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ffmpeg/libavformat/rtmpproto.c
Martin Storsjö fab8156b2f avio: Copy URLContext generic options into child URLContexts 
Since all URLContexts have the same AVOptions, such AVOptions
will be applied on the outermost context only and removed from the
dict, while they probably make sense on all contexts.

This makes sure that rw_timeout gets propagated to the innermost
URLContext (to make sure it gets passed to the tcp protocol, when
opening a http connection for instance).

Alternatively, such matching options would be kept in the dict
and only removed after the ffurl_connect call.

Signed-off-by: Martin Storsjö <martin@martin.st>
2016-03-24 10:34:19 +02:00

3110 lines
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/*
* RTMP network protocol
* Copyright (c) 2009 Konstantin Shishkov
*
* This file is part of Libav.
*
* Libav 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.
*
* Libav is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with Libav; if not, write to the Free Software
* 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/hmac.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 128
#define PLAYPATH_MAX_LENGTH 256
#define TCURL_MAX_LENGTH 512
#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 client_report_size; ///< number of bytes after which client should report to server
uint32_t bytes_read; ///< number of bytes read from server
uint32_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 server_bw; ///< server bandwidth
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 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(&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_free(rt->tracked_methods[i].name);
av_free(rt->tracked_methods);
rt->tracked_methods = NULL;
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)) < 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) {
ff_amf_write_field_name(&p, "swfUrl");
ff_amf_write_string(&p, rt->swfurl);
}
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);
}
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;
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;
if (pkt.type == RTMP_PT_CHUNK_SIZE) {
if ((ret = handle_chunk_size(s, &pkt)) < 0)
return ret;
ff_rtmp_packet_destroy(&pkt);
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;
}
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 speficication)
if ((ret = ff_rtmp_packet_create(&pkt, RTMP_NETWORK_CHANNEL,
RTMP_PT_SERVER_BW, 0, 4)) < 0)
return ret;
p = pkt.data;
bytestream_put_be32(&p, rt->server_bw);
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;
// Send Peer Bandwidth
if ((ret = ff_rtmp_packet_create(&pkt, RTMP_NETWORK_CHANNEL,
RTMP_PT_CLIENT_BW, 0, 5)) < 0)
return ret;
p = pkt.data;
bytestream_put_be32(&p, rt->server_bw);
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;
// Ping request
if ((ret = ff_rtmp_packet_create(&pkt, RTMP_NETWORK_CHANNEL,
RTMP_PT_PING, 0, 6)) < 0)
return ret;
p = pkt.data;
bytestream_put_be16(&p, 0); // 0 -> Stream Begin
bytestream_put_be32(&p, 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_SYSTEM_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 preapare for receiving media streams.
*/
static int gen_fcpublish_stream(URLContext *s, RTMPContext *rt)
{
RTMPPacket pkt;
uint8_t *p;
int ret;
if ((ret = ff_rtmp_packet_create(&pkt, RTMP_SYSTEM_CHANNEL, RTMP_PT_INVOKE,
0, 25 + strlen(rt->playpath))) < 0)
return ret;
av_log(s, AV_LOG_DEBUG, "FCPublish stream...\n");
p = pkt.data;
ff_amf_write_string(&p, "FCPublish");
ff_amf_write_number(&p, ++rt->nb_invokes);
ff_amf_write_null(&p);
ff_amf_write_string(&p, rt->playpath);
return rtmp_send_packet(rt, &pkt, 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_PING,
1, 10)) < 0)
return ret;
p = pkt.data;
bytestream_put_be16(&p, 3);
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_PING,
ppkt->timestamp + 1, 6)) < 0)
return ret;
p = pkt.data;
bytestream_put_be16(&p, 7);
bytestream_put_be32(&p, AV_RB32(ppkt->data+2));
return rtmp_send_packet(rt, &pkt, 0);
}
/**
* Generate 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_PING,
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 server bandwidth message and send it to the server.
*/
static int gen_server_bw(URLContext *s, RTMPContext *rt)
{
RTMPPacket pkt;
uint8_t *p;
int ret;
if ((ret = ff_rtmp_packet_create(&pkt, RTMP_NETWORK_CHANNEL, RTMP_PT_SERVER_BW,
0, 4)) < 0)
return ret;
p = pkt.data;
bytestream_put_be32(&p, rt->server_bw);
return rtmp_send_packet(rt, &pkt, 0);
}
/**
* Generate check bandwidth message and send it to the server.
*/
static int gen_check_bw(URLContext *s, RTMPContext *rt)
{
RTMPPacket pkt;
uint8_t *p;
int ret;
if ((ret = ff_rtmp_packet_create(&pkt, RTMP_SYSTEM_CHANNEL, RTMP_PT_INVOKE,
0, 21)) < 0)
return ret;
p = pkt.data;
ff_amf_write_string(&p, "_checkbw");
ff_amf_write_number(&p, ++rt->nb_invokes);
ff_amf_write_null(&p);
return rtmp_send_packet(rt, &pkt, 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);
}
int ff_rtmp_calc_digest(const uint8_t *src, int len, int gap,
const uint8_t *key, int keylen, uint8_t *dst)
{
AVHMAC *hmac;
hmac = av_hmac_alloc(AV_HMAC_SHA256);
if (!hmac)
return AVERROR(ENOMEM);
av_hmac_init(hmac, key, keylen);
if (gap <= 0) {
av_hmac_update(hmac, src, len);
} else { //skip 32 bytes used for storing digest
av_hmac_update(hmac, src, gap);
av_hmac_update(hmac, src + gap + 32, len - gap - 32);
}
av_hmac_final(hmac, dst, 32);
av_hmac_free(hmac);
return 0;
}
int ff_rtmp_calc_digest_pos(const uint8_t *buf, int off, int mod_val,
int add_val)
{
int i, digest_pos = 0;
for (i = 0; i < 4; i++)
digest_pos += buf[i + off];
digest_pos = digest_pos % mod_val + add_val;
return digest_pos;
}
/**
* Put HMAC-SHA2 digest of packet data (except for the bytes where this digest
* will be stored) into that packet.
*
* @param buf handshake data (1536 bytes)
* @param encrypted use an encrypted connection (RTMPE)
* @return offset to the digest inside input data
*/
static int rtmp_handshake_imprint_with_digest(uint8_t *buf, int encrypted)
{
int ret, digest_pos;
if (encrypted)
digest_pos = ff_rtmp_calc_digest_pos(buf, 772, 728, 776);
else
digest_pos = ff_rtmp_calc_digest_pos(buf, 8, 728, 12);
ret = ff_rtmp_calc_digest(buf, RTMP_HANDSHAKE_PACKET_SIZE, digest_pos,
rtmp_player_key, PLAYER_KEY_OPEN_PART_LEN,
buf + digest_pos);
if (ret < 0)
return ret;
return digest_pos;
}
/**
* Verify that the received server response has the expected digest value.
*
* @param buf handshake data received from the server (1536 bytes)
* @param off position to search digest offset from
* @return 0 if digest is valid, digest position otherwise
*/
static int rtmp_validate_digest(uint8_t *buf, int off)
{
uint8_t digest[32];
int ret, digest_pos;
digest_pos = ff_rtmp_calc_digest_pos(buf, off, 728, off + 4);
ret = ff_rtmp_calc_digest(buf, RTMP_HANDSHAKE_PACKET_SIZE, digest_pos,
rtmp_server_key, SERVER_KEY_OPEN_PART_LEN,
digest);
if (ret < 0)
return ret;
if (!memcmp(digest, buf + digest_pos, 32))
return digest_pos;
return 0;
}
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, out_size;
URLContext *stream;
char swfhash[32];
int swfsize;
int ret = 0;
/* Get the SWF player file. */
if ((ret = ffurl_open(&stream, rt->swfverify, AVIO_FLAG_READ,
&s->interrupt_callback, NULL, s->protocols, 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)) {
/* 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 CONFIG_ZLIB
if ((ret = rtmp_uncompress_swfplayer(in_data + 8, in_size - 8,
&out_data, &out_size)) < 0)
goto fail;
#else
av_log(s, AV_LOG_ERROR,
"Zlib is required for decompressing the SWF player file.\n");
ret = AVERROR(EINVAL);
goto fail;
#endif
swfsize = out_size;
swfdata = out_data;
} 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_ping(URLContext *s, RTMPPacket *pkt)
{
RTMPContext *rt = s->priv_data;
int t, ret;
if (pkt->size < 2) {
av_log(s, AV_LOG_ERROR, "Too short ping packet (%d)\n",
pkt->size);
return AVERROR_INVALIDDATA;
}
t = AV_RB16(pkt->data);
if (t == 6) {
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_client_bw(URLContext *s, RTMPPacket *pkt)
{
RTMPContext *rt = s->priv_data;
if (pkt->size < 4) {
av_log(s, AV_LOG_ERROR,
"Client bandwidth report packet is less than 4 bytes long (%d)\n",
pkt->size);
return AVERROR_INVALIDDATA;
}
rt->client_report_size = AV_RB32(pkt->data);
if (rt->client_report_size <= 0) {
av_log(s, AV_LOG_ERROR, "Incorrect client bandwidth %d\n",
rt->client_report_size);
return AVERROR_INVALIDDATA;
}
av_log(s, AV_LOG_DEBUG, "Client bandwidth = %d\n", rt->client_report_size);
rt->client_report_size >>= 1;
return 0;
}
static int handle_server_bw(URLContext *s, RTMPPacket *pkt)
{
RTMPContext *rt = s->priv_data;
if (pkt->size < 4) {
av_log(s, AV_LOG_ERROR,
"Too short server bandwidth report packet (%d)\n",
pkt->size);
return AVERROR_INVALIDDATA;
}
rt->server_bw = AV_RB32(pkt->data);
if (rt->server_bw <= 0) {
av_log(s, AV_LOG_ERROR, "Incorrect server bandwidth %d\n",
rt->server_bw);
return AVERROR_INVALIDDATA;
}
av_log(s, AV_LOG_DEBUG, "Server bandwidth = %d\n", rt->server_bw);
return 0;
}
static int 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;
}
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_PING, 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_field_name(&pp, "clientid");
snprintf(statusmsg, sizeof(statusmsg), "%s", LIBAVFORMAT_IDENT);
ff_amf_write_string(&pp, statusmsg);
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[64];
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);
// 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 otherwiss
*/
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 resonse 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_server_bw(s, rt)) < 0)
goto fail;
}
if ((ret = gen_create_stream(s, rt)) < 0)
goto fail;
if (rt->is_input) {
/* Send the FCSubscribe command when the name of live
* stream is defined by the user or if it's a live stream. */
if (rt->subscribe) {
if ((ret = gen_fcsubscribe_stream(s, rt, rt->subscribe)) < 0)
goto fail;
} else if (rt->live == -1) {
if ((ret = gen_fcsubscribe_stream(s, rt, rt->playpath)) < 0)
goto fail;
}
}
} else if (!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_PING:
if ((ret = handle_ping(s, pkt)) < 0)
return ret;
break;
case RTMP_PT_CLIENT_BW:
if ((ret = handle_client_bw(s, pkt)) < 0)
return ret;
break;
case RTMP_PT_SERVER_BW:
if ((ret = handle_server_bw(s, pkt)) < 0)
return ret;
break;
case RTMP_PT_INVOKE:
if ((ret = handle_invoke(s, pkt)) < 0)
return ret;
break;
case RTMP_PT_VIDEO:
case RTMP_PT_AUDIO:
case RTMP_PT_METADATA:
case RTMP_PT_NOTIFY:
/* Audio, Video and Metadata packets are parsed in get_packet() */
break;
default:
av_log(s, AV_LOG_VERBOSE, "Unknown packet type received 0x%02X\n", pkt->type);
break;
}
return 0;
}
static int handle_metadata(RTMPContext *rt, RTMPPacket *pkt)
{
int ret, old_flv_size, type;
const uint8_t *next;
uint8_t *p;
uint32_t size;
uint32_t ts, cts, pts = 0;
old_flv_size = update_offset(rt, pkt->size);
if ((ret = av_reallocp(&rt->flv_data, rt->flv_size)) < 0) {
rt->flv_size = rt->flv_off = 0;
return ret;
}
next = pkt->data;
p = rt->flv_data + old_flv_size;
/* copy data while rewriting timestamps */
ts = pkt->timestamp;
while (next - pkt->data < pkt->size - RTMP_HEADER) {
type = bytestream_get_byte(&next);
size = bytestream_get_be24(&next);
cts = bytestream_get_be24(&next);
cts |= bytestream_get_byte(&next) << 24;
if (!pts)
pts = cts;
ts += cts - pts;
pts = cts;
if (size + 3 + 4 > pkt->data + pkt->size - next)
break;
bytestream_put_byte(&p, type);
bytestream_put_be24(&p, size);
bytestream_put_be24(&p, ts);
bytestream_put_byte(&p, ts >> 24);
memcpy(p, next, size + 3 + 4);
p += size + 3;
bytestream_put_be32(&p, size + RTMP_HEADER);
next += size + 3 + 4;
}
if (p != rt->flv_data + rt->flv_size) {
av_log(NULL, AV_LOG_WARNING, "Incomplete flv packets in "
"RTMP_PT_METADATA packet\n");
rt->flv_size = p - rt->flv_data;
}
return 0;
}
/**
* Interact with the server by receiving and sending RTMP packets until
* there is some significant data (media data or expected status notification).
*
* @param s reading context
* @param for_header non-zero value tells function to work until it
* gets notification from the server that playing has been started,
* otherwise function will work until some media data is received (or
* an error happens)
* @return 0 for successful operation, negative value in case of error
*/
static int get_packet(URLContext *s, int for_header)
{
RTMPContext *rt = s->priv_data;
int ret;
if (rt->state == STATE_STOPPED)
return AVERROR_EOF;
for (;;) {
RTMPPacket rpkt = { 0 };
if ((ret = ff_rtmp_packet_read(rt->stream, &rpkt,
rt->in_chunk_size, &rt->prev_pkt[0],
&rt->nb_prev_pkt[0])) <= 0) {
if (ret == 0) {
return AVERROR(EAGAIN);
} else {
return AVERROR(EIO);
}
}
// Track timestamp for later use
rt->last_timestamp = rpkt.timestamp;
rt->bytes_read += ret;
if (rt->bytes_read > rt->last_bytes_read + rt->client_report_size) {
av_log(s, AV_LOG_DEBUG, "Sending bytes read report\n");
if ((ret = gen_bytes_read(s, rt, rpkt.timestamp + 1)) < 0)
return ret;
rt->last_bytes_read = rt->bytes_read;
}
ret = rtmp_parse_result(s, rt, &rpkt);
// 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 0;
}
ff_rtmp_packet_destroy(&rpkt);
}
}
static int rtmp_close(URLContext *h)
{
RTMPContext *rt = h->priv_data;
int ret = 0, i, j;
if (!rt->is_input) {
rt->flv_data = NULL;
if (rt->out_pkt.size)
ff_rtmp_packet_destroy(&rt->out_pkt);
if (rt->state > STATE_FCPUBLISH)
ret = gen_fcunpublish_stream(h, rt);
}
if (rt->state > STATE_HANDSHAKED)
ret = gen_delete_stream(h, rt);
for (i = 0; i < 2; i++) {
for (j = 0; j < rt->nb_prev_pkt[i]; j++)
ff_rtmp_packet_destroy(&rt->prev_pkt[i][j]);
av_freep(&rt->prev_pkt[i]);
}
free_tracked_methods(rt);
av_freep(&rt->flv_data);
ffurl_close(rt->stream);
return ret;
}
/**
* Insert a fake onMetadata packet into the FLV stream to notify the FLV
* demuxer about the duration of the stream.
*
* This should only be done if there was no real onMetadata packet sent by the
* server at the start of the stream and if we were able to retrieve a valid
* duration via a getStreamLength call.
*
* @return 0 for successful operation, negative value in case of error
*/
static int inject_fake_duration_metadata(RTMPContext *rt)
{
// We need to insert the metdata 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_free(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)
{
RTMPContext *rt = s->priv_data;
char proto[8], hostname[256], path[1024], auth[100], *fname;
char *old_app, *qmark, fname_buffer[1024];
uint8_t buf[2048];
int port;
AVDictionary *opts = NULL;
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);
if (strchr(path, ' ')) {
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");
}
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(&rt->stream, buf, AVIO_FLAG_READ_WRITE,
&s->interrupt_callback, &opts, s->protocols, 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) {
fname = next;
rt->app[0] = '\0';
} else {
// make sure we do not mismatch a playpath for an application instance
char *c = strchr(p + 1, ':');
fname = strchr(p + 1, '/');
if (!fname || (c && c < fname)) {
fname = p + 1;
av_strlcpy(rt->app, path + 1, 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.
av_free(rt->app);
rt->app = old_app;
}
if (!rt->playpath) {
int len = strlen(fname);
rt->playpath = av_malloc(PLAYPATH_MAX_LENGTH);
if (!rt->playpath) {
ret = AVERROR(ENOMEM);
goto fail;
}
if (!strchr(fname, ':') && len >= 4 &&
(!strcmp(fname + len - 4, ".f4v") ||
!strcmp(fname + len - 4, ".mp4"))) {
memcpy(rt->playpath, "mp4:", 5);
} else {
if (len >= 4 && !strcmp(fname + len - 4, ".flv"))
fname[len - 4] = '\0';
rt->playpath[0] = 0;
}
av_strlcat(rt->playpath, fname, PLAYPATH_MAX_LENGTH);
}
if (!rt->tcurl) {
rt->tcurl = av_malloc(TCURL_MAX_LENGTH);
if (!rt->tcurl) {
ret = AVERROR(ENOMEM);
goto fail;
}
ff_url_join(rt->tcurl, TCURL_MAX_LENGTH, proto, NULL, hostname,
port, "/%s", rt->app);
}
if (!rt->flashver) {
rt->flashver = av_malloc(FLASHVER_MAX_LENGTH);
if (!rt->flashver) {
ret = AVERROR(ENOMEM);
goto fail;
}
if (rt->is_input) {
snprintf(rt->flashver, FLASHVER_MAX_LENGTH, "%s %d,%d,%d,%d",
RTMP_CLIENT_PLATFORM, RTMP_CLIENT_VER1, RTMP_CLIENT_VER2,
RTMP_CLIENT_VER3, RTMP_CLIENT_VER4);
} else {
snprintf(rt->flashver, FLASHVER_MAX_LENGTH,
"FMLE/3.0 (compatible; %s)", LIBAVFORMAT_IDENT);
}
}
rt->client_report_size = 1048576;
rt->bytes_read = 0;
rt->has_audio = 0;
rt->has_video = 0;
rt->received_metadata = 0;
rt->last_bytes_read = 0;
rt->server_bw = 2500000;
rt->duration = 0;
av_log(s, AV_LOG_DEBUG, "Proto = %s, path = %s, app = %s, fname = %s\n",
proto, path, rt->app, rt->playpath);
if (!rt->listen) {
if ((ret = gen_connect(s, rt)) < 0)
goto fail;
} else {
if ((ret = read_connect(s, s->priv_data)) < 0)
goto fail;
}
do {
ret = get_packet(s, 1);
} while (ret == AVERROR(EAGAIN));
if (ret < 0)
goto fail;
if (rt->do_reconnect) {
int i;
ffurl_close(rt->stream);
rt->stream = NULL;
rt->do_reconnect = 0;
rt->nb_invokes = 0;
for (i = 0; i < 2; i++)
memset(rt->prev_pkt[i], 0,
sizeof(**rt->prev_pkt) * rt->nb_prev_pkt[i]);
free_tracked_methods(rt);
goto reconnect;
}
if (rt->is_input) {
// generate FLV header for demuxer
rt->flv_size = 13;
if ((ret = av_reallocp(&rt->flv_data, rt->flv_size)) < 0)
goto fail;
rt->flv_off = 0;
memcpy(rt->flv_data, "FLV\1\0\0\0\0\011\0\0\0\0", rt->flv_size);
// Read packets until we reach the first A/V packet or read metadata.
// If there was a metadata package in front of the A/V packets, we can
// build the FLV header from this. If we do not receive any metadata,
// the FLV decoder will allocate the needed streams when their first
// audio or video packet arrives.
while (!rt->has_audio && !rt->has_video && !rt->received_metadata) {
if ((ret = get_packet(s, 0)) < 0)
goto fail;
}
// Either after we have read the metadata or (if there is none) the
// first packet of an A/V stream, we have a better knowledge about the
// streams, so set the FLV header accordingly.
if (rt->has_audio) {
rt->flv_data[4] |= FLV_HEADER_FLAG_HASAUDIO;
}
if (rt->has_video) {
rt->flv_data[4] |= FLV_HEADER_FLAG_HASVIDEO;
}
// If we received the first packet of an A/V stream and no metadata but
// the server returned a valid duration, create a fake metadata packet
// to inform the FLV decoder about the duration.
if (!rt->received_metadata && rt->duration > 0) {
if ((ret = inject_fake_duration_metadata(rt)) < 0)
goto fail;
}
} else {
rt->flv_size = 0;
rt->flv_data = NULL;
rt->flv_off = 0;
rt->skip_bytes = 13;
}
s->max_packet_size = rt->stream->max_packet_size;
s->is_streamed = 1;
return 0;
fail:
av_dict_free(&opts);
rtmp_close(s);
return ret;
}
static int rtmp_read(URLContext *s, uint8_t *buf, int size)
{
RTMPContext *rt = s->priv_data;
int orig_size = size;
int ret;
while (size > 0) {
int data_left = rt->flv_size - rt->flv_off;
if (data_left >= size) {
memcpy(buf, rt->flv_data + rt->flv_off, size);
rt->flv_off += size;
return orig_size;
}
if (data_left > 0) {
memcpy(buf, rt->flv_data + rt->flv_off, data_left);
buf += data_left;
size -= data_left;
rt->flv_off = rt->flv_size;
return data_left;
}
if ((ret = get_packet(s, 0)) < 0)
return ret;
}
return orig_size;
}
static int64_t rtmp_seek(URLContext *s, int stream_index, int64_t timestamp,
int flags)
{
RTMPContext *rt = s->priv_data;
int ret;
av_log(s, AV_LOG_DEBUG,
"Seek on stream index %d at timestamp %"PRId64" with flags %08x\n",
stream_index, timestamp, flags);
if ((ret = gen_seek(s, rt, timestamp)) < 0) {
av_log(s, AV_LOG_ERROR,
"Unable to send seek command on stream index %d at timestamp "
"%"PRId64" with flags %08x\n",
stream_index, timestamp, flags);
return ret;
}
rt->flv_off = rt->flv_size;
rt->state = STATE_SEEKING;
return timestamp;
}
static int rtmp_pause(URLContext *s, int pause)
{
RTMPContext *rt = s->priv_data;
int ret;
av_log(s, AV_LOG_DEBUG, "Pause at timestamp %d\n",
rt->last_timestamp);
if ((ret = gen_pause(s, rt, pause, rt->last_timestamp)) < 0) {
av_log(s, AV_LOG_ERROR, "Unable to send pause command at timestamp %d\n",
rt->last_timestamp);
return ret;
}
return 0;
}
static int rtmp_write(URLContext *s, const uint8_t *buf, int size)
{
RTMPContext *rt = s->priv_data;
int size_temp = size;
int pktsize, pkttype, copy;
uint32_t ts;
const uint8_t *buf_temp = buf;
uint8_t c;
int ret;
do {
if (rt->skip_bytes) {
int skip = FFMIN(rt->skip_bytes, size_temp);
buf_temp += skip;
size_temp -= skip;
rt->skip_bytes -= skip;
continue;
}
if (rt->flv_header_bytes < RTMP_HEADER) {
const uint8_t *header = rt->flv_header;
int channel = RTMP_AUDIO_CHANNEL;
copy = FFMIN(RTMP_HEADER - rt->flv_header_bytes, size_temp);
bytestream_get_buffer(&buf_temp, rt->flv_header + rt->flv_header_bytes, copy);
rt->flv_header_bytes += copy;
size_temp -= copy;
if (rt->flv_header_bytes < RTMP_HEADER)
break;
pkttype = bytestream_get_byte(&header);
pktsize = bytestream_get_be24(&header);
ts = bytestream_get_be24(&header);
ts |= bytestream_get_byte(&header) << 24;
bytestream_get_be24(&header);
rt->flv_size = pktsize;
if (pkttype == RTMP_PT_VIDEO)
channel = RTMP_VIDEO_CHANNEL;
if (((pkttype == RTMP_PT_VIDEO || pkttype == RTMP_PT_AUDIO) && ts == 0) ||
pkttype == RTMP_PT_NOTIFY) {
if ((ret = ff_rtmp_check_alloc_array(&rt->prev_pkt[1],
&rt->nb_prev_pkt[1],
channel)) < 0)
return ret;
// Force sending a full 12 bytes header by clearing the
// channel id, to make it not match a potential earlier
// packet in the same channel.
rt->prev_pkt[1][channel].channel_id = 0;
}
//this can be a big packet, it's better to send it right here
if ((ret = ff_rtmp_packet_create(&rt->out_pkt, channel,
pkttype, ts, pktsize)) < 0)
return ret;
rt->out_pkt.extra = rt->stream_id;
rt->flv_data = rt->out_pkt.data;
}
copy = FFMIN(rt->flv_size - rt->flv_off, size_temp);
bytestream_get_buffer(&buf_temp, rt->flv_data + rt->flv_off, copy);
rt->flv_off += copy;
size_temp -= copy;
if (rt->flv_off == rt->flv_size) {
rt->skip_bytes = 4;
if (rt->out_pkt.type == RTMP_PT_NOTIFY) {
// For onMetaData and |RtmpSampleAccess packets, we want
// @setDataFrame prepended to the packet before it gets sent.
// However, not all RTMP_PT_NOTIFY packets (e.g., onTextData
// and onCuePoint).
uint8_t commandbuffer[64];
int stringlen = 0;
GetByteContext gbc;
bytestream2_init(&gbc, rt->flv_data, rt->flv_size);
if (!ff_amf_read_string(&gbc, commandbuffer, sizeof(commandbuffer),
&stringlen)) {
if (!strcmp(commandbuffer, "onMetaData") ||
!strcmp(commandbuffer, "|RtmpSampleAccess")) {
uint8_t *ptr;
if ((ret = av_reallocp(&rt->out_pkt.data, rt->out_pkt.size + 16)) < 0) {
rt->flv_size = rt->flv_off = rt->flv_header_bytes = 0;
return ret;
}
memmove(rt->out_pkt.data + 16, rt->out_pkt.data, rt->out_pkt.size);
rt->out_pkt.size += 16;
ptr = rt->out_pkt.data;
ff_amf_write_string(&ptr, "@setDataFrame");
}
}
}
if ((ret = rtmp_send_packet(rt, &rt->out_pkt, 0)) < 0)
return ret;
rt->flv_size = 0;
rt->flv_off = 0;
rt->flv_header_bytes = 0;
rt->flv_nb_packets++;
}
} while (buf_temp - buf < size);
if (rt->flv_nb_packets < rt->flush_interval)
return size;
rt->flv_nb_packets = 0;
/* set stream into nonblocking mode */
rt->stream->flags |= AVIO_FLAG_NONBLOCK;
/* try to read one byte from the stream */
ret = ffurl_read(rt->stream, &c, 1);
/* switch the stream back into blocking mode */
rt->stream->flags &= ~AVIO_FLAG_NONBLOCK;
if (ret == AVERROR(EAGAIN)) {
/* no incoming data to handle */
return size;
} else if (ret < 0) {
return ret;
} else if (ret == 1) {
RTMPPacket rpkt = { 0 };
if ((ret = ff_rtmp_packet_read_internal(rt->stream, &rpkt,
rt->in_chunk_size,
&rt->prev_pkt[0],
&rt->nb_prev_pkt[0], c)) <= 0)
return ret;
if ((ret = rtmp_parse_result(s, rt, &rpkt)) < 0)
return ret;
ff_rtmp_packet_destroy(&rpkt);
}
return size;
}
#define OFFSET(x) offsetof(RTMPContext, x)
#define DEC AV_OPT_FLAG_DECODING_PARAM
#define ENC AV_OPT_FLAG_ENCODING_PARAM
static const AVOption rtmp_options[] = {
{"rtmp_app", "Name of application to connect to on the RTMP server", OFFSET(app), AV_OPT_TYPE_STRING, {.str = NULL }, 0, 0, DEC|ENC},
{"rtmp_buffer", "Set buffer time in milliseconds. The default is 3000.", OFFSET(client_buffer_time), AV_OPT_TYPE_INT, {.i64 = 3000}, 0, INT_MAX, DEC|ENC},
{"rtmp_conn", "Append arbitrary AMF data to the Connect message", OFFSET(conn), AV_OPT_TYPE_STRING, {.str = NULL }, 0, 0, DEC|ENC},
{"rtmp_flashver", "Version of the Flash plugin used to run the SWF player.", OFFSET(flashver), AV_OPT_TYPE_STRING, {.str = NULL }, 0, 0, DEC|ENC},
{"rtmp_flush_interval", "Number of packets flushed in the same request (RTMPT only).", OFFSET(flush_interval), AV_OPT_TYPE_INT, {.i64 = 10}, 0, INT_MAX, ENC},
{"rtmp_live", "Specify that the media is a live stream.", OFFSET(live), AV_OPT_TYPE_INT, {.i64 = -2}, INT_MIN, INT_MAX, DEC, "rtmp_live"},
{"any", "both", 0, AV_OPT_TYPE_CONST, {.i64 = -2}, 0, 0, DEC, "rtmp_live"},
{"live", "live stream", 0, AV_OPT_TYPE_CONST, {.i64 = -1}, 0, 0, DEC, "rtmp_live"},
{"recorded", "recorded stream", 0, AV_OPT_TYPE_CONST, {.i64 = 0}, 0, 0, DEC, "rtmp_live"},
{"rtmp_pageurl", "URL of the web page in which the media was embedded. By default no value will be sent.", OFFSET(pageurl), AV_OPT_TYPE_STRING, {.str = NULL }, 0, 0, DEC},
{"rtmp_playpath", "Stream identifier to play or to publish", OFFSET(playpath), AV_OPT_TYPE_STRING, {.str = NULL }, 0, 0, DEC|ENC},
{"rtmp_subscribe", "Name of live stream to subscribe to. Defaults to rtmp_playpath.", OFFSET(subscribe), AV_OPT_TYPE_STRING, {.str = NULL }, 0, 0, DEC},
{"rtmp_swfhash", "SHA256 hash of the decompressed SWF file (32 bytes).", OFFSET(swfhash), AV_OPT_TYPE_BINARY, .flags = DEC},
{"rtmp_swfsize", "Size of the decompressed SWF file, required for SWFVerification.", OFFSET(swfsize), AV_OPT_TYPE_INT, {.i64 = 0}, 0, INT_MAX, DEC},
{"rtmp_swfurl", "URL of the SWF player. By default no value will be sent", OFFSET(swfurl), AV_OPT_TYPE_STRING, {.str = NULL }, 0, 0, DEC|ENC},
{"rtmp_swfverify", "URL to player swf file, compute hash/size automatically.", OFFSET(swfverify), AV_OPT_TYPE_STRING, {.str = NULL }, 0, 0, DEC},
{"rtmp_tcurl", "URL of the target stream. Defaults to proto://host[:port]/app.", OFFSET(tcurl), AV_OPT_TYPE_STRING, {.str = NULL }, 0, 0, DEC|ENC},
{"rtmp_listen", "Listen for incoming rtmp connections", OFFSET(listen), AV_OPT_TYPE_INT, {.i64 = 0}, INT_MIN, INT_MAX, DEC, "rtmp_listen" },
{"listen", "Listen for incoming rtmp connections", OFFSET(listen), AV_OPT_TYPE_INT, {.i64 = 0}, INT_MIN, INT_MAX, DEC, "rtmp_listen" },
{"timeout", "Maximum timeout (in seconds) to wait for incoming connections. -1 is infinite. Implies -rtmp_listen 1", OFFSET(listen_timeout), AV_OPT_TYPE_INT, {.i64 = -1}, INT_MIN, INT_MAX, DEC, "rtmp_listen" },
{ NULL },
};
#define RTMP_PROTOCOL(flavor) \
static const AVClass flavor##_class = { \
.class_name = #flavor, \
.item_name = av_default_item_name, \
.option = rtmp_options, \
.version = LIBAVUTIL_VERSION_INT, \
}; \
\
const URLProtocol ff_##flavor##_protocol = { \
.name = #flavor, \
.url_open = 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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