mirror of
https://github.com/FFmpeg/FFmpeg.git
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3edff185ab
* qatar/master: (21 commits)
ipmovie: do not read audio packets before the codec is known
truemotion2: check size before GetBitContext initialisation
avio: Only do implicit network initialization for network protocols
avio: Add an URLProtocol flag for indicating that a protocol uses network
adpcm: ADPCM Electronic Arts has always two channels
matroskadec: Fix a bug where a pointer was cached to an array that might later move due to a realloc()
fate: Add missing reference file from 9b4767e4
.
mov: Support MOV_CH_LAYOUT_USE_DESCRIPTIONS for labeled descriptions.
4xm: Prevent buffer overreads.
mjpegdec: parse RSTn to prevent skipping other data in mjpeg_decode_scan
vp3: add fate test for non-zero last coefficient
vp3: fix streams with non-zero last coefficient
swscale: remove unused U/V arguments from yuv2rgb_write().
timer: K&R formatting cosmetics
lavf: cosmetics, reformat av_read_frame().
lavf: refactor av_read_frame() to make it easier to understand.
Report an error if pitch_lag is zero in AMR-NB decoder.
Revert "4xm: Prevent buffer overreads."
4xm: Prevent buffer overreads.
4xm: pass the correct remaining buffer size to decode_i2_frame().
...
Conflicts:
libavcodec/4xm.c
libavcodec/mjpegdec.c
libavcodec/truemotion2.c
libavformat/ipmovie.c
libavformat/mov_chan.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
1005 lines
34 KiB
C
1005 lines
34 KiB
C
/*
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* RTMP network protocol
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* Copyright (c) 2009 Kostya Shishkov
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*
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* This file is part of FFmpeg.
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*
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* FFmpeg is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Lesser General Public
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* License as published by the Free Software Foundation; either
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* version 2.1 of the License, or (at your option) any later version.
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*
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* FFmpeg is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Lesser General Public License for more details.
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*
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* You should have received a copy of the GNU Lesser General Public
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* License along with FFmpeg; if not, write to the Free Software
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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*/
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/**
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* @file
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* RTMP protocol
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*/
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#include "libavcodec/bytestream.h"
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#include "libavutil/avstring.h"
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#include "libavutil/intfloat.h"
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#include "libavutil/lfg.h"
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#include "libavutil/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 "rtmppkt.h"
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#include "url.h"
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//#define DEBUG
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/** RTMP protocol handler state */
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typedef enum {
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STATE_START, ///< client has not done anything yet
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STATE_HANDSHAKED, ///< client has performed handshake
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STATE_RELEASING, ///< client releasing stream before publish it (for output)
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STATE_FCPUBLISH, ///< client FCPublishing stream (for output)
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STATE_CONNECTING, ///< client connected to server successfully
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STATE_READY, ///< client has sent all needed commands and waits for server reply
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STATE_PLAYING, ///< client has started receiving multimedia data from server
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STATE_PUBLISHING, ///< client has started sending multimedia data to server (for output)
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STATE_STOPPED, ///< the broadcast has been stopped
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} ClientState;
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/** protocol handler context */
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typedef struct RTMPContext {
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URLContext* stream; ///< TCP stream used in interactions with RTMP server
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RTMPPacket prev_pkt[2][RTMP_CHANNELS]; ///< packet history used when reading and sending packets
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int chunk_size; ///< size of the chunks RTMP packets are divided into
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int is_input; ///< input/output flag
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char playpath[256]; ///< path to filename to play (with possible "mp4:" prefix)
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char app[128]; ///< application
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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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RTMPPacket out_pkt; ///< rtmp packet, created from flv a/v or metadata (for output)
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uint32_t client_report_size; ///< number of bytes after which client should report to server
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uint32_t bytes_read; ///< number of bytes read from server
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uint32_t last_bytes_read; ///< number of bytes read last reported to server
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int skip_bytes; ///< number of bytes to skip from the input FLV stream in the next write call
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uint8_t flv_header[11]; ///< partial incoming flv packet header
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int flv_header_bytes; ///< number of initialized bytes in flv_header
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int nb_invokes; ///< keeps track of invoke messages
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int create_stream_invoke; ///< invoke id for the create stream command
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} 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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/**
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* Generate 'connect' call and send it to the server.
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*/
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static void gen_connect(URLContext *s, RTMPContext *rt, const char *proto,
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const char *host, int port)
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{
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RTMPPacket pkt;
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uint8_t ver[64], *p;
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char tcurl[512];
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ff_rtmp_packet_create(&pkt, RTMP_SYSTEM_CHANNEL, RTMP_PT_INVOKE, 0, 4096);
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p = pkt.data;
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ff_url_join(tcurl, sizeof(tcurl), proto, NULL, host, port, "/%s", rt->app);
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ff_amf_write_string(&p, "connect");
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ff_amf_write_number(&p, ++rt->nb_invokes);
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ff_amf_write_object_start(&p);
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ff_amf_write_field_name(&p, "app");
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ff_amf_write_string(&p, rt->app);
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if (rt->is_input) {
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snprintf(ver, sizeof(ver), "%s %d,%d,%d,%d", RTMP_CLIENT_PLATFORM, RTMP_CLIENT_VER1,
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RTMP_CLIENT_VER2, RTMP_CLIENT_VER3, RTMP_CLIENT_VER4);
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} else {
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snprintf(ver, sizeof(ver), "FMLE/3.0 (compatible; %s)", LIBAVFORMAT_IDENT);
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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, ver);
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ff_amf_write_field_name(&p, "tcUrl");
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ff_amf_write_string(&p, tcurl);
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if (rt->is_input) {
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ff_amf_write_field_name(&p, "fpad");
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ff_amf_write_bool(&p, 0);
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ff_amf_write_field_name(&p, "capabilities");
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ff_amf_write_number(&p, 15.0);
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ff_amf_write_field_name(&p, "audioCodecs");
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ff_amf_write_number(&p, 1639.0);
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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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}
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ff_amf_write_object_end(&p);
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pkt.data_size = p - pkt.data;
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ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size, rt->prev_pkt[1]);
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ff_rtmp_packet_destroy(&pkt);
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}
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/**
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* Generate 'releaseStream' call and send it to the server. It should make
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* the server release some channel for media streams.
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*/
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static void gen_release_stream(URLContext *s, RTMPContext *rt)
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{
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RTMPPacket pkt;
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uint8_t *p;
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ff_rtmp_packet_create(&pkt, RTMP_SYSTEM_CHANNEL, RTMP_PT_INVOKE, 0,
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29 + strlen(rt->playpath));
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av_log(s, AV_LOG_DEBUG, "Releasing stream...\n");
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p = pkt.data;
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ff_amf_write_string(&p, "releaseStream");
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ff_amf_write_number(&p, ++rt->nb_invokes);
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ff_amf_write_null(&p);
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ff_amf_write_string(&p, rt->playpath);
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ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size, rt->prev_pkt[1]);
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ff_rtmp_packet_destroy(&pkt);
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}
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/**
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* Generate 'FCPublish' call and send it to the server. It should make
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* the server preapare for receiving media streams.
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*/
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static void gen_fcpublish_stream(URLContext *s, RTMPContext *rt)
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{
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RTMPPacket pkt;
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uint8_t *p;
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ff_rtmp_packet_create(&pkt, RTMP_SYSTEM_CHANNEL, RTMP_PT_INVOKE, 0,
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25 + strlen(rt->playpath));
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av_log(s, AV_LOG_DEBUG, "FCPublish stream...\n");
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p = pkt.data;
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ff_amf_write_string(&p, "FCPublish");
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ff_amf_write_number(&p, ++rt->nb_invokes);
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ff_amf_write_null(&p);
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ff_amf_write_string(&p, rt->playpath);
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ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size, rt->prev_pkt[1]);
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ff_rtmp_packet_destroy(&pkt);
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}
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/**
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* Generate 'FCUnpublish' call and send it to the server. It should make
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* the server destroy stream.
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*/
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static void gen_fcunpublish_stream(URLContext *s, RTMPContext *rt)
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{
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RTMPPacket pkt;
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uint8_t *p;
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ff_rtmp_packet_create(&pkt, RTMP_SYSTEM_CHANNEL, RTMP_PT_INVOKE, 0,
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27 + strlen(rt->playpath));
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av_log(s, AV_LOG_DEBUG, "UnPublishing stream...\n");
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p = pkt.data;
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ff_amf_write_string(&p, "FCUnpublish");
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ff_amf_write_number(&p, ++rt->nb_invokes);
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ff_amf_write_null(&p);
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ff_amf_write_string(&p, rt->playpath);
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ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size, rt->prev_pkt[1]);
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ff_rtmp_packet_destroy(&pkt);
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}
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/**
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* Generate 'createStream' call and send it to the server. It should make
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* the server allocate some channel for media streams.
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*/
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static void gen_create_stream(URLContext *s, RTMPContext *rt)
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{
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RTMPPacket pkt;
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uint8_t *p;
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av_log(s, AV_LOG_DEBUG, "Creating stream...\n");
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ff_rtmp_packet_create(&pkt, RTMP_SYSTEM_CHANNEL, RTMP_PT_INVOKE, 0, 25);
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p = pkt.data;
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ff_amf_write_string(&p, "createStream");
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ff_amf_write_number(&p, ++rt->nb_invokes);
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ff_amf_write_null(&p);
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rt->create_stream_invoke = rt->nb_invokes;
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ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size, rt->prev_pkt[1]);
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ff_rtmp_packet_destroy(&pkt);
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}
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/**
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* Generate 'deleteStream' call and send it to the server. It should make
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* the server remove some channel for media streams.
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*/
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static void gen_delete_stream(URLContext *s, RTMPContext *rt)
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{
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RTMPPacket pkt;
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uint8_t *p;
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av_log(s, AV_LOG_DEBUG, "Deleting stream...\n");
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ff_rtmp_packet_create(&pkt, RTMP_SYSTEM_CHANNEL, RTMP_PT_INVOKE, 0, 34);
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p = pkt.data;
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ff_amf_write_string(&p, "deleteStream");
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ff_amf_write_number(&p, ++rt->nb_invokes);
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ff_amf_write_null(&p);
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ff_amf_write_number(&p, rt->main_channel_id);
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ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size, rt->prev_pkt[1]);
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ff_rtmp_packet_destroy(&pkt);
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}
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/**
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* Generate 'play' call and send it to the server, then ping the server
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* to start actual playing.
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*/
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static void gen_play(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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av_log(s, AV_LOG_DEBUG, "Sending play command for '%s'\n", rt->playpath);
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ff_rtmp_packet_create(&pkt, RTMP_VIDEO_CHANNEL, RTMP_PT_INVOKE, 0,
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20 + strlen(rt->playpath));
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pkt.extra = rt->main_channel_id;
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p = pkt.data;
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ff_amf_write_string(&p, "play");
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ff_amf_write_number(&p, ++rt->nb_invokes);
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ff_amf_write_null(&p);
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ff_amf_write_string(&p, rt->playpath);
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ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size, rt->prev_pkt[1]);
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ff_rtmp_packet_destroy(&pkt);
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// set client buffer time disguised in ping packet
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ff_rtmp_packet_create(&pkt, RTMP_NETWORK_CHANNEL, RTMP_PT_PING, 1, 10);
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p = pkt.data;
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bytestream_put_be16(&p, 3);
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bytestream_put_be32(&p, 1);
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bytestream_put_be32(&p, 256); //TODO: what is a good value here?
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ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size, rt->prev_pkt[1]);
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ff_rtmp_packet_destroy(&pkt);
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}
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/**
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* Generate 'publish' call and send it to the server.
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*/
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static void gen_publish(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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av_log(s, AV_LOG_DEBUG, "Sending publish command for '%s'\n", rt->playpath);
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ff_rtmp_packet_create(&pkt, RTMP_SOURCE_CHANNEL, RTMP_PT_INVOKE, 0,
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30 + strlen(rt->playpath));
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pkt.extra = rt->main_channel_id;
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p = pkt.data;
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ff_amf_write_string(&p, "publish");
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ff_amf_write_number(&p, ++rt->nb_invokes);
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ff_amf_write_null(&p);
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ff_amf_write_string(&p, rt->playpath);
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ff_amf_write_string(&p, "live");
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ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size, rt->prev_pkt[1]);
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ff_rtmp_packet_destroy(&pkt);
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}
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/**
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* Generate ping reply and send it to the server.
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*/
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static void gen_pong(URLContext *s, RTMPContext *rt, RTMPPacket *ppkt)
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{
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RTMPPacket pkt;
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uint8_t *p;
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ff_rtmp_packet_create(&pkt, RTMP_NETWORK_CHANNEL, RTMP_PT_PING, ppkt->timestamp + 1, 6);
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p = pkt.data;
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bytestream_put_be16(&p, 7);
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bytestream_put_be32(&p, AV_RB32(ppkt->data+2));
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ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size, rt->prev_pkt[1]);
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ff_rtmp_packet_destroy(&pkt);
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}
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/**
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* Generate report on bytes read so far and send it to the server.
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*/
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static void gen_bytes_read(URLContext *s, RTMPContext *rt, uint32_t ts)
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{
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RTMPPacket pkt;
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uint8_t *p;
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ff_rtmp_packet_create(&pkt, RTMP_NETWORK_CHANNEL, RTMP_PT_BYTES_READ, ts, 4);
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p = pkt.data;
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bytestream_put_be32(&p, rt->bytes_read);
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ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size, rt->prev_pkt[1]);
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ff_rtmp_packet_destroy(&pkt);
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}
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//TODO: Move HMAC code somewhere. Eventually.
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#define HMAC_IPAD_VAL 0x36
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#define HMAC_OPAD_VAL 0x5C
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/**
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* Calculate HMAC-SHA2 digest for RTMP handshake packets.
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*
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* @param src input buffer
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* @param len input buffer length (should be 1536)
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* @param gap offset in buffer where 32 bytes should not be taken into account
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* when calculating digest (since it will be used to store that digest)
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* @param key digest key
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* @param keylen digest key length
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* @param dst buffer where calculated digest will be stored (32 bytes)
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*/
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static void rtmp_calc_digest(const uint8_t *src, int len, int gap,
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const uint8_t *key, int keylen, uint8_t *dst)
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{
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struct AVSHA *sha;
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uint8_t hmac_buf[64+32] = {0};
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int i;
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sha = av_mallocz(av_sha_size);
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if (keylen < 64) {
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memcpy(hmac_buf, key, keylen);
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} else {
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av_sha_init(sha, 256);
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av_sha_update(sha,key, keylen);
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av_sha_final(sha, hmac_buf);
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}
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for (i = 0; i < 64; i++)
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hmac_buf[i] ^= HMAC_IPAD_VAL;
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av_sha_init(sha, 256);
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av_sha_update(sha, hmac_buf, 64);
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if (gap <= 0) {
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av_sha_update(sha, src, len);
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} else { //skip 32 bytes used for storing digest
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av_sha_update(sha, src, gap);
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av_sha_update(sha, src + gap + 32, len - gap - 32);
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}
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av_sha_final(sha, hmac_buf + 64);
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|
|
for (i = 0; i < 64; i++)
|
|
hmac_buf[i] ^= HMAC_IPAD_VAL ^ HMAC_OPAD_VAL; //reuse XORed key for opad
|
|
av_sha_init(sha, 256);
|
|
av_sha_update(sha, hmac_buf, 64+32);
|
|
av_sha_final(sha, dst);
|
|
|
|
av_free(sha);
|
|
}
|
|
|
|
/**
|
|
* Put HMAC-SHA2 digest of packet data (except for the bytes where this digest
|
|
* will be stored) into that packet.
|
|
*
|
|
* @param buf handshake data (1536 bytes)
|
|
* @return offset to the digest inside input data
|
|
*/
|
|
static int rtmp_handshake_imprint_with_digest(uint8_t *buf)
|
|
{
|
|
int i, digest_pos = 0;
|
|
|
|
for (i = 8; i < 12; i++)
|
|
digest_pos += buf[i];
|
|
digest_pos = (digest_pos % 728) + 12;
|
|
|
|
rtmp_calc_digest(buf, RTMP_HANDSHAKE_PACKET_SIZE, digest_pos,
|
|
rtmp_player_key, PLAYER_KEY_OPEN_PART_LEN,
|
|
buf + digest_pos);
|
|
return digest_pos;
|
|
}
|
|
|
|
/**
|
|
* Verify that the received server response has the expected digest value.
|
|
*
|
|
* @param buf handshake data received from the server (1536 bytes)
|
|
* @param off position to search digest offset from
|
|
* @return 0 if digest is valid, digest position otherwise
|
|
*/
|
|
static int rtmp_validate_digest(uint8_t *buf, int off)
|
|
{
|
|
int i, digest_pos = 0;
|
|
uint8_t digest[32];
|
|
|
|
for (i = 0; i < 4; i++)
|
|
digest_pos += buf[i + off];
|
|
digest_pos = (digest_pos % 728) + off + 4;
|
|
|
|
rtmp_calc_digest(buf, RTMP_HANDSHAKE_PACKET_SIZE, digest_pos,
|
|
rtmp_server_key, SERVER_KEY_OPEN_PART_LEN,
|
|
digest);
|
|
if (!memcmp(digest, buf + digest_pos, 32))
|
|
return digest_pos;
|
|
return 0;
|
|
}
|
|
|
|
/**
|
|
* Perform handshake with the server by means of exchanging pseudorandom data
|
|
* signed with HMAC-SHA2 digest.
|
|
*
|
|
* @return 0 if handshake succeeds, negative value otherwise
|
|
*/
|
|
static int rtmp_handshake(URLContext *s, RTMPContext *rt)
|
|
{
|
|
AVLFG rnd;
|
|
uint8_t tosend [RTMP_HANDSHAKE_PACKET_SIZE+1] = {
|
|
3, // unencrypted data
|
|
0, 0, 0, 0, // client uptime
|
|
RTMP_CLIENT_VER1,
|
|
RTMP_CLIENT_VER2,
|
|
RTMP_CLIENT_VER3,
|
|
RTMP_CLIENT_VER4,
|
|
};
|
|
uint8_t clientdata[RTMP_HANDSHAKE_PACKET_SIZE];
|
|
uint8_t serverdata[RTMP_HANDSHAKE_PACKET_SIZE+1];
|
|
int i;
|
|
int server_pos, client_pos;
|
|
uint8_t digest[32];
|
|
|
|
av_log(s, AV_LOG_DEBUG, "Handshaking...\n");
|
|
|
|
av_lfg_init(&rnd, 0xDEADC0DE);
|
|
// generate handshake packet - 1536 bytes of pseudorandom data
|
|
for (i = 9; i <= RTMP_HANDSHAKE_PACKET_SIZE; i++)
|
|
tosend[i] = av_lfg_get(&rnd) >> 24;
|
|
client_pos = rtmp_handshake_imprint_with_digest(tosend + 1);
|
|
|
|
ffurl_write(rt->stream, tosend, RTMP_HANDSHAKE_PACKET_SIZE + 1);
|
|
i = ffurl_read_complete(rt->stream, serverdata, RTMP_HANDSHAKE_PACKET_SIZE + 1);
|
|
if (i != RTMP_HANDSHAKE_PACKET_SIZE + 1) {
|
|
av_log(s, AV_LOG_ERROR, "Cannot read RTMP handshake response\n");
|
|
return -1;
|
|
}
|
|
i = ffurl_read_complete(rt->stream, clientdata, RTMP_HANDSHAKE_PACKET_SIZE);
|
|
if (i != RTMP_HANDSHAKE_PACKET_SIZE) {
|
|
av_log(s, AV_LOG_ERROR, "Cannot read RTMP handshake response\n");
|
|
return -1;
|
|
}
|
|
|
|
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) {
|
|
server_pos = rtmp_validate_digest(serverdata + 1, 8);
|
|
if (!server_pos) {
|
|
av_log(s, AV_LOG_ERROR, "Server response validating failed\n");
|
|
return -1;
|
|
}
|
|
}
|
|
|
|
rtmp_calc_digest(tosend + 1 + client_pos, 32, 0,
|
|
rtmp_server_key, sizeof(rtmp_server_key),
|
|
digest);
|
|
rtmp_calc_digest(clientdata, RTMP_HANDSHAKE_PACKET_SIZE-32, 0,
|
|
digest, 32,
|
|
digest);
|
|
if (memcmp(digest, clientdata + RTMP_HANDSHAKE_PACKET_SIZE - 32, 32)) {
|
|
av_log(s, AV_LOG_ERROR, "Signature mismatch\n");
|
|
return -1;
|
|
}
|
|
|
|
for (i = 0; i < RTMP_HANDSHAKE_PACKET_SIZE; i++)
|
|
tosend[i] = av_lfg_get(&rnd) >> 24;
|
|
rtmp_calc_digest(serverdata + 1 + server_pos, 32, 0,
|
|
rtmp_player_key, sizeof(rtmp_player_key),
|
|
digest);
|
|
rtmp_calc_digest(tosend, RTMP_HANDSHAKE_PACKET_SIZE - 32, 0,
|
|
digest, 32,
|
|
tosend + RTMP_HANDSHAKE_PACKET_SIZE - 32);
|
|
|
|
// write reply back to the server
|
|
ffurl_write(rt->stream, tosend, RTMP_HANDSHAKE_PACKET_SIZE);
|
|
} else {
|
|
ffurl_write(rt->stream, serverdata+1, RTMP_HANDSHAKE_PACKET_SIZE);
|
|
}
|
|
|
|
return 0;
|
|
}
|
|
|
|
/**
|
|
* Parse received packet and possibly perform some action depending on
|
|
* the packet contents.
|
|
* @return 0 for no errors, negative values for serious errors which prevent
|
|
* further communications, positive values for uncritical errors
|
|
*/
|
|
static int rtmp_parse_result(URLContext *s, RTMPContext *rt, RTMPPacket *pkt)
|
|
{
|
|
int i, t;
|
|
const uint8_t *data_end = pkt->data + pkt->data_size;
|
|
|
|
#ifdef DEBUG
|
|
ff_rtmp_packet_dump(s, pkt);
|
|
#endif
|
|
|
|
switch (pkt->type) {
|
|
case RTMP_PT_CHUNK_SIZE:
|
|
if (pkt->data_size != 4) {
|
|
av_log(s, AV_LOG_ERROR,
|
|
"Chunk size change packet is not 4 bytes long (%d)\n", pkt->data_size);
|
|
return -1;
|
|
}
|
|
if (!rt->is_input)
|
|
ff_rtmp_packet_write(rt->stream, pkt, rt->chunk_size, rt->prev_pkt[1]);
|
|
rt->chunk_size = AV_RB32(pkt->data);
|
|
if (rt->chunk_size <= 0) {
|
|
av_log(s, AV_LOG_ERROR, "Incorrect chunk size %d\n", rt->chunk_size);
|
|
return -1;
|
|
}
|
|
av_log(s, AV_LOG_DEBUG, "New chunk size = %d\n", rt->chunk_size);
|
|
break;
|
|
case RTMP_PT_PING:
|
|
t = AV_RB16(pkt->data);
|
|
if (t == 6)
|
|
gen_pong(s, rt, pkt);
|
|
break;
|
|
case RTMP_PT_CLIENT_BW:
|
|
if (pkt->data_size < 4) {
|
|
av_log(s, AV_LOG_ERROR,
|
|
"Client bandwidth report packet is less than 4 bytes long (%d)\n",
|
|
pkt->data_size);
|
|
return -1;
|
|
}
|
|
av_log(s, AV_LOG_DEBUG, "Client bandwidth = %d\n", AV_RB32(pkt->data));
|
|
rt->client_report_size = AV_RB32(pkt->data) >> 1;
|
|
break;
|
|
case RTMP_PT_INVOKE:
|
|
//TODO: check for the messages sent for wrong state?
|
|
if (!memcmp(pkt->data, "\002\000\006_error", 9)) {
|
|
uint8_t tmpstr[256];
|
|
|
|
if (!ff_amf_get_field_value(pkt->data + 9, data_end,
|
|
"description", tmpstr, sizeof(tmpstr)))
|
|
av_log(s, AV_LOG_ERROR, "Server error: %s\n",tmpstr);
|
|
return -1;
|
|
} else if (!memcmp(pkt->data, "\002\000\007_result", 10)) {
|
|
switch (rt->state) {
|
|
case STATE_HANDSHAKED:
|
|
if (!rt->is_input) {
|
|
gen_release_stream(s, rt);
|
|
gen_fcpublish_stream(s, rt);
|
|
rt->state = STATE_RELEASING;
|
|
} else {
|
|
rt->state = STATE_CONNECTING;
|
|
}
|
|
gen_create_stream(s, rt);
|
|
break;
|
|
case STATE_FCPUBLISH:
|
|
rt->state = STATE_CONNECTING;
|
|
break;
|
|
case STATE_RELEASING:
|
|
rt->state = STATE_FCPUBLISH;
|
|
/* hack for Wowza Media Server, it does not send result for
|
|
* releaseStream and FCPublish calls */
|
|
if (!pkt->data[10]) {
|
|
int pkt_id = av_int2double(AV_RB64(pkt->data + 11));
|
|
if (pkt_id == rt->create_stream_invoke)
|
|
rt->state = STATE_CONNECTING;
|
|
}
|
|
if (rt->state != STATE_CONNECTING)
|
|
break;
|
|
case STATE_CONNECTING:
|
|
//extract a number from the result
|
|
if (pkt->data[10] || pkt->data[19] != 5 || pkt->data[20]) {
|
|
av_log(s, AV_LOG_WARNING, "Unexpected reply on connect()\n");
|
|
} else {
|
|
rt->main_channel_id = av_int2double(AV_RB64(pkt->data + 21));
|
|
}
|
|
if (rt->is_input) {
|
|
gen_play(s, rt);
|
|
} else {
|
|
gen_publish(s, rt);
|
|
}
|
|
rt->state = STATE_READY;
|
|
break;
|
|
}
|
|
} else if (!memcmp(pkt->data, "\002\000\010onStatus", 11)) {
|
|
const uint8_t* ptr = pkt->data + 11;
|
|
uint8_t tmpstr[256];
|
|
|
|
for (i = 0; i < 2; i++) {
|
|
t = ff_amf_tag_size(ptr, data_end);
|
|
if (t < 0)
|
|
return 1;
|
|
ptr += t;
|
|
}
|
|
t = ff_amf_get_field_value(ptr, data_end,
|
|
"level", tmpstr, sizeof(tmpstr));
|
|
if (!t && !strcmp(tmpstr, "error")) {
|
|
if (!ff_amf_get_field_value(ptr, data_end,
|
|
"description", tmpstr, sizeof(tmpstr)))
|
|
av_log(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;
|
|
}
|
|
break;
|
|
}
|
|
return 0;
|
|
}
|
|
|
|
/**
|
|
* Interact with the server by receiving and sending RTMP packets until
|
|
* there is some significant data (media data or expected status notification).
|
|
*
|
|
* @param s reading context
|
|
* @param for_header non-zero value tells function to work until it
|
|
* gets notification from the server that playing has been started,
|
|
* otherwise function will work until some media data is received (or
|
|
* an error happens)
|
|
* @return 0 for successful operation, negative value in case of error
|
|
*/
|
|
static int get_packet(URLContext *s, int for_header)
|
|
{
|
|
RTMPContext *rt = s->priv_data;
|
|
int ret;
|
|
uint8_t *p;
|
|
const uint8_t *next;
|
|
uint32_t data_size;
|
|
uint32_t ts, cts, pts=0;
|
|
|
|
if (rt->state == STATE_STOPPED)
|
|
return AVERROR_EOF;
|
|
|
|
for (;;) {
|
|
RTMPPacket rpkt = { 0 };
|
|
if ((ret = ff_rtmp_packet_read(rt->stream, &rpkt,
|
|
rt->chunk_size, rt->prev_pkt[0])) <= 0) {
|
|
if (ret == 0) {
|
|
return AVERROR(EAGAIN);
|
|
} else {
|
|
return AVERROR(EIO);
|
|
}
|
|
}
|
|
rt->bytes_read += ret;
|
|
if (rt->bytes_read - rt->last_bytes_read > rt->client_report_size) {
|
|
av_log(s, AV_LOG_DEBUG, "Sending bytes read report\n");
|
|
gen_bytes_read(s, rt, rpkt.timestamp + 1);
|
|
rt->last_bytes_read = rt->bytes_read;
|
|
}
|
|
|
|
ret = rtmp_parse_result(s, rt, &rpkt);
|
|
if (ret < 0) {//serious error in current packet
|
|
ff_rtmp_packet_destroy(&rpkt);
|
|
return -1;
|
|
}
|
|
if (rt->state == STATE_STOPPED) {
|
|
ff_rtmp_packet_destroy(&rpkt);
|
|
return AVERROR_EOF;
|
|
}
|
|
if (for_header && (rt->state == STATE_PLAYING || rt->state == STATE_PUBLISHING)) {
|
|
ff_rtmp_packet_destroy(&rpkt);
|
|
return 0;
|
|
}
|
|
if (!rpkt.data_size || !rt->is_input) {
|
|
ff_rtmp_packet_destroy(&rpkt);
|
|
continue;
|
|
}
|
|
if (rpkt.type == RTMP_PT_VIDEO || rpkt.type == RTMP_PT_AUDIO ||
|
|
(rpkt.type == RTMP_PT_NOTIFY && !memcmp("\002\000\012onMetaData", rpkt.data, 13))) {
|
|
ts = rpkt.timestamp;
|
|
|
|
// generate packet header and put data into buffer for FLV demuxer
|
|
rt->flv_off = 0;
|
|
rt->flv_size = rpkt.data_size + 15;
|
|
rt->flv_data = p = av_realloc(rt->flv_data, rt->flv_size);
|
|
bytestream_put_byte(&p, rpkt.type);
|
|
bytestream_put_be24(&p, rpkt.data_size);
|
|
bytestream_put_be24(&p, ts);
|
|
bytestream_put_byte(&p, ts >> 24);
|
|
bytestream_put_be24(&p, 0);
|
|
bytestream_put_buffer(&p, rpkt.data, rpkt.data_size);
|
|
bytestream_put_be32(&p, 0);
|
|
ff_rtmp_packet_destroy(&rpkt);
|
|
return 0;
|
|
} else if (rpkt.type == RTMP_PT_METADATA) {
|
|
// we got raw FLV data, make it available for FLV demuxer
|
|
rt->flv_off = 0;
|
|
rt->flv_size = rpkt.data_size;
|
|
rt->flv_data = av_realloc(rt->flv_data, rt->flv_size);
|
|
/* rewrite timestamps */
|
|
next = rpkt.data;
|
|
ts = rpkt.timestamp;
|
|
while (next - rpkt.data < rpkt.data_size - 11) {
|
|
next++;
|
|
data_size = bytestream_get_be24(&next);
|
|
p=next;
|
|
cts = bytestream_get_be24(&next);
|
|
cts |= bytestream_get_byte(&next) << 24;
|
|
if (pts==0)
|
|
pts=cts;
|
|
ts += cts - pts;
|
|
pts = cts;
|
|
bytestream_put_be24(&p, ts);
|
|
bytestream_put_byte(&p, ts >> 24);
|
|
next += data_size + 3 + 4;
|
|
}
|
|
memcpy(rt->flv_data, rpkt.data, rpkt.data_size);
|
|
ff_rtmp_packet_destroy(&rpkt);
|
|
return 0;
|
|
}
|
|
ff_rtmp_packet_destroy(&rpkt);
|
|
}
|
|
}
|
|
|
|
static int rtmp_close(URLContext *h)
|
|
{
|
|
RTMPContext *rt = h->priv_data;
|
|
|
|
if (!rt->is_input) {
|
|
rt->flv_data = NULL;
|
|
if (rt->out_pkt.data_size)
|
|
ff_rtmp_packet_destroy(&rt->out_pkt);
|
|
if (rt->state > STATE_FCPUBLISH)
|
|
gen_fcunpublish_stream(h, rt);
|
|
}
|
|
if (rt->state > STATE_HANDSHAKED)
|
|
gen_delete_stream(h, rt);
|
|
|
|
av_freep(&rt->flv_data);
|
|
ffurl_close(rt->stream);
|
|
return 0;
|
|
}
|
|
|
|
/**
|
|
* Open RTMP connection and verify that the stream can be played.
|
|
*
|
|
* URL syntax: rtmp://server[:port][/app][/playpath]
|
|
* where 'app' is first one or two directories in the path
|
|
* (e.g. /ondemand/, /flash/live/, etc.)
|
|
* and 'playpath' is a file name (the rest of the path,
|
|
* may be prefixed with "mp4:")
|
|
*/
|
|
static int rtmp_open(URLContext *s, const char *uri, int flags)
|
|
{
|
|
RTMPContext *rt = s->priv_data;
|
|
char proto[8], hostname[256], path[1024], *fname;
|
|
uint8_t buf[2048];
|
|
int port;
|
|
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 (port < 0)
|
|
port = RTMP_DEFAULT_PORT;
|
|
ff_url_join(buf, sizeof(buf), "tcp", NULL, hostname, port, NULL);
|
|
|
|
if (ffurl_open(&rt->stream, buf, AVIO_FLAG_READ_WRITE,
|
|
&s->interrupt_callback, NULL) < 0) {
|
|
av_log(s , AV_LOG_ERROR, "Cannot open connection %s\n", buf);
|
|
goto fail;
|
|
}
|
|
|
|
rt->state = STATE_START;
|
|
if (rtmp_handshake(s, rt))
|
|
goto fail;
|
|
|
|
rt->chunk_size = 128;
|
|
rt->state = STATE_HANDSHAKED;
|
|
//extract "app" part from path
|
|
if (!strncmp(path, "/ondemand/", 10)) {
|
|
fname = path + 10;
|
|
memcpy(rt->app, "ondemand", 9);
|
|
} else {
|
|
char *p = strchr(path + 1, '/');
|
|
if (!p) {
|
|
fname = path + 1;
|
|
rt->app[0] = '\0';
|
|
} else {
|
|
char *c = strchr(p + 1, ':');
|
|
fname = strchr(p + 1, '/');
|
|
if (!fname || c < fname) {
|
|
fname = p + 1;
|
|
av_strlcpy(rt->app, path + 1, p - path);
|
|
} else {
|
|
fname++;
|
|
av_strlcpy(rt->app, path + 1, fname - path - 1);
|
|
}
|
|
}
|
|
}
|
|
if (!strchr(fname, ':') &&
|
|
(!strcmp(fname + strlen(fname) - 4, ".f4v") ||
|
|
!strcmp(fname + strlen(fname) - 4, ".mp4"))) {
|
|
memcpy(rt->playpath, "mp4:", 5);
|
|
} else {
|
|
rt->playpath[0] = 0;
|
|
}
|
|
strncat(rt->playpath, fname, sizeof(rt->playpath) - 5);
|
|
|
|
rt->client_report_size = 1048576;
|
|
rt->bytes_read = 0;
|
|
rt->last_bytes_read = 0;
|
|
|
|
av_log(s, AV_LOG_DEBUG, "Proto = %s, path = %s, app = %s, fname = %s\n",
|
|
proto, path, rt->app, rt->playpath);
|
|
gen_connect(s, rt, proto, hostname, port);
|
|
|
|
do {
|
|
ret = get_packet(s, 1);
|
|
} while (ret == EAGAIN);
|
|
if (ret < 0)
|
|
goto fail;
|
|
|
|
if (rt->is_input) {
|
|
// generate FLV header for demuxer
|
|
rt->flv_size = 13;
|
|
rt->flv_data = av_realloc(rt->flv_data, rt->flv_size);
|
|
rt->flv_off = 0;
|
|
memcpy(rt->flv_data, "FLV\1\5\0\0\0\011\0\0\0\0", rt->flv_size);
|
|
} else {
|
|
rt->flv_size = 0;
|
|
rt->flv_data = NULL;
|
|
rt->flv_off = 0;
|
|
rt->skip_bytes = 13;
|
|
}
|
|
|
|
s->max_packet_size = rt->stream->max_packet_size;
|
|
s->is_streamed = 1;
|
|
return 0;
|
|
|
|
fail:
|
|
rtmp_close(s);
|
|
return AVERROR(EIO);
|
|
}
|
|
|
|
static int rtmp_read(URLContext *s, uint8_t *buf, int size)
|
|
{
|
|
RTMPContext *rt = s->priv_data;
|
|
int orig_size = size;
|
|
int ret;
|
|
|
|
while (size > 0) {
|
|
int data_left = rt->flv_size - rt->flv_off;
|
|
|
|
if (data_left >= size) {
|
|
memcpy(buf, rt->flv_data + rt->flv_off, size);
|
|
rt->flv_off += size;
|
|
return orig_size;
|
|
}
|
|
if (data_left > 0) {
|
|
memcpy(buf, rt->flv_data + rt->flv_off, data_left);
|
|
buf += data_left;
|
|
size -= data_left;
|
|
rt->flv_off = rt->flv_size;
|
|
return data_left;
|
|
}
|
|
if ((ret = get_packet(s, 0)) < 0)
|
|
return ret;
|
|
}
|
|
return orig_size;
|
|
}
|
|
|
|
static int rtmp_write(URLContext *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;
|
|
|
|
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
|
|
ff_rtmp_packet_create(&rt->out_pkt, RTMP_SOURCE_CHANNEL, pkttype, ts, pktsize);
|
|
rt->out_pkt.extra = rt->main_channel_id;
|
|
rt->flv_data = rt->out_pkt.data;
|
|
|
|
if (pkttype == RTMP_PT_NOTIFY)
|
|
ff_amf_write_string(&rt->flv_data, "@setDataFrame");
|
|
}
|
|
|
|
if (rt->flv_size - rt->flv_off > size_temp) {
|
|
bytestream_get_buffer(&buf_temp, rt->flv_data + rt->flv_off, size_temp);
|
|
rt->flv_off += size_temp;
|
|
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;
|
|
|
|
ff_rtmp_packet_write(rt->stream, &rt->out_pkt, rt->chunk_size, rt->prev_pkt[1]);
|
|
ff_rtmp_packet_destroy(&rt->out_pkt);
|
|
rt->flv_size = 0;
|
|
rt->flv_off = 0;
|
|
rt->flv_header_bytes = 0;
|
|
}
|
|
} while (buf_temp - buf < size);
|
|
return size;
|
|
}
|
|
|
|
URLProtocol ff_rtmp_protocol = {
|
|
.name = "rtmp",
|
|
.url_open = rtmp_open,
|
|
.url_read = rtmp_read,
|
|
.url_write = rtmp_write,
|
|
.url_close = rtmp_close,
|
|
.priv_data_size = sizeof(RTMPContext),
|
|
.flags = URL_PROTOCOL_FLAG_NETWORK,
|
|
};
|