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pineapple/externals/ffmpeg/libavformat/rtmpproto.c

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early-access version 1432
2021-02-09 04:25:58 +01:00
/*
* RTMP network protocol
* Copyright (c) 2009 Konstantin Shishkov
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
/**
* @file
* RTMP protocol
*/
#include "libavcodec/bytestream.h"
#include "libavutil/avstring.h"
#include "libavutil/base64.h"
#include "libavutil/intfloat.h"
#include "libavutil/lfg.h"
#include "libavutil/md5.h"
#include "libavutil/opt.h"
#include "libavutil/random_seed.h"
#include "avformat.h"
#include "internal.h"
#include "network.h"
#include "flv.h"
#include "rtmp.h"
#include "rtmpcrypt.h"
#include "rtmppkt.h"
#include "url.h"
#if CONFIG_ZLIB
#include <zlib.h>
#endif
#define APP_MAX_LENGTH 1024
#define TCURL_MAX_LENGTH 1024
#define FLASHVER_MAX_LENGTH 64
#define RTMP_PKTDATA_DEFAULT_SIZE 4096
#define RTMP_HEADER 11
/** RTMP protocol handler state */
typedef enum {
STATE_START, ///< client has not done anything yet
STATE_HANDSHAKED, ///< client has performed handshake
STATE_FCPUBLISH, ///< client FCPublishing stream (for output)
STATE_PLAYING, ///< client has started receiving multimedia data from server
STATE_SEEKING, ///< client has started the seek operation. Back on STATE_PLAYING when the time comes
STATE_PUBLISHING, ///< client has started sending multimedia data to server (for output)
STATE_RECEIVING, ///< received a publish command (for input)
STATE_SENDING, ///< received a play command (for output)
STATE_STOPPED, ///< the broadcast has been stopped
} ClientState;
typedef struct TrackedMethod {
char *name;
int id;
} TrackedMethod;
/** protocol handler context */
typedef struct RTMPContext {
const AVClass *class;
URLContext* stream; ///< TCP stream used in interactions with RTMP server
RTMPPacket *prev_pkt[2]; ///< packet history used when reading and sending packets ([0] for reading, [1] for writing)
int nb_prev_pkt[2]; ///< number of elements in prev_pkt
int in_chunk_size; ///< size of the chunks incoming RTMP packets are divided into
int out_chunk_size; ///< size of the chunks outgoing RTMP packets are divided into
int is_input; ///< input/output flag
char *playpath; ///< stream identifier to play (with possible "mp4:" prefix)
int live; ///< 0: recorded, -1: live, -2: both
char *app; ///< name of application
char *conn; ///< append arbitrary AMF data to the Connect message
ClientState state; ///< current state
int stream_id; ///< ID assigned by the server for the stream
uint8_t* flv_data; ///< buffer with data for demuxer
int flv_size; ///< current buffer size
int flv_off; ///< number of bytes read from current buffer
int flv_nb_packets; ///< number of flv packets published
RTMPPacket out_pkt; ///< rtmp packet, created from flv a/v or metadata (for output)
uint32_t receive_report_size; ///< number of bytes after which we should report the number of received bytes to the peer
uint64_t bytes_read; ///< number of bytes read from server
uint64_t last_bytes_read; ///< number of bytes read last reported to server
uint32_t last_timestamp; ///< last timestamp received in a packet
int skip_bytes; ///< number of bytes to skip from the input FLV stream in the next write call
int has_audio; ///< presence of audio data
int has_video; ///< presence of video data
int received_metadata; ///< Indicates if we have received metadata about the streams
uint8_t flv_header[RTMP_HEADER]; ///< partial incoming flv packet header
int flv_header_bytes; ///< number of initialized bytes in flv_header
int nb_invokes; ///< keeps track of invoke messages
char* tcurl; ///< url of the target stream
char* flashver; ///< version of the flash plugin
char* swfhash; ///< SHA256 hash of the decompressed SWF file (32 bytes)
int swfhash_len; ///< length of the SHA256 hash
int swfsize; ///< size of the decompressed SWF file
char* swfurl; ///< url of the swf player
char* swfverify; ///< URL to player swf file, compute hash/size automatically
char swfverification[42]; ///< hash of the SWF verification
char* pageurl; ///< url of the web page
char* subscribe; ///< name of live stream to subscribe
int max_sent_unacked; ///< max unacked sent bytes
int client_buffer_time; ///< client buffer time in ms
int flush_interval; ///< number of packets flushed in the same request (RTMPT only)
int encrypted; ///< use an encrypted connection (RTMPE only)
TrackedMethod*tracked_methods; ///< tracked methods buffer
int nb_tracked_methods; ///< number of tracked methods
int tracked_methods_size; ///< size of the tracked methods buffer
int listen; ///< listen mode flag
int listen_timeout; ///< listen timeout to wait for new connections
int nb_streamid; ///< The next stream id to return on createStream calls
double duration; ///< Duration of the stream in seconds as returned by the server (only valid if non-zero)
char username[50];
char password[50];
char auth_params[500];
int do_reconnect;
int auth_tried;
} RTMPContext;
#define PLAYER_KEY_OPEN_PART_LEN 30 ///< length of partial key used for first client digest signing
/** Client key used for digest signing */
static const uint8_t rtmp_player_key[] = {
'G', 'e', 'n', 'u', 'i', 'n', 'e', ' ', 'A', 'd', 'o', 'b', 'e', ' ',
'F', 'l', 'a', 's', 'h', ' ', 'P', 'l', 'a', 'y', 'e', 'r', ' ', '0', '0', '1',
0xF0, 0xEE, 0xC2, 0x4A, 0x80, 0x68, 0xBE, 0xE8, 0x2E, 0x00, 0xD0, 0xD1, 0x02,
0x9E, 0x7E, 0x57, 0x6E, 0xEC, 0x5D, 0x2D, 0x29, 0x80, 0x6F, 0xAB, 0x93, 0xB8,
0xE6, 0x36, 0xCF, 0xEB, 0x31, 0xAE
};
#define SERVER_KEY_OPEN_PART_LEN 36 ///< length of partial key used for first server digest signing
/** Key used for RTMP server digest signing */
static const uint8_t rtmp_server_key[] = {
'G', 'e', 'n', 'u', 'i', 'n', 'e', ' ', 'A', 'd', 'o', 'b', 'e', ' ',
'F', 'l', 'a', 's', 'h', ' ', 'M', 'e', 'd', 'i', 'a', ' ',
'S', 'e', 'r', 'v', 'e', 'r', ' ', '0', '0', '1',
0xF0, 0xEE, 0xC2, 0x4A, 0x80, 0x68, 0xBE, 0xE8, 0x2E, 0x00, 0xD0, 0xD1, 0x02,
0x9E, 0x7E, 0x57, 0x6E, 0xEC, 0x5D, 0x2D, 0x29, 0x80, 0x6F, 0xAB, 0x93, 0xB8,
0xE6, 0x36, 0xCF, 0xEB, 0x31, 0xAE
};
static int handle_chunk_size(URLContext *s, RTMPPacket *pkt);
static int handle_window_ack_size(URLContext *s, RTMPPacket *pkt);
static int handle_set_peer_bw(URLContext *s, RTMPPacket *pkt);
static int add_tracked_method(RTMPContext *rt, const char *name, int id)
{
int err;
if (rt->nb_tracked_methods + 1 > rt->tracked_methods_size) {
rt->tracked_methods_size = (rt->nb_tracked_methods + 1) * 2;
if ((err = av_reallocp_array(&rt->tracked_methods, rt->tracked_methods_size,
sizeof(*rt->tracked_methods))) < 0) {
rt->nb_tracked_methods = 0;
rt->tracked_methods_size = 0;
return err;
}
}
rt->tracked_methods[rt->nb_tracked_methods].name = av_strdup(name);
if (!rt->tracked_methods[rt->nb_tracked_methods].name)
return AVERROR(ENOMEM);
rt->tracked_methods[rt->nb_tracked_methods].id = id;
rt->nb_tracked_methods++;
return 0;
}
static void del_tracked_method(RTMPContext *rt, int index)
{
memmove(&rt->tracked_methods[index], &rt->tracked_methods[index + 1],
sizeof(*rt->tracked_methods) * (rt->nb_tracked_methods - index - 1));
rt->nb_tracked_methods--;
}
static int find_tracked_method(URLContext *s, RTMPPacket *pkt, int offset,
char **tracked_method)
{
RTMPContext *rt = s->priv_data;
GetByteContext gbc;
double pkt_id;
int ret;
int i;
bytestream2_init(&gbc, pkt->data + offset, pkt->size - offset);
if ((ret = ff_amf_read_number(&gbc, &pkt_id)) < 0)
return ret;
for (i = 0; i < rt->nb_tracked_methods; i++) {
if (rt->tracked_methods[i].id != pkt_id)
continue;
*tracked_method = rt->tracked_methods[i].name;
del_tracked_method(rt, i);
break;
}
return 0;
}
static void free_tracked_methods(RTMPContext *rt)
{
int i;
for (i = 0; i < rt->nb_tracked_methods; i ++)
av_freep(&rt->tracked_methods[i].name);
av_freep(&rt->tracked_methods);
rt->tracked_methods_size = 0;
rt->nb_tracked_methods = 0;
}
static int rtmp_send_packet(RTMPContext *rt, RTMPPacket *pkt, int track)
{
int ret;
if (pkt->type == RTMP_PT_INVOKE && track) {
GetByteContext gbc;
char name[128];
double pkt_id;
int len;
bytestream2_init(&gbc, pkt->data, pkt->size);
if ((ret = ff_amf_read_string(&gbc, name, sizeof(name), &len)) < 0)
goto fail;
if ((ret = ff_amf_read_number(&gbc, &pkt_id)) < 0)
goto fail;
if ((ret = add_tracked_method(rt, name, pkt_id)) < 0)
goto fail;
}
ret = ff_rtmp_packet_write(rt->stream, pkt, rt->out_chunk_size,
&rt->prev_pkt[1], &rt->nb_prev_pkt[1]);
fail:
ff_rtmp_packet_destroy(pkt);
return ret;
}
static int rtmp_write_amf_data(URLContext *s, char *param, uint8_t **p)
{
char *field, *value;
char type;
/* The type must be B for Boolean, N for number, S for string, O for
* object, or Z for null. For Booleans the data must be either 0 or 1 for
* FALSE or TRUE, respectively. Likewise for Objects the data must be
* 0 or 1 to end or begin an object, respectively. Data items in subobjects
* may be named, by prefixing the type with 'N' and specifying the name
* before the value (ie. NB:myFlag:1). This option may be used multiple times
* to construct arbitrary AMF sequences. */
if (param[0] && param[1] == ':') {
type = param[0];
value = param + 2;
} else if (param[0] == 'N' && param[1] && param[2] == ':') {
type = param[1];
field = param + 3;
value = strchr(field, ':');
if (!value)
goto fail;
*value = '\0';
value++;
ff_amf_write_field_name(p, field);
} else {
goto fail;
}
switch (type) {
case 'B':
ff_amf_write_bool(p, value[0] != '0');
break;
case 'S':
ff_amf_write_string(p, value);
break;
case 'N':
ff_amf_write_number(p, strtod(value, NULL));
break;
case 'Z':
ff_amf_write_null(p);
break;
case 'O':
if (value[0] != '0')
ff_amf_write_object_start(p);
else
ff_amf_write_object_end(p);
break;
default:
goto fail;
break;
}
return 0;
fail:
av_log(s, AV_LOG_ERROR, "Invalid AMF parameter: %s\n", param);
return AVERROR(EINVAL);
}
/**
* Generate 'connect' call and send it to the server.
*/
static int gen_connect(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, 4096 + APP_MAX_LENGTH)) < 0)
return ret;
p = pkt.data;
ff_amf_write_string(&p, "connect");
ff_amf_write_number(&p, ++rt->nb_invokes);
ff_amf_write_object_start(&p);
ff_amf_write_field_name(&p, "app");
ff_amf_write_string2(&p, rt->app, rt->auth_params);
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 || rt->swfverify) {
ff_amf_write_field_name(&p, "swfUrl");
if (rt->swfurl)
ff_amf_write_string(&p, rt->swfurl);
else
ff_amf_write_string(&p, rt->swfverify);
}
ff_amf_write_field_name(&p, "tcUrl");
ff_amf_write_string2(&p, rt->tcurl, rt->auth_params);
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) {
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.size = p - pkt.data;
return rtmp_send_packet(rt, &pkt, 1);
}
#define RTMP_CTRL_ABORT_MESSAGE (2)
static int read_connect(URLContext *s, RTMPContext *rt)
{
RTMPPacket pkt = { 0 };
uint8_t *p;
const uint8_t *cp;
int ret;
char command[64];
int stringlen;
double seqnum;
uint8_t tmpstr[256];
GetByteContext gbc;
// handle RTMP Protocol Control Messages
for (;;) {
if ((ret = ff_rtmp_packet_read(rt->stream, &pkt, rt->in_chunk_size,
&rt->prev_pkt[0], &rt->nb_prev_pkt[0])) < 0)
return ret;
#ifdef DEBUG
ff_rtmp_packet_dump(s, &pkt);
#endif
if (pkt.type == RTMP_PT_CHUNK_SIZE) {
if ((ret = handle_chunk_size(s, &pkt)) < 0) {
ff_rtmp_packet_destroy(&pkt);
return ret;
}
} else if (pkt.type == RTMP_CTRL_ABORT_MESSAGE) {
av_log(s, AV_LOG_ERROR, "received abort message\n");
ff_rtmp_packet_destroy(&pkt);
return AVERROR_UNKNOWN;
} else if (pkt.type == RTMP_PT_BYTES_READ) {
av_log(s, AV_LOG_TRACE, "received acknowledgement\n");
} else if (pkt.type == RTMP_PT_WINDOW_ACK_SIZE) {
if ((ret = handle_window_ack_size(s, &pkt)) < 0) {
ff_rtmp_packet_destroy(&pkt);
return ret;
}
} else if (pkt.type == RTMP_PT_SET_PEER_BW) {
if ((ret = handle_set_peer_bw(s, &pkt)) < 0) {
ff_rtmp_packet_destroy(&pkt);
return ret;
}
} else if (pkt.type == RTMP_PT_INVOKE) {
// received RTMP Command Message
break;
} else {
av_log(s, AV_LOG_ERROR, "Unknown control message type (%d)\n", pkt.type);
}
ff_rtmp_packet_destroy(&pkt);
}
cp = pkt.data;
bytestream2_init(&gbc, cp, pkt.size);
if (ff_amf_read_string(&gbc, command, sizeof(command), &stringlen)) {
av_log(s, AV_LOG_ERROR, "Unable to read command string\n");
ff_rtmp_packet_destroy(&pkt);
return AVERROR_INVALIDDATA;
}
if (strcmp(command, "connect")) {
av_log(s, AV_LOG_ERROR, "Expecting connect, got %s\n", command);
ff_rtmp_packet_destroy(&pkt);
return AVERROR_INVALIDDATA;
}
ret = ff_amf_read_number(&gbc, &seqnum);
if (ret)
av_log(s, AV_LOG_WARNING, "SeqNum not found\n");
/* Here one could parse an AMF Object with data as flashVers and others. */
ret = ff_amf_get_field_value(gbc.buffer,
gbc.buffer + bytestream2_get_bytes_left(&gbc),
"app", tmpstr, sizeof(tmpstr));
if (ret)
av_log(s, AV_LOG_WARNING, "App field not found in connect\n");
if (!ret && strcmp(tmpstr, rt->app))
av_log(s, AV_LOG_WARNING, "App field don't match up: %s <-> %s\n",
tmpstr, rt->app);
ff_rtmp_packet_destroy(&pkt);
// Send Window Acknowledgement Size (as defined in specification)
if ((ret = ff_rtmp_packet_create(&pkt, RTMP_NETWORK_CHANNEL,
RTMP_PT_WINDOW_ACK_SIZE, 0, 4)) < 0)
return ret;
p = pkt.data;
// Inform the peer about how often we want acknowledgements about what
// we send. (We don't check for the acknowledgements currently.)
bytestream_put_be32(&p, rt->max_sent_unacked);
pkt.size = p - pkt.data;
ret = ff_rtmp_packet_write(rt->stream, &pkt, rt->out_chunk_size,
&rt->prev_pkt[1], &rt->nb_prev_pkt[1]);
ff_rtmp_packet_destroy(&pkt);
if (ret < 0)
return ret;
// Set Peer Bandwidth
if ((ret = ff_rtmp_packet_create(&pkt, RTMP_NETWORK_CHANNEL,
RTMP_PT_SET_PEER_BW, 0, 5)) < 0)
return ret;
p = pkt.data;
// Tell the peer to only send this many bytes unless it gets acknowledgements.
// This could be any arbitrary value we want here.
bytestream_put_be32(&p, rt->max_sent_unacked);
bytestream_put_byte(&p, 2); // dynamic
pkt.size = p - pkt.data;
ret = ff_rtmp_packet_write(rt->stream, &pkt, rt->out_chunk_size,
&rt->prev_pkt[1], &rt->nb_prev_pkt[1]);
ff_rtmp_packet_destroy(&pkt);
if (ret < 0)
return ret;
// User control
if ((ret = ff_rtmp_packet_create(&pkt, RTMP_NETWORK_CHANNEL,
RTMP_PT_USER_CONTROL, 0, 6)) < 0)
return ret;
p = pkt.data;
bytestream_put_be16(&p, 0); // 0 -> Stream Begin
bytestream_put_be32(&p, 0); // Stream 0
ret = ff_rtmp_packet_write(rt->stream, &pkt, rt->out_chunk_size,
&rt->prev_pkt[1], &rt->nb_prev_pkt[1]);
ff_rtmp_packet_destroy(&pkt);
if (ret < 0)
return ret;
// Chunk size
if ((ret = ff_rtmp_packet_create(&pkt, RTMP_NETWORK_CHANNEL,
RTMP_PT_CHUNK_SIZE, 0, 4)) < 0)
return ret;
p = pkt.data;
bytestream_put_be32(&p, rt->out_chunk_size);
ret = ff_rtmp_packet_write(rt->stream, &pkt, rt->out_chunk_size,
&rt->prev_pkt[1], &rt->nb_prev_pkt[1]);
ff_rtmp_packet_destroy(&pkt);
if (ret < 0)
return ret;
// Send _result NetConnection.Connect.Success to connect
if ((ret = ff_rtmp_packet_create(&pkt, RTMP_SYSTEM_CHANNEL,
RTMP_PT_INVOKE, 0,
RTMP_PKTDATA_DEFAULT_SIZE)) < 0)
return ret;
p = pkt.data;
ff_amf_write_string(&p, "_result");
ff_amf_write_number(&p, seqnum);
ff_amf_write_object_start(&p);
ff_amf_write_field_name(&p, "fmsVer");
ff_amf_write_string(&p, "FMS/3,0,1,123");
ff_amf_write_field_name(&p, "capabilities");
ff_amf_write_number(&p, 31);
ff_amf_write_object_end(&p);
ff_amf_write_object_start(&p);
ff_amf_write_field_name(&p, "level");
ff_amf_write_string(&p, "status");
ff_amf_write_field_name(&p, "code");
ff_amf_write_string(&p, "NetConnection.Connect.Success");
ff_amf_write_field_name(&p, "description");
ff_amf_write_string(&p, "Connection succeeded.");
ff_amf_write_field_name(&p, "objectEncoding");
ff_amf_write_number(&p, 0);
ff_amf_write_object_end(&p);
pkt.size = p - pkt.data;
ret = ff_rtmp_packet_write(rt->stream, &pkt, rt->out_chunk_size,
&rt->prev_pkt[1], &rt->nb_prev_pkt[1]);
ff_rtmp_packet_destroy(&pkt);
if (ret < 0)
return ret;
if ((ret = ff_rtmp_packet_create(&pkt, RTMP_SYSTEM_CHANNEL,
RTMP_PT_INVOKE, 0, 30)) < 0)
return ret;
p = pkt.data;
ff_amf_write_string(&p, "onBWDone");
ff_amf_write_number(&p, 0);
ff_amf_write_null(&p);
ff_amf_write_number(&p, 8192);
pkt.size = p - pkt.data;
ret = ff_rtmp_packet_write(rt->stream, &pkt, rt->out_chunk_size,
&rt->prev_pkt[1], &rt->nb_prev_pkt[1]);
ff_rtmp_packet_destroy(&pkt);
return ret;
}
/**
* Generate 'releaseStream' call and send it to the server. It should make
* the server release some channel for media streams.
*/
static int gen_release_stream(URLContext *s, RTMPContext *rt)
{
RTMPPacket pkt;
uint8_t *p;
int ret;
if ((ret = ff_rtmp_packet_create(&pkt, RTMP_SYSTEM_CHANNEL, RTMP_PT_INVOKE,
0, 29 + strlen(rt->playpath))) < 0)
return ret;
av_log(s, AV_LOG_DEBUG, "Releasing stream...\n");
p = pkt.data;
ff_amf_write_string(&p, "releaseStream");
ff_amf_write_number(&p, ++rt->nb_invokes);
ff_amf_write_null(&p);
ff_amf_write_string(&p, rt->playpath);
return rtmp_send_packet(rt, &pkt, 1);
}
/**
* Generate 'FCPublish' call and send it to the server. It should make
* the server prepare for receiving media streams.
*/
static int gen_fcpublish_stream(URLContext *s, RTMPContext *rt)
{
RTMPPacket pkt;
uint8_t *p;
int ret;
if ((ret = ff_rtmp_packet_create(&pkt, RTMP_SYSTEM_CHANNEL, RTMP_PT_INVOKE,
0, 25 + strlen(rt->playpath))) < 0)
return ret;
av_log(s, AV_LOG_DEBUG, "FCPublish stream...\n");
p = pkt.data;
ff_amf_write_string(&p, "FCPublish");
ff_amf_write_number(&p, ++rt->nb_invokes);
ff_amf_write_null(&p);
ff_amf_write_string(&p, rt->playpath);
return rtmp_send_packet(rt, &pkt, 1);
}
/**
* Generate 'FCUnpublish' call and send it to the server. It should make
* the server destroy stream.
*/
static int gen_fcunpublish_stream(URLContext *s, RTMPContext *rt)
{
RTMPPacket pkt;
uint8_t *p;
int ret;
if ((ret = ff_rtmp_packet_create(&pkt, RTMP_SYSTEM_CHANNEL, RTMP_PT_INVOKE,
0, 27 + strlen(rt->playpath))) < 0)
return ret;
av_log(s, AV_LOG_DEBUG, "UnPublishing stream...\n");
p = pkt.data;
ff_amf_write_string(&p, "FCUnpublish");
ff_amf_write_number(&p, ++rt->nb_invokes);
ff_amf_write_null(&p);
ff_amf_write_string(&p, rt->playpath);
return rtmp_send_packet(rt, &pkt, 0);
}
/**
* Generate 'createStream' call and send it to the server. It should make
* the server allocate some channel for media streams.
*/
static int gen_create_stream(URLContext *s, RTMPContext *rt)
{
RTMPPacket pkt;
uint8_t *p;
int ret;
av_log(s, AV_LOG_DEBUG, "Creating stream...\n");
if ((ret = ff_rtmp_packet_create(&pkt, RTMP_SYSTEM_CHANNEL, RTMP_PT_INVOKE,
0, 25)) < 0)
return ret;
p = pkt.data;
ff_amf_write_string(&p, "createStream");
ff_amf_write_number(&p, ++rt->nb_invokes);
ff_amf_write_null(&p);
return rtmp_send_packet(rt, &pkt, 1);
}
/**
* Generate 'deleteStream' call and send it to the server. It should make
* the server remove some channel for media streams.
*/
static int gen_delete_stream(URLContext *s, RTMPContext *rt)
{
RTMPPacket pkt;
uint8_t *p;
int ret;
av_log(s, AV_LOG_DEBUG, "Deleting stream...\n");
if ((ret = ff_rtmp_packet_create(&pkt, RTMP_SYSTEM_CHANNEL, RTMP_PT_INVOKE,
0, 34)) < 0)
return ret;
p = pkt.data;
ff_amf_write_string(&p, "deleteStream");
ff_amf_write_number(&p, ++rt->nb_invokes);
ff_amf_write_null(&p);
ff_amf_write_number(&p, rt->stream_id);
return rtmp_send_packet(rt, &pkt, 0);
}
/**
* Generate 'getStreamLength' call and send it to the server. If the server
* knows the duration of the selected stream, it will reply with the duration
* in seconds.
*/
static int gen_get_stream_length(URLContext *s, RTMPContext *rt)
{
RTMPPacket pkt;
uint8_t *p;
int ret;
if ((ret = ff_rtmp_packet_create(&pkt, RTMP_SOURCE_CHANNEL, RTMP_PT_INVOKE,
0, 31 + strlen(rt->playpath))) < 0)
return ret;
p = pkt.data;
ff_amf_write_string(&p, "getStreamLength");
ff_amf_write_number(&p, ++rt->nb_invokes);
ff_amf_write_null(&p);
ff_amf_write_string(&p, rt->playpath);
return rtmp_send_packet(rt, &pkt, 1);
}
/**
* Generate client buffer time and send it to the server.
*/
static int gen_buffer_time(URLContext *s, RTMPContext *rt)
{
RTMPPacket pkt;
uint8_t *p;
int ret;
if ((ret = ff_rtmp_packet_create(&pkt, RTMP_NETWORK_CHANNEL, RTMP_PT_USER_CONTROL,
1, 10)) < 0)
return ret;
p = pkt.data;
bytestream_put_be16(&p, 3); // SetBuffer Length
bytestream_put_be32(&p, rt->stream_id);
bytestream_put_be32(&p, rt->client_buffer_time);
return rtmp_send_packet(rt, &pkt, 0);
}
/**
* Generate 'play' call and send it to the server, then ping the server
* to start actual playing.
*/
static int gen_play(URLContext *s, RTMPContext *rt)
{
RTMPPacket pkt;
uint8_t *p;
int ret;
av_log(s, AV_LOG_DEBUG, "Sending play command for '%s'\n", rt->playpath);
if ((ret = ff_rtmp_packet_create(&pkt, RTMP_SOURCE_CHANNEL, RTMP_PT_INVOKE,
0, 29 + strlen(rt->playpath))) < 0)
return ret;
pkt.extra = rt->stream_id;
p = pkt.data;
ff_amf_write_string(&p, "play");
ff_amf_write_number(&p, ++rt->nb_invokes);
ff_amf_write_null(&p);
ff_amf_write_string(&p, rt->playpath);
ff_amf_write_number(&p, rt->live * 1000);
return rtmp_send_packet(rt, &pkt, 1);
}
static int gen_seek(URLContext *s, RTMPContext *rt, int64_t timestamp)
{
RTMPPacket pkt;
uint8_t *p;
int ret;
av_log(s, AV_LOG_DEBUG, "Sending seek command for timestamp %"PRId64"\n",
timestamp);
if ((ret = ff_rtmp_packet_create(&pkt, 3, RTMP_PT_INVOKE, 0, 26)) < 0)
return ret;
pkt.extra = rt->stream_id;
p = pkt.data;
ff_amf_write_string(&p, "seek");
ff_amf_write_number(&p, 0); //no tracking back responses
ff_amf_write_null(&p); //as usual, the first null param
ff_amf_write_number(&p, timestamp); //where we want to jump
return rtmp_send_packet(rt, &pkt, 1);
}
/**
* Generate a pause packet that either pauses or unpauses the current stream.
*/
static int gen_pause(URLContext *s, RTMPContext *rt, int pause, uint32_t timestamp)
{
RTMPPacket pkt;
uint8_t *p;
int ret;
av_log(s, AV_LOG_DEBUG, "Sending pause command for timestamp %d\n",
timestamp);
if ((ret = ff_rtmp_packet_create(&pkt, 3, RTMP_PT_INVOKE, 0, 29)) < 0)
return ret;
pkt.extra = rt->stream_id;
p = pkt.data;
ff_amf_write_string(&p, "pause");
ff_amf_write_number(&p, 0); //no tracking back responses
ff_amf_write_null(&p); //as usual, the first null param
ff_amf_write_bool(&p, pause); // pause or unpause
ff_amf_write_number(&p, timestamp); //where we pause the stream
return rtmp_send_packet(rt, &pkt, 1);
}
/**
* Generate 'publish' call and send it to the server.
*/
static int gen_publish(URLContext *s, RTMPContext *rt)
{
RTMPPacket pkt;
uint8_t *p;
int ret;
av_log(s, AV_LOG_DEBUG, "Sending publish command for '%s'\n", rt->playpath);
if ((ret = ff_rtmp_packet_create(&pkt, RTMP_SOURCE_CHANNEL, RTMP_PT_INVOKE,
0, 30 + strlen(rt->playpath))) < 0)
return ret;
pkt.extra = rt->stream_id;
p = pkt.data;
ff_amf_write_string(&p, "publish");
ff_amf_write_number(&p, ++rt->nb_invokes);
ff_amf_write_null(&p);
ff_amf_write_string(&p, rt->playpath);
ff_amf_write_string(&p, "live");
return rtmp_send_packet(rt, &pkt, 1);
}
/**
* Generate ping reply and send it to the server.
*/
static int gen_pong(URLContext *s, RTMPContext *rt, RTMPPacket *ppkt)
{
RTMPPacket pkt;
uint8_t *p;
int ret;
if (ppkt->size < 6) {
av_log(s, AV_LOG_ERROR, "Too short ping packet (%d)\n",
ppkt->size);
return AVERROR_INVALIDDATA;
}
if ((ret = ff_rtmp_packet_create(&pkt, RTMP_NETWORK_CHANNEL,RTMP_PT_USER_CONTROL,
ppkt->timestamp + 1, 6)) < 0)
return ret;
p = pkt.data;
bytestream_put_be16(&p, 7); // PingResponse
bytestream_put_be32(&p, AV_RB32(ppkt->data+2));
return rtmp_send_packet(rt, &pkt, 0);
}
/**
* Generate SWF verification message and send it to the server.
*/
static int gen_swf_verification(URLContext *s, RTMPContext *rt)
{
RTMPPacket pkt;
uint8_t *p;
int ret;
av_log(s, AV_LOG_DEBUG, "Sending SWF verification...\n");
if ((ret = ff_rtmp_packet_create(&pkt, RTMP_NETWORK_CHANNEL, RTMP_PT_USER_CONTROL,
0, 44)) < 0)
return ret;
p = pkt.data;
bytestream_put_be16(&p, 27);
memcpy(p, rt->swfverification, 42);
return rtmp_send_packet(rt, &pkt, 0);
}
/**
* Generate window acknowledgement size message and send it to the server.
*/
static int gen_window_ack_size(URLContext *s, RTMPContext *rt)
{
RTMPPacket pkt;
uint8_t *p;
int ret;
if ((ret = ff_rtmp_packet_create(&pkt, RTMP_NETWORK_CHANNEL, RTMP_PT_WINDOW_ACK_SIZE,
0, 4)) < 0)
return ret;
p = pkt.data;
bytestream_put_be32(&p, rt->max_sent_unacked);
return rtmp_send_packet(rt, &pkt, 0);
}
/**
* Generate check bandwidth message and send it to the server.
*/
static int gen_check_bw(URLContext *s, RTMPContext *rt)
{
RTMPPacket pkt;
uint8_t *p;
int ret;
if ((ret = ff_rtmp_packet_create(&pkt, RTMP_SYSTEM_CHANNEL, RTMP_PT_INVOKE,
0, 21)) < 0)
return ret;
p = pkt.data;
ff_amf_write_string(&p, "_checkbw");
ff_amf_write_number(&p, ++rt->nb_invokes);
ff_amf_write_null(&p);
return rtmp_send_packet(rt, &pkt, 1);
}
/**
* Generate report on bytes read so far and send it to the server.
*/
static int gen_bytes_read(URLContext *s, RTMPContext *rt, uint32_t ts)
{
RTMPPacket pkt;
uint8_t *p;
int ret;
if ((ret = ff_rtmp_packet_create(&pkt, RTMP_NETWORK_CHANNEL, RTMP_PT_BYTES_READ,
ts, 4)) < 0)
return ret;
p = pkt.data;
bytestream_put_be32(&p, rt->bytes_read);
return rtmp_send_packet(rt, &pkt, 0);
}
static int gen_fcsubscribe_stream(URLContext *s, RTMPContext *rt,
const char *subscribe)
{
RTMPPacket pkt;
uint8_t *p;
int ret;
if ((ret = ff_rtmp_packet_create(&pkt, RTMP_SYSTEM_CHANNEL, RTMP_PT_INVOKE,
0, 27 + strlen(subscribe))) < 0)
return ret;
p = pkt.data;
ff_amf_write_string(&p, "FCSubscribe");
ff_amf_write_number(&p, ++rt->nb_invokes);
ff_amf_write_null(&p);
ff_amf_write_string(&p, subscribe);
return rtmp_send_packet(rt, &pkt, 1);
}
/**
* Put HMAC-SHA2 digest of packet data (except for the bytes where this digest
* will be stored) into that packet.
*
* @param buf handshake data (1536 bytes)
* @param encrypted use an encrypted connection (RTMPE)
* @return offset to the digest inside input data
*/
static int rtmp_handshake_imprint_with_digest(uint8_t *buf, int encrypted)
{
int ret, digest_pos;
if (encrypted)
digest_pos = ff_rtmp_calc_digest_pos(buf, 772, 728, 776);
else
digest_pos = ff_rtmp_calc_digest_pos(buf, 8, 728, 12);
ret = ff_rtmp_calc_digest(buf, RTMP_HANDSHAKE_PACKET_SIZE, digest_pos,
rtmp_player_key, PLAYER_KEY_OPEN_PART_LEN,
buf + digest_pos);
if (ret < 0)
return ret;
return digest_pos;
}
/**
* Verify that the received server response has the expected digest value.
*
* @param buf handshake data received from the server (1536 bytes)
* @param off position to search digest offset from
* @return 0 if digest is valid, digest position otherwise
*/
static int rtmp_validate_digest(uint8_t *buf, int off)
{
uint8_t digest[32];
int ret, digest_pos;
digest_pos = ff_rtmp_calc_digest_pos(buf, off, 728, off + 4);
ret = ff_rtmp_calc_digest(buf, RTMP_HANDSHAKE_PACKET_SIZE, digest_pos,
rtmp_server_key, SERVER_KEY_OPEN_PART_LEN,
digest);
if (ret < 0)
return ret;
if (!memcmp(digest, buf + digest_pos, 32))
return digest_pos;
return 0;
}
static int rtmp_calc_swf_verification(URLContext *s, RTMPContext *rt,
uint8_t *buf)
{
uint8_t *p;
int ret;
if (rt->swfhash_len != 32) {
av_log(s, AV_LOG_ERROR,
"Hash of the decompressed SWF file is not 32 bytes long.\n");
return AVERROR(EINVAL);
}
p = &rt->swfverification[0];
bytestream_put_byte(&p, 1);
bytestream_put_byte(&p, 1);
bytestream_put_be32(&p, rt->swfsize);
bytestream_put_be32(&p, rt->swfsize);
if ((ret = ff_rtmp_calc_digest(rt->swfhash, 32, 0, buf, 32, p)) < 0)
return ret;
return 0;
}
#if CONFIG_ZLIB
static int rtmp_uncompress_swfplayer(uint8_t *in_data, int64_t in_size,
uint8_t **out_data, int64_t *out_size)
{
z_stream zs = { 0 };
void *ptr;
int size;
int ret = 0;
zs.avail_in = in_size;
zs.next_in = in_data;
ret = inflateInit(&zs);
if (ret != Z_OK)
return AVERROR_UNKNOWN;
do {
uint8_t tmp_buf[16384];
zs.avail_out = sizeof(tmp_buf);
zs.next_out = tmp_buf;
ret = inflate(&zs, Z_NO_FLUSH);
if (ret != Z_OK && ret != Z_STREAM_END) {
ret = AVERROR_UNKNOWN;
goto fail;
}
size = sizeof(tmp_buf) - zs.avail_out;
if (!(ptr = av_realloc(*out_data, *out_size + size))) {
ret = AVERROR(ENOMEM);
goto fail;
}
*out_data = ptr;
memcpy(*out_data + *out_size, tmp_buf, size);
*out_size += size;
} while (zs.avail_out == 0);
fail:
inflateEnd(&zs);
return ret;
}
#endif
static int rtmp_calc_swfhash(URLContext *s)
{
RTMPContext *rt = s->priv_data;
uint8_t *in_data = NULL, *out_data = NULL, *swfdata;
int64_t in_size;
URLContext *stream = NULL;
char swfhash[32];
int swfsize;
int ret = 0;
/* Get the SWF player file. */
if ((ret = ffurl_open_whitelist(&stream, rt->swfverify, AVIO_FLAG_READ,
&s->interrupt_callback, NULL,
s->protocol_whitelist, s->protocol_blacklist, s)) < 0) {
av_log(s, AV_LOG_ERROR, "Cannot open connection %s.\n", rt->swfverify);
goto fail;
}
if ((in_size = ffurl_seek(stream, 0, AVSEEK_SIZE)) < 0) {
ret = AVERROR(EIO);
goto fail;
}
if (!(in_data = av_malloc(in_size))) {
ret = AVERROR(ENOMEM);
goto fail;
}
if ((ret = ffurl_read_complete(stream, in_data, in_size)) < 0)
goto fail;
if (in_size < 3) {
ret = AVERROR_INVALIDDATA;
goto fail;
}
if (!memcmp(in_data, "CWS", 3)) {
#if CONFIG_ZLIB
int64_t out_size;
/* Decompress the SWF player file using Zlib. */
if (!(out_data = av_malloc(8))) {
ret = AVERROR(ENOMEM);
goto fail;
}
*in_data = 'F'; // magic stuff
memcpy(out_data, in_data, 8);
out_size = 8;
if ((ret = rtmp_uncompress_swfplayer(in_data + 8, in_size - 8,
&out_data, &out_size)) < 0)
goto fail;
swfsize = out_size;
swfdata = out_data;
#else
av_log(s, AV_LOG_ERROR,
"Zlib is required for decompressing the SWF player file.\n");
ret = AVERROR(EINVAL);
goto fail;
#endif
} else {
swfsize = in_size;
swfdata = in_data;
}
/* Compute the SHA256 hash of the SWF player file. */
if ((ret = ff_rtmp_calc_digest(swfdata, swfsize, 0,
"Genuine Adobe Flash Player 001", 30,
swfhash)) < 0)
goto fail;
/* Set SWFVerification parameters. */
av_opt_set_bin(rt, "rtmp_swfhash", swfhash, 32, 0);
rt->swfsize = swfsize;
fail:
av_freep(&in_data);
av_freep(&out_data);
ffurl_close(stream);
return ret;
}
/**
* Perform handshake with the server by means of exchanging pseudorandom data
* signed with HMAC-SHA2 digest.
*
* @return 0 if handshake succeeds, negative value otherwise
*/
static int rtmp_handshake(URLContext *s, RTMPContext *rt)
{
AVLFG rnd;
uint8_t tosend [RTMP_HANDSHAKE_PACKET_SIZE+1] = {
3, // unencrypted data
0, 0, 0, 0, // client uptime
RTMP_CLIENT_VER1,
RTMP_CLIENT_VER2,
RTMP_CLIENT_VER3,
RTMP_CLIENT_VER4,
};
uint8_t clientdata[RTMP_HANDSHAKE_PACKET_SIZE];
uint8_t serverdata[RTMP_HANDSHAKE_PACKET_SIZE+1];
int i;
int server_pos, client_pos;
uint8_t digest[32], signature[32];
int ret, type = 0;
av_log(s, AV_LOG_DEBUG, "Handshaking...\n");
av_lfg_init(&rnd, 0xDEADC0DE);
// generate handshake packet - 1536 bytes of pseudorandom data
for (i = 9; i <= RTMP_HANDSHAKE_PACKET_SIZE; i++)
tosend[i] = av_lfg_get(&rnd) >> 24;
if (CONFIG_FFRTMPCRYPT_PROTOCOL && rt->encrypted) {
/* When the client wants to use RTMPE, we have to change the command
* byte to 0x06 which means to use encrypted data and we have to set
* the flash version to at least 9.0.115.0. */
tosend[0] = 6;
tosend[5] = 128;
tosend[6] = 0;
tosend[7] = 3;
tosend[8] = 2;
/* Initialize the Diffie-Hellmann context and generate the public key
* to send to the server. */
if ((ret = ff_rtmpe_gen_pub_key(rt->stream, tosend + 1)) < 0)
return ret;
}
client_pos = rtmp_handshake_imprint_with_digest(tosend + 1, rt->encrypted);
if (client_pos < 0)
return client_pos;
if ((ret = ffurl_write(rt->stream, tosend,
RTMP_HANDSHAKE_PACKET_SIZE + 1)) < 0) {
av_log(s, AV_LOG_ERROR, "Cannot write RTMP handshake request\n");
return ret;
}
if ((ret = ffurl_read_complete(rt->stream, serverdata,
RTMP_HANDSHAKE_PACKET_SIZE + 1)) < 0) {
av_log(s, AV_LOG_ERROR, "Cannot read RTMP handshake response\n");
return ret;
}
if ((ret = ffurl_read_complete(rt->stream, clientdata,
RTMP_HANDSHAKE_PACKET_SIZE)) < 0) {
av_log(s, AV_LOG_ERROR, "Cannot read RTMP handshake response\n");
return ret;
}
av_log(s, AV_LOG_DEBUG, "Type answer %d\n", serverdata[0]);
av_log(s, AV_LOG_DEBUG, "Server version %d.%d.%d.%d\n",
serverdata[5], serverdata[6], serverdata[7], serverdata[8]);
if (rt->is_input && serverdata[5] >= 3) {
server_pos = rtmp_validate_digest(serverdata + 1, 772);
if (server_pos < 0)
return server_pos;
if (!server_pos) {
type = 1;
server_pos = rtmp_validate_digest(serverdata + 1, 8);
if (server_pos < 0)
return server_pos;
if (!server_pos) {
av_log(s, AV_LOG_ERROR, "Server response validating failed\n");
return AVERROR(EIO);
}
}
/* Generate SWFVerification token (SHA256 HMAC hash of decompressed SWF,
* key are the last 32 bytes of the server handshake. */
if (rt->swfsize) {
if ((ret = rtmp_calc_swf_verification(s, rt, serverdata + 1 +
RTMP_HANDSHAKE_PACKET_SIZE - 32)) < 0)
return ret;
}
ret = ff_rtmp_calc_digest(tosend + 1 + client_pos, 32, 0,
rtmp_server_key, sizeof(rtmp_server_key),
digest);
if (ret < 0)
return ret;
ret = ff_rtmp_calc_digest(clientdata, RTMP_HANDSHAKE_PACKET_SIZE - 32,
0, digest, 32, signature);
if (ret < 0)
return ret;
if (CONFIG_FFRTMPCRYPT_PROTOCOL && rt->encrypted) {
/* Compute the shared secret key sent by the server and initialize
* the RC4 encryption. */
if ((ret = ff_rtmpe_compute_secret_key(rt->stream, serverdata + 1,
tosend + 1, type)) < 0)
return ret;
/* Encrypt the signature received by the server. */
ff_rtmpe_encrypt_sig(rt->stream, signature, digest, serverdata[0]);
}
if (memcmp(signature, clientdata + RTMP_HANDSHAKE_PACKET_SIZE - 32, 32)) {
av_log(s, AV_LOG_ERROR, "Signature mismatch\n");
return AVERROR(EIO);
}
for (i = 0; i < RTMP_HANDSHAKE_PACKET_SIZE; i++)
tosend[i] = av_lfg_get(&rnd) >> 24;
ret = ff_rtmp_calc_digest(serverdata + 1 + server_pos, 32, 0,
rtmp_player_key, sizeof(rtmp_player_key),
digest);
if (ret < 0)
return ret;
ret = ff_rtmp_calc_digest(tosend, RTMP_HANDSHAKE_PACKET_SIZE - 32, 0,
digest, 32,
tosend + RTMP_HANDSHAKE_PACKET_SIZE - 32);
if (ret < 0)
return ret;
if (CONFIG_FFRTMPCRYPT_PROTOCOL && rt->encrypted) {
/* Encrypt the signature to be send to the server. */
ff_rtmpe_encrypt_sig(rt->stream, tosend +
RTMP_HANDSHAKE_PACKET_SIZE - 32, digest,
serverdata[0]);
}
// write reply back to the server
if ((ret = ffurl_write(rt->stream, tosend,
RTMP_HANDSHAKE_PACKET_SIZE)) < 0)
return ret;
if (CONFIG_FFRTMPCRYPT_PROTOCOL && rt->encrypted) {
/* Set RC4 keys for encryption and update the keystreams. */
if ((ret = ff_rtmpe_update_keystream(rt->stream)) < 0)
return ret;
}
} else {
if (CONFIG_FFRTMPCRYPT_PROTOCOL && rt->encrypted) {
/* Compute the shared secret key sent by the server and initialize
* the RC4 encryption. */
if ((ret = ff_rtmpe_compute_secret_key(rt->stream, serverdata + 1,
tosend + 1, 1)) < 0)
return ret;
if (serverdata[0] == 9) {
/* Encrypt the signature received by the server. */
ff_rtmpe_encrypt_sig(rt->stream, signature, digest,
serverdata[0]);
}
}
if ((ret = ffurl_write(rt->stream, serverdata + 1,
RTMP_HANDSHAKE_PACKET_SIZE)) < 0)
return ret;
if (CONFIG_FFRTMPCRYPT_PROTOCOL && rt->encrypted) {
/* Set RC4 keys for encryption and update the keystreams. */
if ((ret = ff_rtmpe_update_keystream(rt->stream)) < 0)
return ret;
}
}
return 0;
}
static int rtmp_receive_hs_packet(RTMPContext* rt, uint32_t *first_int,
uint32_t *second_int, char *arraydata,
int size)
{
int inoutsize;
inoutsize = ffurl_read_complete(rt->stream, arraydata,
RTMP_HANDSHAKE_PACKET_SIZE);
if (inoutsize <= 0)
return AVERROR(EIO);
if (inoutsize != RTMP_HANDSHAKE_PACKET_SIZE) {
av_log(rt, AV_LOG_ERROR, "Erroneous Message size %d"
" not following standard\n", (int)inoutsize);
return AVERROR(EINVAL);
}
*first_int = AV_RB32(arraydata);
*second_int = AV_RB32(arraydata + 4);
return 0;
}
static int rtmp_send_hs_packet(RTMPContext* rt, uint32_t first_int,
uint32_t second_int, char *arraydata, int size)
{
int inoutsize;
AV_WB32(arraydata, first_int);
AV_WB32(arraydata + 4, second_int);
inoutsize = ffurl_write(rt->stream, arraydata,
RTMP_HANDSHAKE_PACKET_SIZE);
if (inoutsize != RTMP_HANDSHAKE_PACKET_SIZE) {
av_log(rt, AV_LOG_ERROR, "Unable to write answer\n");
return AVERROR(EIO);
}
return 0;
}
/**
* rtmp handshake server side
*/
static int rtmp_server_handshake(URLContext *s, RTMPContext *rt)
{
uint8_t buffer[RTMP_HANDSHAKE_PACKET_SIZE];
uint32_t hs_epoch;
uint32_t hs_my_epoch;
uint8_t hs_c1[RTMP_HANDSHAKE_PACKET_SIZE];
uint8_t hs_s1[RTMP_HANDSHAKE_PACKET_SIZE];
uint32_t zeroes;
uint32_t temp = 0;
int randomidx = 0;
int inoutsize = 0;
int ret;
inoutsize = ffurl_read_complete(rt->stream, buffer, 1); // Receive C0
if (inoutsize <= 0) {
av_log(s, AV_LOG_ERROR, "Unable to read handshake\n");
return AVERROR(EIO);
}
// Check Version
if (buffer[0] != 3) {
av_log(s, AV_LOG_ERROR, "RTMP protocol version mismatch\n");
return AVERROR(EIO);
}
if (ffurl_write(rt->stream, buffer, 1) <= 0) { // Send S0
av_log(s, AV_LOG_ERROR,
"Unable to write answer - RTMP S0\n");
return AVERROR(EIO);
}
/* Receive C1 */
ret = rtmp_receive_hs_packet(rt, &hs_epoch, &zeroes, hs_c1,
RTMP_HANDSHAKE_PACKET_SIZE);
if (ret) {
av_log(s, AV_LOG_ERROR, "RTMP Handshake C1 Error\n");
return ret;
}
/* Send S1 */
/* By now same epoch will be sent */
hs_my_epoch = hs_epoch;
/* Generate random */
for (randomidx = 8; randomidx < (RTMP_HANDSHAKE_PACKET_SIZE);
randomidx += 4)
AV_WB32(hs_s1 + randomidx, av_get_random_seed());
ret = rtmp_send_hs_packet(rt, hs_my_epoch, 0, hs_s1,
RTMP_HANDSHAKE_PACKET_SIZE);
if (ret) {
av_log(s, AV_LOG_ERROR, "RTMP Handshake S1 Error\n");
return ret;
}
/* Send S2 */
ret = rtmp_send_hs_packet(rt, hs_epoch, 0, hs_c1,
RTMP_HANDSHAKE_PACKET_SIZE);
if (ret) {
av_log(s, AV_LOG_ERROR, "RTMP Handshake S2 Error\n");
return ret;
}
/* Receive C2 */
ret = rtmp_receive_hs_packet(rt, &temp, &zeroes, buffer,
RTMP_HANDSHAKE_PACKET_SIZE);
if (ret) {
av_log(s, AV_LOG_ERROR, "RTMP Handshake C2 Error\n");
return ret;
}
if (temp != hs_my_epoch)
av_log(s, AV_LOG_WARNING,
"Erroneous C2 Message epoch does not match up with C1 epoch\n");
if (memcmp(buffer + 8, hs_s1 + 8,
RTMP_HANDSHAKE_PACKET_SIZE - 8))
av_log(s, AV_LOG_WARNING,
"Erroneous C2 Message random does not match up\n");
return 0;
}
static int handle_chunk_size(URLContext *s, RTMPPacket *pkt)
{
RTMPContext *rt = s->priv_data;
int ret;
if (pkt->size < 4) {
av_log(s, AV_LOG_ERROR,
"Too short chunk size change packet (%d)\n",
pkt->size);
return AVERROR_INVALIDDATA;
}
if (!rt->is_input) {
/* Send the same chunk size change packet back to the server,
* setting the outgoing chunk size to the same as the incoming one. */
if ((ret = ff_rtmp_packet_write(rt->stream, pkt, rt->out_chunk_size,
&rt->prev_pkt[1], &rt->nb_prev_pkt[1])) < 0)
return ret;
rt->out_chunk_size = AV_RB32(pkt->data);
}
rt->in_chunk_size = AV_RB32(pkt->data);
if (rt->in_chunk_size <= 0) {
av_log(s, AV_LOG_ERROR, "Incorrect chunk size %d\n",
rt->in_chunk_size);
return AVERROR_INVALIDDATA;
}
av_log(s, AV_LOG_DEBUG, "New incoming chunk size = %d\n",
rt->in_chunk_size);
return 0;
}
static int handle_user_control(URLContext *s, RTMPPacket *pkt)
{
RTMPContext *rt = s->priv_data;
int t, ret;
if (pkt->size < 2) {
av_log(s, AV_LOG_ERROR, "Too short user control packet (%d)\n",
pkt->size);
return AVERROR_INVALIDDATA;
}
t = AV_RB16(pkt->data);
if (t == 6) { // PingRequest
if ((ret = gen_pong(s, rt, pkt)) < 0)
return ret;
} else if (t == 26) {
if (rt->swfsize) {
if ((ret = gen_swf_verification(s, rt)) < 0)
return ret;
} else {
av_log(s, AV_LOG_WARNING, "Ignoring SWFVerification request.\n");
}
}
return 0;
}
static int handle_set_peer_bw(URLContext *s, RTMPPacket *pkt)
{
RTMPContext *rt = s->priv_data;
if (pkt->size < 4) {
av_log(s, AV_LOG_ERROR,
"Peer bandwidth packet is less than 4 bytes long (%d)\n",
pkt->size);
return AVERROR_INVALIDDATA;
}
// We currently don't check how much the peer has acknowledged of
// what we have sent. To do that properly, we should call
// gen_window_ack_size here, to tell the peer that we want an
// acknowledgement with (at least) that interval.
rt->max_sent_unacked = AV_RB32(pkt->data);
if (rt->max_sent_unacked <= 0) {
av_log(s, AV_LOG_ERROR, "Incorrect set peer bandwidth %d\n",
rt->max_sent_unacked);
return AVERROR_INVALIDDATA;
}
av_log(s, AV_LOG_DEBUG, "Max sent, unacked = %d\n", rt->max_sent_unacked);
return 0;
}
static int handle_window_ack_size(URLContext *s, RTMPPacket *pkt)
{
RTMPContext *rt = s->priv_data;
if (pkt->size < 4) {
av_log(s, AV_LOG_ERROR,
"Too short window acknowledgement size packet (%d)\n",
pkt->size);
return AVERROR_INVALIDDATA;
}
rt->receive_report_size = AV_RB32(pkt->data);
if (rt->receive_report_size <= 0) {
av_log(s, AV_LOG_ERROR, "Incorrect window acknowledgement size %d\n",
rt->receive_report_size);
return AVERROR_INVALIDDATA;
}
av_log(s, AV_LOG_DEBUG, "Window acknowledgement size = %d\n", rt->receive_report_size);
// Send an Acknowledgement packet after receiving half the maximum
// size, to make sure the peer can keep on sending without waiting
// for acknowledgements.
rt->receive_report_size >>= 1;
return 0;
}
static int do_adobe_auth(RTMPContext *rt, const char *user, const char *salt,
const char *opaque, const char *challenge)
{
uint8_t hash[16];
char hashstr[AV_BASE64_SIZE(sizeof(hash))], challenge2[10];
struct AVMD5 *md5 = av_md5_alloc();
if (!md5)
return AVERROR(ENOMEM);
snprintf(challenge2, sizeof(challenge2), "%08x", av_get_random_seed());
av_md5_init(md5);
av_md5_update(md5, user, strlen(user));
av_md5_update(md5, salt, strlen(salt));
av_md5_update(md5, rt->password, strlen(rt->password));
av_md5_final(md5, hash);
av_base64_encode(hashstr, sizeof(hashstr), hash,
sizeof(hash));
av_md5_init(md5);
av_md5_update(md5, hashstr, strlen(hashstr));
if (opaque)
av_md5_update(md5, opaque, strlen(opaque));
else if (challenge)
av_md5_update(md5, challenge, strlen(challenge));
av_md5_update(md5, challenge2, strlen(challenge2));
av_md5_final(md5, hash);
av_base64_encode(hashstr, sizeof(hashstr), hash,
sizeof(hash));
snprintf(rt->auth_params, sizeof(rt->auth_params),
"?authmod=%s&user=%s&challenge=%s&response=%s",
"adobe", user, challenge2, hashstr);
if (opaque)
av_strlcatf(rt->auth_params, sizeof(rt->auth_params),
"&opaque=%s", opaque);
av_free(md5);
return 0;
}
static int do_llnw_auth(RTMPContext *rt, const char *user, const char *nonce)
{
uint8_t hash[16];
char hashstr1[33], hashstr2[33];
const char *realm = "live";
const char *method = "publish";
const char *qop = "auth";
const char *nc = "00000001";
char cnonce[10];
struct AVMD5 *md5 = av_md5_alloc();
if (!md5)
return AVERROR(ENOMEM);
snprintf(cnonce, sizeof(cnonce), "%08x", av_get_random_seed());
av_md5_init(md5);
av_md5_update(md5, user, strlen(user));
av_md5_update(md5, ":", 1);
av_md5_update(md5, realm, strlen(realm));
av_md5_update(md5, ":", 1);
av_md5_update(md5, rt->password, strlen(rt->password));
av_md5_final(md5, hash);
ff_data_to_hex(hashstr1, hash, 16, 1);
hashstr1[32] = '\0';
av_md5_init(md5);
av_md5_update(md5, method, strlen(method));
av_md5_update(md5, ":/", 2);
av_md5_update(md5, rt->app, strlen(rt->app));
if (!strchr(rt->app, '/'))
av_md5_update(md5, "/_definst_", strlen("/_definst_"));
av_md5_final(md5, hash);
ff_data_to_hex(hashstr2, hash, 16, 1);
hashstr2[32] = '\0';
av_md5_init(md5);
av_md5_update(md5, hashstr1, strlen(hashstr1));
av_md5_update(md5, ":", 1);
if (nonce)
av_md5_update(md5, nonce, strlen(nonce));
av_md5_update(md5, ":", 1);
av_md5_update(md5, nc, strlen(nc));
av_md5_update(md5, ":", 1);
av_md5_update(md5, cnonce, strlen(cnonce));
av_md5_update(md5, ":", 1);
av_md5_update(md5, qop, strlen(qop));
av_md5_update(md5, ":", 1);
av_md5_update(md5, hashstr2, strlen(hashstr2));
av_md5_final(md5, hash);
ff_data_to_hex(hashstr1, hash, 16, 1);
snprintf(rt->auth_params, sizeof(rt->auth_params),
"?authmod=%s&user=%s&nonce=%s&cnonce=%s&nc=%s&response=%s",
"llnw", user, nonce, cnonce, nc, hashstr1);
av_free(md5);
return 0;
}
static int handle_connect_error(URLContext *s, const char *desc)
{
RTMPContext *rt = s->priv_data;
char buf[300], *ptr, authmod[15];
int i = 0, ret = 0;
const char *user = "", *salt = "", *opaque = NULL,
*challenge = NULL, *cptr = NULL, *nonce = NULL;
if (!(cptr = strstr(desc, "authmod=adobe")) &&
!(cptr = strstr(desc, "authmod=llnw"))) {
av_log(s, AV_LOG_ERROR,
"Unknown connect error (unsupported authentication method?)\n");
return AVERROR_UNKNOWN;
}
cptr += strlen("authmod=");
while (*cptr && *cptr != ' ' && i < sizeof(authmod) - 1)
authmod[i++] = *cptr++;
authmod[i] = '\0';
if (!rt->username[0] || !rt->password[0]) {
av_log(s, AV_LOG_ERROR, "No credentials set\n");
return AVERROR_UNKNOWN;
}
if (strstr(desc, "?reason=authfailed")) {
av_log(s, AV_LOG_ERROR, "Incorrect username/password\n");
return AVERROR_UNKNOWN;
} else if (strstr(desc, "?reason=nosuchuser")) {
av_log(s, AV_LOG_ERROR, "Incorrect username\n");
return AVERROR_UNKNOWN;
}
if (rt->auth_tried) {
av_log(s, AV_LOG_ERROR, "Authentication failed\n");
return AVERROR_UNKNOWN;
}
rt->auth_params[0] = '\0';
if (strstr(desc, "code=403 need auth")) {
snprintf(rt->auth_params, sizeof(rt->auth_params),
"?authmod=%s&user=%s", authmod, rt->username);
return 0;
}
if (!(cptr = strstr(desc, "?reason=needauth"))) {
av_log(s, AV_LOG_ERROR, "No auth parameters found\n");
return AVERROR_UNKNOWN;
}
av_strlcpy(buf, cptr + 1, sizeof(buf));
ptr = buf;
while (ptr) {
char *next = strchr(ptr, '&');
char *value = strchr(ptr, '=');
if (next)
*next++ = '\0';
if (value) {
*value++ = '\0';
if (!strcmp(ptr, "user")) {
user = value;
} else if (!strcmp(ptr, "salt")) {
salt = value;
} else if (!strcmp(ptr, "opaque")) {
opaque = value;
} else if (!strcmp(ptr, "challenge")) {
challenge = value;
} else if (!strcmp(ptr, "nonce")) {
nonce = value;
} else {
av_log(s, AV_LOG_INFO, "Ignoring unsupported var %s\n", ptr);
}
} else {
av_log(s, AV_LOG_WARNING, "Variable %s has NULL value\n", ptr);
}
ptr = next;
}
if (!strcmp(authmod, "adobe")) {
if ((ret = do_adobe_auth(rt, user, salt, opaque, challenge)) < 0)
return ret;
} else {
if ((ret = do_llnw_auth(rt, user, nonce)) < 0)
return ret;
}
rt->auth_tried = 1;
return 0;
}
static int handle_invoke_error(URLContext *s, RTMPPacket *pkt)
{
RTMPContext *rt = s->priv_data;
const uint8_t *data_end = pkt->data + pkt->size;
char *tracked_method = NULL;
int level = AV_LOG_ERROR;
uint8_t tmpstr[256];
int ret;
if ((ret = find_tracked_method(s, pkt, 9, &tracked_method)) < 0)
return ret;
if (!ff_amf_get_field_value(pkt->data + 9, data_end,
"description", tmpstr, sizeof(tmpstr))) {
if (tracked_method && (!strcmp(tracked_method, "_checkbw") ||
!strcmp(tracked_method, "releaseStream") ||
!strcmp(tracked_method, "FCSubscribe") ||
!strcmp(tracked_method, "FCPublish"))) {
/* Gracefully ignore Adobe-specific historical artifact errors. */
level = AV_LOG_WARNING;
ret = 0;
} else if (tracked_method && !strcmp(tracked_method, "getStreamLength")) {
level = rt->live ? AV_LOG_DEBUG : AV_LOG_WARNING;
ret = 0;
} else if (tracked_method && !strcmp(tracked_method, "connect")) {
ret = handle_connect_error(s, tmpstr);
if (!ret) {
rt->do_reconnect = 1;
level = AV_LOG_VERBOSE;
}
} else
ret = AVERROR_UNKNOWN;
av_log(s, level, "Server error: %s\n", tmpstr);
}
av_free(tracked_method);
return ret;
}
static int write_begin(URLContext *s)
{
RTMPContext *rt = s->priv_data;
PutByteContext pbc;
RTMPPacket spkt = { 0 };
int ret;
// Send Stream Begin 1
if ((ret = ff_rtmp_packet_create(&spkt, RTMP_NETWORK_CHANNEL,
RTMP_PT_USER_CONTROL, 0, 6)) < 0) {
av_log(s, AV_LOG_ERROR, "Unable to create response packet\n");
return ret;
}
bytestream2_init_writer(&pbc, spkt.data, spkt.size);
bytestream2_put_be16(&pbc, 0); // 0 -> Stream Begin
bytestream2_put_be32(&pbc, rt->nb_streamid);
ret = ff_rtmp_packet_write(rt->stream, &spkt, rt->out_chunk_size,
&rt->prev_pkt[1], &rt->nb_prev_pkt[1]);
ff_rtmp_packet_destroy(&spkt);
return ret;
}
static int write_status(URLContext *s, RTMPPacket *pkt,
const char *status, const char *filename)
{
RTMPContext *rt = s->priv_data;
RTMPPacket spkt = { 0 };
char statusmsg[128];
uint8_t *pp;
int ret;
if ((ret = ff_rtmp_packet_create(&spkt, RTMP_SYSTEM_CHANNEL,
RTMP_PT_INVOKE, 0,
RTMP_PKTDATA_DEFAULT_SIZE)) < 0) {
av_log(s, AV_LOG_ERROR, "Unable to create response packet\n");
return ret;
}
pp = spkt.data;
spkt.extra = pkt->extra;
ff_amf_write_string(&pp, "onStatus");
ff_amf_write_number(&pp, 0);
ff_amf_write_null(&pp);
ff_amf_write_object_start(&pp);
ff_amf_write_field_name(&pp, "level");
ff_amf_write_string(&pp, "status");
ff_amf_write_field_name(&pp, "code");
ff_amf_write_string(&pp, status);
ff_amf_write_field_name(&pp, "description");
snprintf(statusmsg, sizeof(statusmsg),
"%s is now published", filename);
ff_amf_write_string(&pp, statusmsg);
ff_amf_write_field_name(&pp, "details");
ff_amf_write_string(&pp, filename);
ff_amf_write_object_end(&pp);
spkt.size = pp - spkt.data;
ret = ff_rtmp_packet_write(rt->stream, &spkt, rt->out_chunk_size,
&rt->prev_pkt[1], &rt->nb_prev_pkt[1]);
ff_rtmp_packet_destroy(&spkt);
return ret;
}
static int send_invoke_response(URLContext *s, RTMPPacket *pkt)
{
RTMPContext *rt = s->priv_data;
double seqnum;
char filename[128];
char command[64];
int stringlen;
char *pchar;
const uint8_t *p = pkt->data;
uint8_t *pp = NULL;
RTMPPacket spkt = { 0 };
GetByteContext gbc;
int ret;
bytestream2_init(&gbc, p, pkt->size);
if (ff_amf_read_string(&gbc, command, sizeof(command),
&stringlen)) {
av_log(s, AV_LOG_ERROR, "Error in PT_INVOKE\n");
return AVERROR_INVALIDDATA;
}
ret = ff_amf_read_number(&gbc, &seqnum);
if (ret)
return ret;
ret = ff_amf_read_null(&gbc);
if (ret)
return ret;
if (!strcmp(command, "FCPublish") ||
!strcmp(command, "publish")) {
ret = ff_amf_read_string(&gbc, filename,
sizeof(filename), &stringlen);
if (ret) {
if (ret == AVERROR(EINVAL))
av_log(s, AV_LOG_ERROR, "Unable to parse stream name - name too long?\n");
else
av_log(s, AV_LOG_ERROR, "Unable to parse stream name\n");
return ret;
}
// check with url
if (s->filename) {
pchar = strrchr(s->filename, '/');
if (!pchar) {
av_log(s, AV_LOG_WARNING,
"Unable to find / in url %s, bad format\n",
s->filename);
pchar = s->filename;
}
pchar++;
if (strcmp(pchar, filename))
av_log(s, AV_LOG_WARNING, "Unexpected stream %s, expecting"
" %s\n", filename, pchar);
}
rt->state = STATE_RECEIVING;
}
if (!strcmp(command, "FCPublish")) {
if ((ret = ff_rtmp_packet_create(&spkt, RTMP_SYSTEM_CHANNEL,
RTMP_PT_INVOKE, 0,
RTMP_PKTDATA_DEFAULT_SIZE)) < 0) {
av_log(s, AV_LOG_ERROR, "Unable to create response packet\n");
return ret;
}
pp = spkt.data;
ff_amf_write_string(&pp, "onFCPublish");
} else if (!strcmp(command, "publish")) {
ret = write_begin(s);
if (ret < 0)
return ret;
// Send onStatus(NetStream.Publish.Start)
return write_status(s, pkt, "NetStream.Publish.Start",
filename);
} else if (!strcmp(command, "play")) {
ret = write_begin(s);
if (ret < 0)
return ret;
rt->state = STATE_SENDING;
return write_status(s, pkt, "NetStream.Play.Start",
filename);
} else {
if ((ret = ff_rtmp_packet_create(&spkt, RTMP_SYSTEM_CHANNEL,
RTMP_PT_INVOKE, 0,
RTMP_PKTDATA_DEFAULT_SIZE)) < 0) {
av_log(s, AV_LOG_ERROR, "Unable to create response packet\n");
return ret;
}
pp = spkt.data;
ff_amf_write_string(&pp, "_result");
ff_amf_write_number(&pp, seqnum);
ff_amf_write_null(&pp);
if (!strcmp(command, "createStream")) {
rt->nb_streamid++;
if (rt->nb_streamid == 0 || rt->nb_streamid == 2)
rt->nb_streamid++; /* Values 0 and 2 are reserved */
ff_amf_write_number(&pp, rt->nb_streamid);
/* By now we don't control which streams are removed in
* deleteStream. There is no stream creation control
* if a client creates more than 2^32 - 2 streams. */
}
}
spkt.size = pp - spkt.data;
ret = ff_rtmp_packet_write(rt->stream, &spkt, rt->out_chunk_size,
&rt->prev_pkt[1], &rt->nb_prev_pkt[1]);
ff_rtmp_packet_destroy(&spkt);
return ret;
}
/**
* Read the AMF_NUMBER response ("_result") to a function call
* (e.g. createStream()). This response should be made up of the AMF_STRING
* "result", a NULL object and then the response encoded as AMF_NUMBER. On a
* successful response, we will return set the value to number (otherwise number
* will not be changed).
*
* @return 0 if reading the value succeeds, negative value otherwise
*/
static int read_number_result(RTMPPacket *pkt, double *number)
{
// We only need to fit "_result" in this.
uint8_t strbuffer[8];
int stringlen;
double numbuffer;
GetByteContext gbc;
bytestream2_init(&gbc, pkt->data, pkt->size);
// Value 1/4: "_result" as AMF_STRING
if (ff_amf_read_string(&gbc, strbuffer, sizeof(strbuffer), &stringlen))
return AVERROR_INVALIDDATA;
if (strcmp(strbuffer, "_result"))
return AVERROR_INVALIDDATA;
// Value 2/4: The callee reference number
if (ff_amf_read_number(&gbc, &numbuffer))
return AVERROR_INVALIDDATA;
// Value 3/4: Null
if (ff_amf_read_null(&gbc))
return AVERROR_INVALIDDATA;
// Value 4/4: The response as AMF_NUMBER
if (ff_amf_read_number(&gbc, &numbuffer))
return AVERROR_INVALIDDATA;
else
*number = numbuffer;
return 0;
}
static int handle_invoke_result(URLContext *s, RTMPPacket *pkt)
{
RTMPContext *rt = s->priv_data;
char *tracked_method = NULL;
int ret = 0;
if ((ret = find_tracked_method(s, pkt, 10, &tracked_method)) < 0)
return ret;
if (!tracked_method) {
/* Ignore this reply when the current method is not tracked. */
return ret;
}
if (!strcmp(tracked_method, "connect")) {
if (!rt->is_input) {
if ((ret = gen_release_stream(s, rt)) < 0)
goto fail;
if ((ret = gen_fcpublish_stream(s, rt)) < 0)
goto fail;
} else {
if ((ret = gen_window_ack_size(s, rt)) < 0)
goto fail;
}
if ((ret = gen_create_stream(s, rt)) < 0)
goto fail;
if (rt->is_input) {
/* Send the FCSubscribe command when the name of live
* stream is defined by the user or if it's a live stream. */
if (rt->subscribe) {
if ((ret = gen_fcsubscribe_stream(s, rt, rt->subscribe)) < 0)
goto fail;
} else if (rt->live == -1) {
if ((ret = gen_fcsubscribe_stream(s, rt, rt->playpath)) < 0)
goto fail;
}
}
} else if (!strcmp(tracked_method, "createStream")) {
double stream_id;
if (read_number_result(pkt, &stream_id)) {
av_log(s, AV_LOG_WARNING, "Unexpected reply on connect()\n");
} else {
rt->stream_id = stream_id;
}
if (!rt->is_input) {
if ((ret = gen_publish(s, rt)) < 0)
goto fail;
} else {
if (rt->live != -1) {
if ((ret = gen_get_stream_length(s, rt)) < 0)
goto fail;
}
if ((ret = gen_play(s, rt)) < 0)
goto fail;
if ((ret = gen_buffer_time(s, rt)) < 0)
goto fail;
}
} else if (!strcmp(tracked_method, "getStreamLength")) {
if (read_number_result(pkt, &rt->duration)) {
av_log(s, AV_LOG_WARNING, "Unexpected reply on getStreamLength()\n");
}
}
fail:
av_free(tracked_method);
return ret;
}
static int handle_invoke_status(URLContext *s, RTMPPacket *pkt)
{
RTMPContext *rt = s->priv_data;
const uint8_t *data_end = pkt->data + pkt->size;
const uint8_t *ptr = pkt->data + RTMP_HEADER;
uint8_t tmpstr[256];
int i, t;
for (i = 0; i < 2; i++) {
t = ff_amf_tag_size(ptr, data_end);
if (t < 0)
return 1;
ptr += t;
}
t = ff_amf_get_field_value(ptr, data_end, "level", tmpstr, sizeof(tmpstr));
if (!t && !strcmp(tmpstr, "error")) {
t = ff_amf_get_field_value(ptr, data_end,
"description", tmpstr, sizeof(tmpstr));
if (t || !tmpstr[0])
t = ff_amf_get_field_value(ptr, data_end, "code",
tmpstr, sizeof(tmpstr));
if (!t)
av_log(s, AV_LOG_ERROR, "Server error: %s\n", tmpstr);
return -1;
}
t = ff_amf_get_field_value(ptr, data_end, "code", tmpstr, sizeof(tmpstr));
if (!t && !strcmp(tmpstr, "NetStream.Play.Start")) rt->state = STATE_PLAYING;
if (!t && !strcmp(tmpstr, "NetStream.Play.Stop")) rt->state = STATE_STOPPED;
if (!t && !strcmp(tmpstr, "NetStream.Play.UnpublishNotify")) rt->state = STATE_STOPPED;
if (!t && !strcmp(tmpstr, "NetStream.Publish.Start")) rt->state = STATE_PUBLISHING;
if (!t && !strcmp(tmpstr, "NetStream.Seek.Notify")) rt->state = STATE_PLAYING;
return 0;
}
static int handle_invoke(URLContext *s, RTMPPacket *pkt)
{
RTMPContext *rt = s->priv_data;
int ret = 0;
//TODO: check for the messages sent for wrong state?
if (ff_amf_match_string(pkt->data, pkt->size, "_error")) {
if ((ret = handle_invoke_error(s, pkt)) < 0)
return ret;
} else if (ff_amf_match_string(pkt->data, pkt->size, "_result")) {
if ((ret = handle_invoke_result(s, pkt)) < 0)
return ret;
} else if (ff_amf_match_string(pkt->data, pkt->size, "onStatus")) {
if ((ret = handle_invoke_status(s, pkt)) < 0)
return ret;
} else if (ff_amf_match_string(pkt->data, pkt->size, "onBWDone")) {
if ((ret = gen_check_bw(s, rt)) < 0)
return ret;
} else if (ff_amf_match_string(pkt->data, pkt->size, "releaseStream") ||
ff_amf_match_string(pkt->data, pkt->size, "FCPublish") ||
ff_amf_match_string(pkt->data, pkt->size, "publish") ||
ff_amf_match_string(pkt->data, pkt->size, "play") ||
ff_amf_match_string(pkt->data, pkt->size, "_checkbw") ||
ff_amf_match_string(pkt->data, pkt->size, "createStream")) {
if ((ret = send_invoke_response(s, pkt)) < 0)
return ret;
}
return ret;
}
static int update_offset(RTMPContext *rt, int size)
{
int old_flv_size;
// generate packet header and put data into buffer for FLV demuxer
if (rt->flv_off < rt->flv_size) {
// There is old unread data in the buffer, thus append at the end
old_flv_size = rt->flv_size;
rt->flv_size += size;
} else {
// All data has been read, write the new data at the start of the buffer
old_flv_size = 0;
rt->flv_size = size;
rt->flv_off = 0;
}
return old_flv_size;
}
static int append_flv_data(RTMPContext *rt, RTMPPacket *pkt, int skip)
{
int old_flv_size, ret;
PutByteContext pbc;
const uint8_t *data = pkt->data + skip;
const int size = pkt->size - skip;
uint32_t ts = pkt->timestamp;
if (pkt->type == RTMP_PT_AUDIO) {
rt->has_audio = 1;
} else if (pkt->type == RTMP_PT_VIDEO) {
rt->has_video = 1;
}
old_flv_size = update_offset(rt, size + 15);
if ((ret = av_reallocp(&rt->flv_data, rt->flv_size)) < 0) {
rt->flv_size = rt->flv_off = 0;
return ret;
}
bytestream2_init_writer(&pbc, rt->flv_data, rt->flv_size);
bytestream2_skip_p(&pbc, old_flv_size);
bytestream2_put_byte(&pbc, pkt->type);
bytestream2_put_be24(&pbc, size);
bytestream2_put_be24(&pbc, ts);
bytestream2_put_byte(&pbc, ts >> 24);
bytestream2_put_be24(&pbc, 0);
bytestream2_put_buffer(&pbc, data, size);
bytestream2_put_be32(&pbc, size + RTMP_HEADER);
return 0;
}
static int handle_notify(URLContext *s, RTMPPacket *pkt)
{
RTMPContext *rt = s->priv_data;
uint8_t commandbuffer[64];
char statusmsg[128];
int stringlen, ret, skip = 0;
GetByteContext gbc;
bytestream2_init(&gbc, pkt->data, pkt->size);
if (ff_amf_read_string(&gbc, commandbuffer, sizeof(commandbuffer),
&stringlen))
return AVERROR_INVALIDDATA;
if (!strcmp(commandbuffer, "onMetaData")) {
// metadata properties should be stored in a mixed array
if (bytestream2_get_byte(&gbc) == AMF_DATA_TYPE_MIXEDARRAY) {
// We have found a metaData Array so flv can determine the streams
// from this.
rt->received_metadata = 1;
// skip 32-bit max array index
bytestream2_skip(&gbc, 4);
while (bytestream2_get_bytes_left(&gbc) > 3) {
if (ff_amf_get_string(&gbc, statusmsg, sizeof(statusmsg),
&stringlen))
return AVERROR_INVALIDDATA;
// We do not care about the content of the property (yet).
stringlen = ff_amf_tag_size(gbc.buffer, gbc.buffer_end);
if (stringlen < 0)
return AVERROR_INVALIDDATA;
bytestream2_skip(&gbc, stringlen);
// The presence of the following properties indicates that the
// respective streams are present.
if (!strcmp(statusmsg, "videocodecid")) {
rt->has_video = 1;
}
if (!strcmp(statusmsg, "audiocodecid")) {
rt->has_audio = 1;
}
}
if (bytestream2_get_be24(&gbc) != AMF_END_OF_OBJECT)
return AVERROR_INVALIDDATA;
}
}
// Skip the @setDataFrame string and validate it is a notification
if (!strcmp(commandbuffer, "@setDataFrame")) {
skip = gbc.buffer - pkt->data;
ret = ff_amf_read_string(&gbc, statusmsg,
sizeof(statusmsg), &stringlen);
if (ret < 0)
return AVERROR_INVALIDDATA;
}
return append_flv_data(rt, pkt, skip);
}
/**
* Parse received packet and possibly perform some action depending on
* the packet contents.
* @return 0 for no errors, negative values for serious errors which prevent
* further communications, positive values for uncritical errors
*/
static int rtmp_parse_result(URLContext *s, RTMPContext *rt, RTMPPacket *pkt)
{
int ret;
#ifdef DEBUG
ff_rtmp_packet_dump(s, pkt);
#endif
switch (pkt->type) {
case RTMP_PT_BYTES_READ:
av_log(s, AV_LOG_TRACE, "received bytes read report\n");
break;
case RTMP_PT_CHUNK_SIZE:
if ((ret = handle_chunk_size(s, pkt)) < 0)
return ret;
break;
case RTMP_PT_USER_CONTROL:
if ((ret = handle_user_control(s, pkt)) < 0)
return ret;
break;
case RTMP_PT_SET_PEER_BW:
if ((ret = handle_set_peer_bw(s, pkt)) < 0)
return ret;
break;
case RTMP_PT_WINDOW_ACK_SIZE:
if ((ret = handle_window_ack_size(s, pkt)) < 0)
return ret;
break;
case RTMP_PT_INVOKE:
if ((ret = handle_invoke(s, pkt)) < 0)
return ret;
break;
case RTMP_PT_VIDEO:
case RTMP_PT_AUDIO:
case RTMP_PT_METADATA:
case RTMP_PT_NOTIFY:
/* Audio, Video and Metadata packets are parsed in get_packet() */
break;
default:
av_log(s, AV_LOG_VERBOSE, "Unknown packet type received 0x%02X\n", pkt->type);
break;
}
return 0;
}
static int handle_metadata(RTMPContext *rt, RTMPPacket *pkt)
{
int ret, old_flv_size, type;
const uint8_t *next;
uint8_t *p;
uint32_t size;
uint32_t ts, cts, pts = 0;
old_flv_size = update_offset(rt, pkt->size);
if ((ret = av_reallocp(&rt->flv_data, rt->flv_size)) < 0) {
rt->flv_size = rt->flv_off = 0;
return ret;
}
next = pkt->data;
p = rt->flv_data + old_flv_size;
/* copy data while rewriting timestamps */
ts = pkt->timestamp;
while (next - pkt->data < pkt->size - RTMP_HEADER) {
type = bytestream_get_byte(&next);
size = bytestream_get_be24(&next);
cts = bytestream_get_be24(&next);
cts |= bytestream_get_byte(&next) << 24;
if (!pts)
pts = cts;
ts += cts - pts;
pts = cts;
if (size + 3 + 4 > pkt->data + pkt->size - next)
break;
bytestream_put_byte(&p, type);
bytestream_put_be24(&p, size);
bytestream_put_be24(&p, ts);
bytestream_put_byte(&p, ts >> 24);
memcpy(p, next, size + 3 + 4);
p += size + 3;
bytestream_put_be32(&p, size + RTMP_HEADER);
next += size + 3 + 4;
}
if (p != rt->flv_data + rt->flv_size) {
av_log(rt, AV_LOG_WARNING, "Incomplete flv packets in "
"RTMP_PT_METADATA packet\n");
rt->flv_size = p - rt->flv_data;
}
return 0;
}
/**
* Interact with the server by receiving and sending RTMP packets until
* there is some significant data (media data or expected status notification).
*
* @param s reading context
* @param for_header non-zero value tells function to work until it
* gets notification from the server that playing has been started,
* otherwise function will work until some media data is received (or
* an error happens)
* @return 0 for successful operation, negative value in case of error
*/
static int get_packet(URLContext *s, int for_header)
{
RTMPContext *rt = s->priv_data;
int ret;
if (rt->state == STATE_STOPPED)
return AVERROR_EOF;
for (;;) {
RTMPPacket rpkt = { 0 };
if ((ret = ff_rtmp_packet_read(rt->stream, &rpkt,
rt->in_chunk_size, &rt->prev_pkt[0],
&rt->nb_prev_pkt[0])) <= 0) {
if (ret == 0) {
return AVERROR(EAGAIN);
} else {
return AVERROR(EIO);
}
}
// Track timestamp for later use
rt->last_timestamp = rpkt.timestamp;
rt->bytes_read += ret;
if (rt->bytes_read - rt->last_bytes_read > rt->receive_report_size) {
av_log(s, AV_LOG_DEBUG, "Sending bytes read report\n");
if ((ret = gen_bytes_read(s, rt, rpkt.timestamp + 1)) < 0) {
ff_rtmp_packet_destroy(&rpkt);
return ret;
}
rt->last_bytes_read = rt->bytes_read;
}
ret = rtmp_parse_result(s, rt, &rpkt);
// At this point we must check if we are in the seek state and continue
// with the next packet. handle_invoke will get us out of this state
// when the right message is encountered
if (rt->state == STATE_SEEKING) {
ff_rtmp_packet_destroy(&rpkt);
// We continue, let the natural flow of things happen:
// AVERROR(EAGAIN) or handle_invoke gets us out of here
continue;
}
if (ret < 0) {//serious error in current packet
ff_rtmp_packet_destroy(&rpkt);
return ret;
}
if (rt->do_reconnect && for_header) {
ff_rtmp_packet_destroy(&rpkt);
return 0;
}
if (rt->state == STATE_STOPPED) {
ff_rtmp_packet_destroy(&rpkt);
return AVERROR_EOF;
}
if (for_header && (rt->state == STATE_PLAYING ||
rt->state == STATE_PUBLISHING ||
rt->state == STATE_SENDING ||
rt->state == STATE_RECEIVING)) {
ff_rtmp_packet_destroy(&rpkt);
return 0;
}
if (!rpkt.size || !rt->is_input) {
ff_rtmp_packet_destroy(&rpkt);
continue;
}
if (rpkt.type == RTMP_PT_VIDEO || rpkt.type == RTMP_PT_AUDIO) {
ret = append_flv_data(rt, &rpkt, 0);
ff_rtmp_packet_destroy(&rpkt);
return ret;
} else if (rpkt.type == RTMP_PT_NOTIFY) {
ret = handle_notify(s, &rpkt);
ff_rtmp_packet_destroy(&rpkt);
return ret;
} else if (rpkt.type == RTMP_PT_METADATA) {
ret = handle_metadata(rt, &rpkt);
ff_rtmp_packet_destroy(&rpkt);
return ret;
}
ff_rtmp_packet_destroy(&rpkt);
}
}
static int rtmp_close(URLContext *h)
{
RTMPContext *rt = h->priv_data;
int ret = 0, i, j;
if (!rt->is_input) {
rt->flv_data = NULL;
if (rt->out_pkt.size)
ff_rtmp_packet_destroy(&rt->out_pkt);
if (rt->state > STATE_FCPUBLISH)
ret = gen_fcunpublish_stream(h, rt);
}
if (rt->state > STATE_HANDSHAKED)
ret = gen_delete_stream(h, rt);
for (i = 0; i < 2; i++) {
for (j = 0; j < rt->nb_prev_pkt[i]; j++)
ff_rtmp_packet_destroy(&rt->prev_pkt[i][j]);
av_freep(&rt->prev_pkt[i]);
}
free_tracked_methods(rt);
av_freep(&rt->flv_data);
ffurl_closep(&rt->stream);
return ret;
}
/**
* Insert a fake onMetadata packet into the FLV stream to notify the FLV
* demuxer about the duration of the stream.
*
* This should only be done if there was no real onMetadata packet sent by the
* server at the start of the stream and if we were able to retrieve a valid
* duration via a getStreamLength call.
*
* @return 0 for successful operation, negative value in case of error
*/
static int inject_fake_duration_metadata(RTMPContext *rt)
{
// We need to insert the metadata packet directly after the FLV
// header, i.e. we need to move all other already read data by the
// size of our fake metadata packet.
uint8_t* p;
// Keep old flv_data pointer
uint8_t* old_flv_data = rt->flv_data;
// Allocate a new flv_data pointer with enough space for the additional package
if (!(rt->flv_data = av_malloc(rt->flv_size + 55))) {
rt->flv_data = old_flv_data;
return AVERROR(ENOMEM);
}
// Copy FLV header
memcpy(rt->flv_data, old_flv_data, 13);
// Copy remaining packets
memcpy(rt->flv_data + 13 + 55, old_flv_data + 13, rt->flv_size - 13);
// Increase the size by the injected packet
rt->flv_size += 55;
// Delete the old FLV data
av_freep(&old_flv_data);
p = rt->flv_data + 13;
bytestream_put_byte(&p, FLV_TAG_TYPE_META);
bytestream_put_be24(&p, 40); // size of data part (sum of all parts below)
bytestream_put_be24(&p, 0); // timestamp
bytestream_put_be32(&p, 0); // reserved
// first event name as a string
bytestream_put_byte(&p, AMF_DATA_TYPE_STRING);
// "onMetaData" as AMF string
bytestream_put_be16(&p, 10);
bytestream_put_buffer(&p, "onMetaData", 10);
// mixed array (hash) with size and string/type/data tuples
bytestream_put_byte(&p, AMF_DATA_TYPE_MIXEDARRAY);
bytestream_put_be32(&p, 1); // metadata_count
// "duration" as AMF string
bytestream_put_be16(&p, 8);
bytestream_put_buffer(&p, "duration", 8);
bytestream_put_byte(&p, AMF_DATA_TYPE_NUMBER);
bytestream_put_be64(&p, av_double2int(rt->duration));
// Finalise object
bytestream_put_be16(&p, 0); // Empty string
bytestream_put_byte(&p, AMF_END_OF_OBJECT);
bytestream_put_be32(&p, 40 + RTMP_HEADER); // size of data part (sum of all parts above)
return 0;
}
/**
* Open RTMP connection and verify that the stream can be played.
*
* URL syntax: rtmp://server[:port][/app][/playpath]
* where 'app' is first one or two directories in the path
* (e.g. /ondemand/, /flash/live/, etc.)
* and 'playpath' is a file name (the rest of the path,
* may be prefixed with "mp4:")
*/
static int rtmp_open(URLContext *s, const char *uri, int flags, AVDictionary **opts)
{
RTMPContext *rt = s->priv_data;
char proto[8], hostname[256], path[1024], auth[100], *fname;
char *old_app, *qmark, *n, fname_buffer[1024];
uint8_t buf[2048];
int port;
int ret;
if (rt->listen_timeout > 0)
rt->listen = 1;
rt->is_input = !(flags & AVIO_FLAG_WRITE);
av_url_split(proto, sizeof(proto), auth, sizeof(auth),
hostname, sizeof(hostname), &port,
path, sizeof(path), s->filename);
n = strchr(path, ' ');
if (n) {
av_log(s, AV_LOG_WARNING,
"Detected librtmp style URL parameters, these aren't supported "
"by the libavformat internal RTMP handler currently enabled. "
"See the documentation for the correct way to pass parameters.\n");
*n = '\0'; // Trim not supported part
}
if (auth[0]) {
char *ptr = strchr(auth, ':');
if (ptr) {
*ptr = '\0';
av_strlcpy(rt->username, auth, sizeof(rt->username));
av_strlcpy(rt->password, ptr + 1, sizeof(rt->password));
}
}
if (rt->listen && strcmp(proto, "rtmp")) {
av_log(s, AV_LOG_ERROR, "rtmp_listen not available for %s\n",
proto);
return AVERROR(EINVAL);
}
if (!strcmp(proto, "rtmpt") || !strcmp(proto, "rtmpts")) {
if (!strcmp(proto, "rtmpts"))
av_dict_set(opts, "ffrtmphttp_tls", "1", 1);
/* open the http tunneling connection */
ff_url_join(buf, sizeof(buf), "ffrtmphttp", NULL, hostname, port, NULL);
} else if (!strcmp(proto, "rtmps")) {
/* open the tls connection */
if (port < 0)
port = RTMPS_DEFAULT_PORT;
ff_url_join(buf, sizeof(buf), "tls", NULL, hostname, port, NULL);
} else if (!strcmp(proto, "rtmpe") || (!strcmp(proto, "rtmpte"))) {
if (!strcmp(proto, "rtmpte"))
av_dict_set(opts, "ffrtmpcrypt_tunneling", "1", 1);
/* open the encrypted connection */
ff_url_join(buf, sizeof(buf), "ffrtmpcrypt", NULL, hostname, port, NULL);
rt->encrypted = 1;
} else {
/* open the tcp connection */
if (port < 0)
port = RTMP_DEFAULT_PORT;
if (rt->listen)
ff_url_join(buf, sizeof(buf), "tcp", NULL, hostname, port,
"?listen&listen_timeout=%d",
rt->listen_timeout * 1000);
else
ff_url_join(buf, sizeof(buf), "tcp", NULL, hostname, port, NULL);
}
reconnect:
if ((ret = ffurl_open_whitelist(&rt->stream, buf, AVIO_FLAG_READ_WRITE,
&s->interrupt_callback, opts,
s->protocol_whitelist, s->protocol_blacklist, s)) < 0) {
av_log(s , AV_LOG_ERROR, "Cannot open connection %s\n", buf);
goto fail;
}
if (rt->swfverify) {
if ((ret = rtmp_calc_swfhash(s)) < 0)
goto fail;
}
rt->state = STATE_START;
if (!rt->listen && (ret = rtmp_handshake(s, rt)) < 0)
goto fail;
if (rt->listen && (ret = rtmp_server_handshake(s, rt)) < 0)
goto fail;
rt->out_chunk_size = 128;
rt->in_chunk_size = 128; // Probably overwritten later
rt->state = STATE_HANDSHAKED;
// Keep the application name when it has been defined by the user.
old_app = rt->app;
rt->app = av_malloc(APP_MAX_LENGTH);
if (!rt->app) {
ret = AVERROR(ENOMEM);
goto fail;
}
//extract "app" part from path
qmark = strchr(path, '?');
if (qmark && strstr(qmark, "slist=")) {
char* amp;
// After slist we have the playpath, the full path is used as app
av_strlcpy(rt->app, path + 1, APP_MAX_LENGTH);
fname = strstr(path, "slist=") + 6;
// Strip any further query parameters from fname
amp = strchr(fname, '&');
if (amp) {
av_strlcpy(fname_buffer, fname, FFMIN(amp - fname + 1,
sizeof(fname_buffer)));
fname = fname_buffer;
}
} else if (!strncmp(path, "/ondemand/", 10)) {
fname = path + 10;
memcpy(rt->app, "ondemand", 9);
} else {
char *next = *path ? path + 1 : path;
char *p = strchr(next, '/');
if (!p) {
if (old_app) {
// If name of application has been defined by the user, assume that
// playpath is provided in the URL
fname = next;
} else {
fname = NULL;
av_strlcpy(rt->app, next, APP_MAX_LENGTH);
}
} else {
// make sure we do not mismatch a playpath for an application instance
char *c = strchr(p + 1, ':');
fname = strchr(p + 1, '/');
if (!fname || (c && c < fname)) {
fname = p + 1;
av_strlcpy(rt->app, path + 1, FFMIN(p - path, APP_MAX_LENGTH));
} else {
fname++;
av_strlcpy(rt->app, path + 1, FFMIN(fname - path - 1, APP_MAX_LENGTH));
}
}
}
if (old_app) {
// The name of application has been defined by the user, override it.
if (strlen(old_app) >= APP_MAX_LENGTH) {
ret = AVERROR(EINVAL);
goto fail;
}
av_free(rt->app);
rt->app = old_app;
}
if (!rt->playpath) {
int max_len = 1;
if (fname)
max_len = strlen(fname) + 5; // add prefix "mp4:"
rt->playpath = av_malloc(max_len);
if (!rt->playpath) {
ret = AVERROR(ENOMEM);
goto fail;
}
if (fname) {
int len = strlen(fname);
if (!strchr(fname, ':') && len >= 4 &&
(!strcmp(fname + len - 4, ".f4v") ||
!strcmp(fname + len - 4, ".mp4"))) {
memcpy(rt->playpath, "mp4:", 5);
} else {
if (len >= 4 && !strcmp(fname + len - 4, ".flv"))
fname[len - 4] = '\0';
rt->playpath[0] = 0;
}
av_strlcat(rt->playpath, fname, max_len);
} else {
rt->playpath[0] = '\0';
}
}
if (!rt->tcurl) {
rt->tcurl = av_malloc(TCURL_MAX_LENGTH);
if (!rt->tcurl) {
ret = AVERROR(ENOMEM);
goto fail;
}
ff_url_join(rt->tcurl, TCURL_MAX_LENGTH, proto, NULL, hostname,
port, "/%s", rt->app);
}
if (!rt->flashver) {
rt->flashver = av_malloc(FLASHVER_MAX_LENGTH);
if (!rt->flashver) {
ret = AVERROR(ENOMEM);
goto fail;
}
if (rt->is_input) {
snprintf(rt->flashver, FLASHVER_MAX_LENGTH, "%s %d,%d,%d,%d",
RTMP_CLIENT_PLATFORM, RTMP_CLIENT_VER1, RTMP_CLIENT_VER2,
RTMP_CLIENT_VER3, RTMP_CLIENT_VER4);
} else {
snprintf(rt->flashver, FLASHVER_MAX_LENGTH,
"FMLE/3.0 (compatible; %s)", LIBAVFORMAT_IDENT);
}
}
rt->receive_report_size = 1048576;
rt->bytes_read = 0;
rt->has_audio = 0;
rt->has_video = 0;
rt->received_metadata = 0;
rt->last_bytes_read = 0;
rt->max_sent_unacked = 2500000;
rt->duration = 0;
av_log(s, AV_LOG_DEBUG, "Proto = %s, path = %s, app = %s, fname = %s\n",
proto, path, rt->app, rt->playpath);
if (!rt->listen) {
if ((ret = gen_connect(s, rt)) < 0)
goto fail;
} else {
if ((ret = read_connect(s, s->priv_data)) < 0)
goto fail;
}
do {
ret = get_packet(s, 1);
} while (ret == AVERROR(EAGAIN));
if (ret < 0)
goto fail;
if (rt->do_reconnect) {
int i;
ffurl_closep(&rt->stream);
rt->do_reconnect = 0;
rt->nb_invokes = 0;
for (i = 0; i < 2; i++)
memset(rt->prev_pkt[i], 0,
sizeof(**rt->prev_pkt) * rt->nb_prev_pkt[i]);
free_tracked_methods(rt);
goto reconnect;
}
if (rt->is_input) {
// generate FLV header for demuxer
rt->flv_size = 13;
if ((ret = av_reallocp(&rt->flv_data, rt->flv_size)) < 0)
goto fail;
rt->flv_off = 0;
memcpy(rt->flv_data, "FLV\1\0\0\0\0\011\0\0\0\0", rt->flv_size);
// Read packets until we reach the first A/V packet or read metadata.
// If there was a metadata package in front of the A/V packets, we can
// build the FLV header from this. If we do not receive any metadata,
// the FLV decoder will allocate the needed streams when their first
// audio or video packet arrives.
while (!rt->has_audio && !rt->has_video && !rt->received_metadata) {
if ((ret = get_packet(s, 0)) < 0)
goto fail;
}
// Either after we have read the metadata or (if there is none) the
// first packet of an A/V stream, we have a better knowledge about the
// streams, so set the FLV header accordingly.
if (rt->has_audio) {
rt->flv_data[4] |= FLV_HEADER_FLAG_HASAUDIO;
}
if (rt->has_video) {
rt->flv_data[4] |= FLV_HEADER_FLAG_HASVIDEO;
}
// If we received the first packet of an A/V stream and no metadata but
// the server returned a valid duration, create a fake metadata packet
// to inform the FLV decoder about the duration.
if (!rt->received_metadata && rt->duration > 0) {
if ((ret = inject_fake_duration_metadata(rt)) < 0)
goto fail;
}
} else {
rt->flv_size = 0;
rt->flv_data = NULL;
rt->flv_off = 0;
rt->skip_bytes = 13;
}
s->max_packet_size = rt->stream->max_packet_size;
s->is_streamed = 1;
return 0;
fail:
av_freep(&rt->playpath);
av_freep(&rt->tcurl);
av_freep(&rt->flashver);
av_dict_free(opts);
rtmp_close(s);
return ret;
}
static int rtmp_read(URLContext *s, uint8_t *buf, int size)
{
RTMPContext *rt = s->priv_data;
int orig_size = size;
int ret;
while (size > 0) {
int data_left = rt->flv_size - rt->flv_off;
if (data_left >= size) {
memcpy(buf, rt->flv_data + rt->flv_off, size);
rt->flv_off += size;
return orig_size;
}
if (data_left > 0) {
memcpy(buf, rt->flv_data + rt->flv_off, data_left);
buf += data_left;
size -= data_left;
rt->flv_off = rt->flv_size;
return data_left;
}
if ((ret = get_packet(s, 0)) < 0)
return ret;
}
return orig_size;
}
static int64_t rtmp_seek(URLContext *s, int stream_index, int64_t timestamp,
int flags)
{
RTMPContext *rt = s->priv_data;
int ret;
av_log(s, AV_LOG_DEBUG,
"Seek on stream index %d at timestamp %"PRId64" with flags %08x\n",
stream_index, timestamp, flags);
if ((ret = gen_seek(s, rt, timestamp)) < 0) {
av_log(s, AV_LOG_ERROR,
"Unable to send seek command on stream index %d at timestamp "
"%"PRId64" with flags %08x\n",
stream_index, timestamp, flags);
return ret;
}
rt->flv_off = rt->flv_size;
rt->state = STATE_SEEKING;
return timestamp;
}
static int rtmp_pause(URLContext *s, int pause)
{
RTMPContext *rt = s->priv_data;
int ret;
av_log(s, AV_LOG_DEBUG, "Pause at timestamp %d\n",
rt->last_timestamp);
if ((ret = gen_pause(s, rt, pause, rt->last_timestamp)) < 0) {
av_log(s, AV_LOG_ERROR, "Unable to send pause command at timestamp %d\n",
rt->last_timestamp);
return ret;
}
return 0;
}
static int rtmp_write(URLContext *s, const uint8_t *buf, int size)
{
RTMPContext *rt = s->priv_data;
int size_temp = size;
int pktsize, pkttype, copy;
uint32_t ts;
const uint8_t *buf_temp = buf;
uint8_t c;
int ret;
do {
if (rt->skip_bytes) {
int skip = FFMIN(rt->skip_bytes, size_temp);
buf_temp += skip;
size_temp -= skip;
rt->skip_bytes -= skip;
continue;
}
if (rt->flv_header_bytes < RTMP_HEADER) {
const uint8_t *header = rt->flv_header;
int channel = RTMP_AUDIO_CHANNEL;
copy = FFMIN(RTMP_HEADER - rt->flv_header_bytes, size_temp);
bytestream_get_buffer(&buf_temp, rt->flv_header + rt->flv_header_bytes, copy);
rt->flv_header_bytes += copy;
size_temp -= copy;
if (rt->flv_header_bytes < RTMP_HEADER)
break;
pkttype = bytestream_get_byte(&header);
pktsize = bytestream_get_be24(&header);
ts = bytestream_get_be24(&header);
ts |= bytestream_get_byte(&header) << 24;
bytestream_get_be24(&header);
rt->flv_size = pktsize;
if (pkttype == RTMP_PT_VIDEO)
channel = RTMP_VIDEO_CHANNEL;
if (((pkttype == RTMP_PT_VIDEO || pkttype == RTMP_PT_AUDIO) && ts == 0) ||
pkttype == RTMP_PT_NOTIFY) {
if ((ret = ff_rtmp_check_alloc_array(&rt->prev_pkt[1],
&rt->nb_prev_pkt[1],
channel)) < 0)
return ret;
// Force sending a full 12 bytes header by clearing the
// channel id, to make it not match a potential earlier
// packet in the same channel.
rt->prev_pkt[1][channel].channel_id = 0;
}
//this can be a big packet, it's better to send it right here
if ((ret = ff_rtmp_packet_create(&rt->out_pkt, channel,
pkttype, ts, pktsize)) < 0)
return ret;
rt->out_pkt.extra = rt->stream_id;
rt->flv_data = rt->out_pkt.data;
}
copy = FFMIN(rt->flv_size - rt->flv_off, size_temp);
bytestream_get_buffer(&buf_temp, rt->flv_data + rt->flv_off, copy);
rt->flv_off += copy;
size_temp -= copy;
if (rt->flv_off == rt->flv_size) {
rt->skip_bytes = 4;
if (rt->out_pkt.type == RTMP_PT_NOTIFY) {
// For onMetaData and |RtmpSampleAccess packets, we want
// @setDataFrame prepended to the packet before it gets sent.
// However, not all RTMP_PT_NOTIFY packets (e.g., onTextData
// and onCuePoint).
uint8_t commandbuffer[64];
int stringlen = 0;
GetByteContext gbc;
bytestream2_init(&gbc, rt->flv_data, rt->flv_size);
if (!ff_amf_read_string(&gbc, commandbuffer, sizeof(commandbuffer),
&stringlen)) {
if (!strcmp(commandbuffer, "onMetaData") ||
!strcmp(commandbuffer, "|RtmpSampleAccess")) {
uint8_t *ptr;
if ((ret = av_reallocp(&rt->out_pkt.data, rt->out_pkt.size + 16)) < 0) {
rt->flv_size = rt->flv_off = rt->flv_header_bytes = 0;
return ret;
}
memmove(rt->out_pkt.data + 16, rt->out_pkt.data, rt->out_pkt.size);
rt->out_pkt.size += 16;
ptr = rt->out_pkt.data;
ff_amf_write_string(&ptr, "@setDataFrame");
}
}
}
if ((ret = rtmp_send_packet(rt, &rt->out_pkt, 0)) < 0)
return ret;
rt->flv_size = 0;
rt->flv_off = 0;
rt->flv_header_bytes = 0;
rt->flv_nb_packets++;
}
} while (buf_temp - buf < size);
if (rt->flv_nb_packets < rt->flush_interval)
return size;
rt->flv_nb_packets = 0;
/* set stream into nonblocking mode */
rt->stream->flags |= AVIO_FLAG_NONBLOCK;
/* try to read one byte from the stream */
ret = ffurl_read(rt->stream, &c, 1);
/* switch the stream back into blocking mode */
rt->stream->flags &= ~AVIO_FLAG_NONBLOCK;
if (ret == AVERROR(EAGAIN)) {
/* no incoming data to handle */
return size;
} else if (ret < 0) {
return ret;
} else if (ret == 1) {
RTMPPacket rpkt = { 0 };
if ((ret = ff_rtmp_packet_read_internal(rt->stream, &rpkt,
rt->in_chunk_size,
&rt->prev_pkt[0],
&rt->nb_prev_pkt[0], c)) <= 0)
return ret;
if ((ret = rtmp_parse_result(s, rt, &rpkt)) < 0)
return ret;
ff_rtmp_packet_destroy(&rpkt);
}
return size;
}
#define OFFSET(x) offsetof(RTMPContext, x)
#define DEC AV_OPT_FLAG_DECODING_PARAM
#define ENC AV_OPT_FLAG_ENCODING_PARAM
static const AVOption rtmp_options[] = {
{"rtmp_app", "Name of application to connect to on the RTMP server", OFFSET(app), AV_OPT_TYPE_STRING, {.str = NULL }, 0, 0, DEC|ENC},
{"rtmp_buffer", "Set buffer time in milliseconds. The default is 3000.", OFFSET(client_buffer_time), AV_OPT_TYPE_INT, {.i64 = 3000}, 0, INT_MAX, DEC|ENC},
{"rtmp_conn", "Append arbitrary AMF data to the Connect message", OFFSET(conn), AV_OPT_TYPE_STRING, {.str = NULL }, 0, 0, DEC|ENC},
{"rtmp_flashver", "Version of the Flash plugin used to run the SWF player.", OFFSET(flashver), AV_OPT_TYPE_STRING, {.str = NULL }, 0, 0, DEC|ENC},
{"rtmp_flush_interval", "Number of packets flushed in the same request (RTMPT only).", OFFSET(flush_interval), AV_OPT_TYPE_INT, {.i64 = 10}, 0, INT_MAX, ENC},
{"rtmp_live", "Specify that the media is a live stream.", OFFSET(live), AV_OPT_TYPE_INT, {.i64 = -2}, INT_MIN, INT_MAX, DEC, "rtmp_live"},
{"any", "both", 0, AV_OPT_TYPE_CONST, {.i64 = -2}, 0, 0, DEC, "rtmp_live"},
{"live", "live stream", 0, AV_OPT_TYPE_CONST, {.i64 = -1}, 0, 0, DEC, "rtmp_live"},
{"recorded", "recorded stream", 0, AV_OPT_TYPE_CONST, {.i64 = 0}, 0, 0, DEC, "rtmp_live"},
{"rtmp_pageurl", "URL of the web page in which the media was embedded. By default no value will be sent.", OFFSET(pageurl), AV_OPT_TYPE_STRING, {.str = NULL }, 0, 0, DEC},
{"rtmp_playpath", "Stream identifier to play or to publish", OFFSET(playpath), AV_OPT_TYPE_STRING, {.str = NULL }, 0, 0, DEC|ENC},
{"rtmp_subscribe", "Name of live stream to subscribe to. Defaults to rtmp_playpath.", OFFSET(subscribe), AV_OPT_TYPE_STRING, {.str = NULL }, 0, 0, DEC},
{"rtmp_swfhash", "SHA256 hash of the decompressed SWF file (32 bytes).", OFFSET(swfhash), AV_OPT_TYPE_BINARY, .flags = DEC},
{"rtmp_swfsize", "Size of the decompressed SWF file, required for SWFVerification.", OFFSET(swfsize), AV_OPT_TYPE_INT, {.i64 = 0}, 0, INT_MAX, DEC},
{"rtmp_swfurl", "URL of the SWF player. By default no value will be sent", OFFSET(swfurl), AV_OPT_TYPE_STRING, {.str = NULL }, 0, 0, DEC|ENC},
{"rtmp_swfverify", "URL to player swf file, compute hash/size automatically.", OFFSET(swfverify), AV_OPT_TYPE_STRING, {.str = NULL }, 0, 0, DEC},
{"rtmp_tcurl", "URL of the target stream. Defaults to proto://host[:port]/app.", OFFSET(tcurl), AV_OPT_TYPE_STRING, {.str = NULL }, 0, 0, DEC|ENC},
{"rtmp_listen", "Listen for incoming rtmp connections", OFFSET(listen), AV_OPT_TYPE_INT, {.i64 = 0}, INT_MIN, INT_MAX, DEC, "rtmp_listen" },
{"listen", "Listen for incoming rtmp connections", OFFSET(listen), AV_OPT_TYPE_INT, {.i64 = 0}, INT_MIN, INT_MAX, DEC, "rtmp_listen" },
{"timeout", "Maximum timeout (in seconds) to wait for incoming connections. -1 is infinite. Implies -rtmp_listen 1", OFFSET(listen_timeout), AV_OPT_TYPE_INT, {.i64 = -1}, INT_MIN, INT_MAX, DEC, "rtmp_listen" },
{ NULL },
};
#define RTMP_PROTOCOL(flavor) \
static const AVClass flavor##_class = { \
.class_name = #flavor, \
.item_name = av_default_item_name, \
.option = rtmp_options, \
.version = LIBAVUTIL_VERSION_INT, \
}; \
\
const URLProtocol ff_##flavor##_protocol = { \
.name = #flavor, \
.url_open2 = rtmp_open, \
.url_read = rtmp_read, \
.url_read_seek = rtmp_seek, \
.url_read_pause = rtmp_pause, \
.url_write = rtmp_write, \
.url_close = rtmp_close, \
.priv_data_size = sizeof(RTMPContext), \
.flags = URL_PROTOCOL_FLAG_NETWORK, \
.priv_data_class= &flavor##_class, \
};
RTMP_PROTOCOL(rtmp)
RTMP_PROTOCOL(rtmpe)
RTMP_PROTOCOL(rtmps)
RTMP_PROTOCOL(rtmpt)
RTMP_PROTOCOL(rtmpte)
RTMP_PROTOCOL(rtmpts)
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