);
|
|
} else {
|
|
snprintf(rt->flashver, FLASHVER_MAX_LENGTH,
|
|
"FMLE/3.0 (compatible; %s)", LIBAVFORMAT_IDENT);
|
|
}
|
|
}
|
|
|
|
rt->client_report_size = 1048576;
|
|
rt->bytes_read = 0;
|
|
rt->last_bytes_read = 0;
|
|
rt->server_bw = 2500000;
|
|
|
|
av_log(s, AV_LOG_DEBUG, "Proto = %s, path = %s, app = %s, fname = %s\n",
|
|
proto, path, rt->app, rt->playpath);
|
|
if ((ret = gen_connect(s, rt)) < 0)
|
|
goto fail;
|
|
|
|
do {
|
|
ret = get_packet(s, 1);
|
|
} while (ret == EAGAIN);
|
|
if (ret < 0)
|
|
goto fail;
|
|
|
|
if (rt->is_input) {
|
|
// generate FLV header for demuxer
|
|
rt->flv_size = 13;
|
|
rt->flv_data = av_realloc(rt->flv_data, rt->flv_size);
|
|
rt->flv_off = 0;
|
|
memcpy(rt->flv_data, "FLV\1\5\0\0\0\011\0\0\0\0", rt->flv_size);
|
|
} else {
|
|
rt->flv_size = 0;
|
|
rt->flv_data = NULL;
|
|
rt->flv_off = 0;
|
|
rt->skip_bytes = 13;
|
|
}
|
|
|
|
s->max_packet_size = rt->stream->max_packet_size;
|
|
s->is_streamed = 1;
|
|
return 0;
|
|
|
|
fail:
|
|
av_dict_free(&opts);
|
|
rtmp_close(s);
|
|
return ret;
|
|
}
|
|
|
|
static int rtmp_read(URLContext *s, uint8_t *buf, int size)
|
|
{
|
|
RTMPContext *rt = s->priv_data;
|
|
int orig_size = size;
|
|
int ret;
|
|
|
|
while (size > 0) {
|
|
int data_left = rt->flv_size - rt->flv_off;
|
|
|
|
if (data_left >= size) {
|
|
memcpy(buf, rt->flv_data + rt->flv_off, size);
|
|
rt->flv_off += size;
|
|
return orig_size;
|
|
}
|
|
if (data_left > 0) {
|
|
memcpy(buf, rt->flv_data + rt->flv_off, data_left);
|
|
buf += data_left;
|
|
size -= data_left;
|
|
rt->flv_off = rt->flv_size;
|
|
return data_left;
|
|
}
|
|
if ((ret = get_packet(s, 0)) < 0)
|
|
return ret;
|
|
}
|
|
return orig_size;
|
|
}
|
|
|
|
static int rtmp_write(URLContext *s, const uint8_t *buf, int size)
|
|
{
|
|
RTMPContext *rt = s->priv_data;
|
|
int size_temp = size;
|
|
int pktsize, pkttype;
|
|
uint32_t ts;
|
|
const uint8_t *buf_temp = buf;
|
|
uint8_t c;
|
|
int ret;
|
|
|
|
do {
|
|
if (rt->skip_bytes) {
|
|
int skip = FFMIN(rt->skip_bytes, size_temp);
|
|
buf_temp += skip;
|
|
size_temp -= skip;
|
|
rt->skip_bytes -= skip;
|
|
continue;
|
|
}
|
|
|
|
if (rt->flv_header_bytes < 11) {
|
|
const uint8_t *header = rt->flv_header;
|
|
int copy = FFMIN(11 - rt->flv_header_bytes, size_temp);
|
|
bytestream_get_buffer(&buf_temp, rt->flv_header + rt->flv_header_bytes, copy);
|
|
rt->flv_header_bytes += copy;
|
|
size_temp -= copy;
|
|
if (rt->flv_header_bytes < 11)
|
|
break;
|
|
|
|
pkttype = bytestream_get_byte(&header);
|
|
pktsize = bytestream_get_be24(&header);
|
|
ts = bytestream_get_be24(&header);
|
|
ts |= bytestream_get_byte(&header) << 24;
|
|
bytestream_get_be24(&header);
|
|
rt->flv_size = pktsize;
|
|
|
|
//force 12bytes header
|
|
if (((pkttype == RTMP_PT_VIDEO || pkttype == RTMP_PT_AUDIO) && ts == 0) ||
|
|
pkttype == RTMP_PT_NOTIFY) {
|
|
if (pkttype == RTMP_PT_NOTIFY)
|
|
pktsize += 16;
|
|
rt->prev_pkt[1][RTMP_SOURCE_CHANNEL].channel_id = 0;
|
|
}
|
|
|
|
//this can be a big packet, it's better to send it right here
|
|
if ((ret = ff_rtmp_packet_create(&rt->out_pkt, RTMP_SOURCE_CHANNEL,
|
|
pkttype, ts, pktsize)) < 0)
|
|
return ret;
|
|
|
|
rt->out_pkt.extra = rt->main_channel_id;
|
|
rt->flv_data = rt->out_pkt.data;
|
|
|
|
if (pkttype == RTMP_PT_NOTIFY)
|
|
ff_amf_write_string(&rt->flv_data, "@setDataFrame");
|
|
}
|
|
|
|
if (rt->flv_size - rt->flv_off > size_temp) {
|
|
bytestream_get_buffer(&buf_temp, rt->flv_data + rt->flv_off, size_temp);
|
|
rt->flv_off += size_temp;
|
|
size_temp = 0;
|
|
} else {
|
|
bytestream_get_buffer(&buf_temp, rt->flv_data + rt->flv_off, rt->flv_size - rt->flv_off);
|
|
size_temp -= rt->flv_size - rt->flv_off;
|
|
rt->flv_off += rt->flv_size - rt->flv_off;
|
|
}
|
|
|
|
if (rt->flv_off == rt->flv_size) {
|
|
rt->skip_bytes = 4;
|
|
|
|
if ((ret = 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], c)) <= 0)
|
|
return ret;
|
|
|
|
if ((ret = rtmp_parse_result(s, rt, &rpkt)) < 0)
|
|
return ret;
|
|
|
|
ff_rtmp_packet_destroy(&rpkt);
|
|
}
|
|
|
|
return size;
|
|
}
|
|
|
|
#define OFFSET(x) offsetof(RTMPContext, x)
|
|
#define DEC AV_OPT_FLAG_DECODING_PARAM
|
|
#define ENC AV_OPT_FLAG_ENCODING_PARAM
|
|
|
|
static const AVOption rtmp_options[] = {
|
|
{"rtmp_app", "Name of application to connect to on the RTMP server", OFFSET(app), AV_OPT_TYPE_STRING, {.str = NULL }, 0, 0, DEC|ENC},
|
|
{"rtmp_buffer", "Set buffer time in milliseconds. The default is 3000.", OFFSET(client_buffer_time), AV_OPT_TYPE_INT, {3000}, 0, INT_MAX, DEC|ENC},
|
|
{"rtmp_conn", "Append arbitrary AMF data to the Connect message", OFFSET(conn), AV_OPT_TYPE_STRING, {.str = NULL }, 0, 0, DEC|ENC},
|
|
{"rtmp_flashver", "Version of the Flash plugin used to run the SWF player.", OFFSET(flashver), AV_OPT_TYPE_STRING, {.str = NULL }, 0, 0, DEC|ENC},
|
|
{"rtmp_flush_interval", "Number of packets flushed in the same request (RTMPT only).", OFFSET(flush_interval), AV_OPT_TYPE_INT, {10}, 0, INT_MAX, ENC},
|
|
{"rtmp_live", "Specify that the media is a live stream.", OFFSET(live), AV_OPT_TYPE_INT, {-2}, INT_MIN, INT_MAX, DEC, "rtmp_live"},
|
|
{"any", "both", 0, AV_OPT_TYPE_CONST, {-2}, 0, 0, DEC, "rtmp_live"},
|
|
{"live", "live stream", 0, AV_OPT_TYPE_CONST, {-1}, 0, 0, DEC, "rtmp_live"},
|
|
{"recorded", "recorded stream", 0, AV_OPT_TYPE_CONST, {0}, 0, 0, DEC, "rtmp_live"},
|
|
{"rtmp_pageurl", "URL of the web page in which the media was embedded. By default no value will be sent.", OFFSET(pageurl), AV_OPT_TYPE_STRING, {.str = NULL }, 0, 0, DEC},
|
|
{"rtmp_playpath", "Stream identifier to play or to publish", OFFSET(playpath), AV_OPT_TYPE_STRING, {.str = NULL }, 0, 0, DEC|ENC},
|
|
{"rtmp_subscribe", "Name of live stream to subscribe to. Defaults to rtmp_playpath.", OFFSET(subscribe), AV_OPT_TYPE_STRING, {.str = NULL }, 0, 0, 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_tcurl", "URL of the target stream. Defaults to proto://host[:port]/app.", OFFSET(tcurl), AV_OPT_TYPE_STRING, {.str = NULL }, 0, 0, DEC|ENC},
|
|
{ NULL },
|
|
};
|
|
|
|
#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, \
|
|
}; \
|
|
\
|
|
URLProtocol ff_##flavor##_protocol = { \
|
|
.name = #flavor, \
|
|
.url_open = rtmp_open, \
|
|
.url_read = rtmp_read, \
|
|
.url_write = rtmp_write, \
|
|
.url_close = rtmp_close, \
|
|
.priv_data_size = sizeof(RTMPContext), \
|
|
.flags = URL_PROTOCOL_FLAG_NETWORK, \
|
|
.priv_data_class= &flavor##_class, \
|
|
};
|
|
|
|
|
|
RTMP_PROTOCOL(rtmp)
|
|
RTMP_PROTOCOL(rtmpe)
|
|
RTMP_PROTOCOL(rtmps)
|
|
RTMP_PROTOCOL(rtmpt)
|
|
RTMP_PROTOCOL(rtmpte)
|
|
RTMP_PROTOCOL(rtmpts)
|