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live555.cpp
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/*****************************************************************************
* live555.cpp : LIVE555 Streaming Media support.
*****************************************************************************
* Copyright (C) 2003-2007 VLC authors and VideoLAN
* $Id$
*
* Authors: Laurent Aimar <[email protected]>
* Derk-Jan Hartman <hartman at videolan. org>
* Derk-Jan Hartman <djhartman at m2x .dot. nl> for M2X
* Sébastien Escudier <sebastien-devel celeos eu>
*
* This program 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.
*
* This program 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 this program; if not, write to the Free Software Foundation,
* Inc., 51 Franklin Street, Fifth Floor, Boston MA 02110-1301, USA.
*****************************************************************************/
/*****************************************************************************
* Preamble
*****************************************************************************/
#ifdef HAVE_CONFIG_H
# include "config.h"
#endif
#include <inttypes.h>
#include <vlc_common.h>
#include <vlc_plugin.h>
#include <vlc_input.h>
#include <vlc_demux.h>
#include <vlc_dialog.h>
#include <vlc_url.h>
#include <vlc_strings.h>
#include <limits.h>
#include <assert.h>
#if defined( _WIN32 )
# include <winsock2.h>
#endif
#include <UsageEnvironment.hh>
#include <BasicUsageEnvironment.hh>
#include <GroupsockHelper.hh>
#include <liveMedia.hh>
#include <liveMedia_version.hh>
#include <Base64.hh>
extern "C" {
#include "../access/mms/asf.h" /* Who said ugly ? */
}
/*****************************************************************************
* Module descriptor
*****************************************************************************/
static int Open ( vlc_object_t * );
static void Close( vlc_object_t * );
#define KASENNA_TEXT N_( "Kasenna RTSP dialect")
#define KASENNA_LONGTEXT N_( "Kasenna servers use an old and nonstandard " \
"dialect of RTSP. With this parameter VLC will try this dialect, but "\
"then it cannot connect to normal RTSP servers." )
#define WMSERVER_TEXT N_("WMServer RTSP dialect")
#define WMSERVER_LONGTEXT N_("WMServer uses a nonstandard dialect " \
"of RTSP. Selecting this parameter will tell VLC to assume some " \
"options contrary to RFC 2326 guidelines.")
#define USER_TEXT N_("Username")
#define USER_LONGTEXT N_("Sets the username for the connection, " \
"if no username or password are set in the url.")
#define PASS_TEXT N_("Password")
#define PASS_LONGTEXT N_("Sets the password for the connection, " \
"if no username or password are set in the url.")
#define FRAME_BUFFER_SIZE_TEXT N_("RTSP frame buffer size")
#define FRAME_BUFFER_SIZE_LONGTEXT N_("RTSP start frame buffer size of the video " \
"track, can be increased in case of broken pictures due " \
"to too small buffer.")
#define DEFAULT_FRAME_BUFFER_SIZE 100000
vlc_module_begin ()
set_description( N_("RTP/RTSP/SDP demuxer (using Live555)" ) )
set_capability( "demux", 50 )
set_shortname( "RTP/RTSP")
set_callbacks( Open, Close )
add_shortcut( "live", "livedotcom" )
set_category( CAT_INPUT )
set_subcategory( SUBCAT_INPUT_DEMUX )
add_submodule ()
set_description( N_("RTSP/RTP access and demux") )
add_shortcut( "rtsp", "pnm", "live", "livedotcom", "satip" )
set_capability( "access_demux", 0 )
set_callbacks( Open, Close )
add_bool( "rtsp-tcp", false,
N_("Use RTP over RTSP (TCP)"),
N_("Use RTP over RTSP (TCP)"), true )
change_safe()
add_integer( "rtp-client-port", -1,
N_("Client port"),
N_("Port to use for the RTP source of the session"), true )
add_bool( "rtsp-mcast", false,
N_("Force multicast RTP via RTSP"),
N_("Force multicast RTP via RTSP"), true )
change_safe()
add_bool( "rtsp-http", false,
N_("Tunnel RTSP and RTP over HTTP"),
N_("Tunnel RTSP and RTP over HTTP"), true )
change_safe()
add_integer( "rtsp-http-port", 80,
N_("HTTP tunnel port"),
N_("Port to use for tunneling the RTSP/RTP over HTTP."),
true )
add_bool( "rtsp-kasenna", false, KASENNA_TEXT,
KASENNA_LONGTEXT, true )
change_safe()
add_bool( "rtsp-wmserver", false, WMSERVER_TEXT,
WMSERVER_LONGTEXT, true)
change_safe()
add_string( "rtsp-user", NULL, USER_TEXT,
USER_LONGTEXT, true )
change_safe()
add_password( "rtsp-pwd", NULL, PASS_TEXT,
PASS_LONGTEXT, true )
add_integer( "rtsp-frame-buffer-size", DEFAULT_FRAME_BUFFER_SIZE,
FRAME_BUFFER_SIZE_TEXT, FRAME_BUFFER_SIZE_LONGTEXT,
true )
vlc_module_end ()
/*****************************************************************************
* Local prototypes
*****************************************************************************/
typedef struct
{
demux_t *p_demux;
MediaSubsession *sub;
es_format_t fmt;
es_out_id_t *p_es;
bool b_muxed;
bool b_quicktime;
bool b_asf;
block_t *p_asf_block;
bool b_discard_trunc;
stream_t *p_out_muxed; /* for muxed stream */
uint8_t *p_buffer;
unsigned int i_buffer;
bool b_rtcp_sync;
char waiting;
int64_t i_pts;
double f_npt;
bool b_selected;
} live_track_t;
struct timeout_thread_t
{
demux_sys_t *p_sys;
vlc_thread_t handle;
bool b_handle_keep_alive;
};
class RTSPClientVlc;
struct demux_sys_t
{
char *p_sdp; /* XXX mallocated */
char *psz_path; /* URL-encoded path */
vlc_url_t url;
MediaSession *ms;
TaskScheduler *scheduler;
UsageEnvironment *env ;
RTSPClientVlc *rtsp;
/* */
int i_track;
live_track_t **track;
/* Weird formats */
asf_header_t asfh;
stream_t *p_out_asf;
bool b_real;
/* */
int64_t i_pcr; /* The clock */
double f_npt;
double f_npt_length;
double f_npt_start;
/* timeout thread information */
int i_timeout; /* session timeout value in seconds */
bool b_timeout_call;/* mark to send an RTSP call to prevent server timeout */
timeout_thread_t *p_timeout; /* the actual thread that makes sure we don't timeout */
/* */
bool b_force_mcast;
bool b_multicast; /* if one of the tracks is multicasted */
bool b_no_data; /* if we never received any data */
int i_no_data_ti; /* consecutive number of TaskInterrupt */
char event_rtsp;
char event_data;
bool b_get_param; /* Does the server support GET_PARAMETER */
bool b_paused; /* Are we paused? */
bool b_error;
int i_live555_ret; /* live555 callback return code */
float f_seek_request;/* In case we receive a seek request while paused*/
};
class RTSPClientVlc : public RTSPClient
{
public:
RTSPClientVlc( UsageEnvironment& env, char const* rtspURL, int verbosityLevel,
char const* applicationName, portNumBits tunnelOverHTTPPortNum,
demux_sys_t *p_sys) :
RTSPClient( env, rtspURL, verbosityLevel, applicationName,
tunnelOverHTTPPortNum
#if LIVEMEDIA_LIBRARY_VERSION_INT >= 1373932800
, -1
#endif
)
{
this->p_sys = p_sys;
}
demux_sys_t *p_sys;
};
static int Demux ( demux_t * );
static int Control( demux_t *, int, va_list );
static int Connect ( demux_t * );
static int SessionsSetup( demux_t * );
static int Play ( demux_t *);
static int ParseASF ( demux_t * );
static int RollOverTcp ( demux_t * );
static void StreamRead ( void *, unsigned int, unsigned int,
struct timeval, unsigned int );
static void StreamClose ( void * );
static void TaskInterruptData( void * );
static void TaskInterruptRTSP( void * );
static void* TimeoutPrevention( void * );
static unsigned char* parseH264ConfigStr( char const* configStr,
unsigned int& configSize );
static unsigned char* parseVorbisConfigStr( char const* configStr,
unsigned int& configSize );
/*****************************************************************************
* DemuxOpen:
*****************************************************************************/
static int Open ( vlc_object_t *p_this )
{
demux_t *p_demux = (demux_t*)p_this;
demux_sys_t *p_sys = NULL;
int i_return;
int i_error = VLC_EGENERIC;
if( p_demux->s )
{
/* See if it looks like a SDP
v, o, s fields are mandatory and in this order */
const uint8_t *p_peek;
if( stream_Peek( p_demux->s, &p_peek, 7 ) < 7 ) return VLC_EGENERIC;
if( memcmp( p_peek, "v=0\r\n", 5 ) &&
memcmp( p_peek, "v=0\n", 4 ) &&
( p_peek[0] < 'a' || p_peek[0] > 'z' || p_peek[1] != '=' ) )
{
return VLC_EGENERIC;
}
}
p_demux->pf_demux = Demux;
p_demux->pf_control= Control;
p_demux->p_sys = p_sys = (demux_sys_t*)calloc( 1, sizeof( demux_sys_t ) );
if( !p_sys ) return VLC_ENOMEM;
msg_Dbg( p_demux, "version " LIVEMEDIA_LIBRARY_VERSION_STRING );
TAB_INIT( p_sys->i_track, p_sys->track );
p_sys->f_npt = 0.;
p_sys->f_npt_start = 0.;
p_sys->f_npt_length = 0.;
p_sys->b_no_data = true;
p_sys->psz_path = strdup( p_demux->psz_location );
p_sys->b_force_mcast = var_InheritBool( p_demux, "rtsp-mcast" );
p_sys->f_seek_request = -1;
/* parse URL for rtsp://[user:[passwd]@]serverip:port/options */
vlc_UrlParse( &p_sys->url, p_sys->psz_path, 0 );
if( ( p_sys->scheduler = BasicTaskScheduler::createNew() ) == NULL )
{
msg_Err( p_demux, "BasicTaskScheduler::createNew failed" );
goto error;
}
if( !( p_sys->env = BasicUsageEnvironment::createNew(*p_sys->scheduler) ) )
{
msg_Err( p_demux, "BasicUsageEnvironment::createNew failed" );
goto error;
}
if( strcasecmp( p_demux->psz_access, "sdp" ) )
{
char *p = p_sys->psz_path;
while( (p = strchr( p, ' ' )) != NULL ) *p = '+';
}
if( strcasecmp( p_demux->psz_access, "satip" ) == 0 )
{
asprintf(&p_sys->p_sdp, "v=0\r\n"
"o=- 0 %s\r\n"
"s=SATIP:stream\r\n"
"i=SATIP RTP Stream\r\n"
"m=video 0 RTP/AVP 33\r\n"
"a=control:rtsp://%s\r\n\r\n",
p_sys->url.psz_host, p_sys->psz_path);
}
if( p_demux->s != NULL )
{
/* Gather the complete sdp file */
int i_sdp = 0;
int i_sdp_max = 1000;
uint8_t *p_sdp = (uint8_t*) malloc( i_sdp_max );
if( !p_sdp )
{
i_error = VLC_ENOMEM;
goto error;
}
for( ;; )
{
int i_read = stream_Read( p_demux->s, &p_sdp[i_sdp],
i_sdp_max - i_sdp - 1 );
if( i_read < 0 )
{
msg_Err( p_demux, "failed to read SDP" );
free( p_sdp );
goto error;
}
i_sdp += i_read;
if( i_read < i_sdp_max - i_sdp - 1 )
{
p_sdp[i_sdp] = '\0';
break;
}
i_sdp_max += 1000;
p_sdp = (uint8_t*)xrealloc( p_sdp, i_sdp_max );
}
p_sys->p_sdp = (char*)p_sdp;
}
else if( ( i_return = Connect( p_demux ) ) != VLC_SUCCESS )
{
msg_Err( p_demux, "Failed to connect with rtsp://%s", p_sys->psz_path );
goto error;
}
if( p_sys->p_sdp == NULL )
{
msg_Err( p_demux, "Failed to retrieve the RTSP Session Description" );
i_error = VLC_ENOMEM;
goto error;
}
if( ( i_return = SessionsSetup( p_demux ) ) != VLC_SUCCESS )
{
msg_Err( p_demux, "Nothing to play for rtsp://%s", p_sys->psz_path );
goto error;
}
if( p_sys->b_real ) goto error;
if( ( i_return = Play( p_demux ) ) != VLC_SUCCESS )
goto error;
if( p_sys->p_out_asf && ParseASF( p_demux ) )
{
msg_Err( p_demux, "cannot find a usable asf header" );
/* TODO Clean tracks */
goto error;
}
if( p_sys->i_track <= 0 )
goto error;
return VLC_SUCCESS;
error:
Close( p_this );
return i_error;
}
/*****************************************************************************
* DemuxClose:
*****************************************************************************/
static void Close( vlc_object_t *p_this )
{
demux_t *p_demux = (demux_t*)p_this;
demux_sys_t *p_sys = p_demux->p_sys;
if( p_sys->p_timeout )
{
vlc_cancel( p_sys->p_timeout->handle );
vlc_join( p_sys->p_timeout->handle, NULL );
free( p_sys->p_timeout );
}
if( p_sys->rtsp && p_sys->ms ) p_sys->rtsp->sendTeardownCommand( *p_sys->ms, NULL );
if( p_sys->ms ) Medium::close( p_sys->ms );
if( p_sys->rtsp ) RTSPClient::close( p_sys->rtsp );
if( p_sys->env ) p_sys->env->reclaim();
for( int i = 0; i < p_sys->i_track; i++ )
{
live_track_t *tk = p_sys->track[i];
if( tk->b_muxed ) stream_Delete( tk->p_out_muxed );
es_format_Clean( &tk->fmt );
free( tk->p_buffer );
free( tk );
}
TAB_CLEAN( p_sys->i_track, p_sys->track );
if( p_sys->p_out_asf ) stream_Delete( p_sys->p_out_asf );
delete p_sys->scheduler;
free( p_sys->p_sdp );
free( p_sys->psz_path );
vlc_UrlClean( &p_sys->url );
free( p_sys );
}
static inline const char *strempty( const char *s ) { return s?s:""; }
static inline Boolean toBool( bool b ) { return b?True:False; } // silly, no?
static void default_live555_callback( RTSPClient* client, int result_code, char* result_string )
{
RTSPClientVlc *client_vlc = static_cast<RTSPClientVlc *> ( client );
demux_sys_t *p_sys = client_vlc->p_sys;
delete []result_string;
p_sys->i_live555_ret = result_code;
p_sys->b_error = p_sys->i_live555_ret != 0;
p_sys->event_rtsp = 1;
}
/* return true if the RTSP command succeeded */
static bool wait_Live555_response( demux_t *p_demux, int i_timeout = 0 /* ms */ )
{
TaskToken task;
demux_sys_t * p_sys = p_demux->p_sys;
p_sys->event_rtsp = 0;
if( i_timeout > 0 )
{
/* Create a task that will be called if we wait more than timeout ms */
task = p_sys->scheduler->scheduleDelayedTask( i_timeout*1000,
TaskInterruptRTSP,
p_demux );
}
p_sys->event_rtsp = 0;
p_sys->b_error = true;
p_sys->i_live555_ret = 0;
p_sys->scheduler->doEventLoop( &p_sys->event_rtsp );
//here, if b_error is true and i_live555_ret = 0 we didn't receive a response
if( i_timeout > 0 )
{
/* remove the task */
p_sys->scheduler->unscheduleDelayedTask( task );
}
return !p_sys->b_error;
}
static void continueAfterDESCRIBE( RTSPClient* client, int result_code,
char* result_string )
{
RTSPClientVlc *client_vlc = static_cast<RTSPClientVlc *> ( client );
demux_sys_t *p_sys = client_vlc->p_sys;
p_sys->i_live555_ret = result_code;
if ( result_code == 0 )
{
char* sdpDescription = result_string;
free( p_sys->p_sdp );
p_sys->p_sdp = NULL;
if( sdpDescription )
{
p_sys->p_sdp = strdup( sdpDescription );
p_sys->b_error = false;
}
}
else
p_sys->b_error = true;
delete[] result_string;
p_sys->event_rtsp = 1;
}
static void continueAfterOPTIONS( RTSPClient* client, int result_code,
char* result_string )
{
RTSPClientVlc *client_vlc = static_cast<RTSPClientVlc *> (client);
demux_sys_t *p_sys = client_vlc->p_sys;
p_sys->b_get_param =
// If OPTIONS fails, assume GET_PARAMETER is not supported but
// still continue on with the stream. Some servers (foscam)
// return 501/not implemented for OPTIONS.
result_code == 0
&& result_string != NULL
&& strstr( result_string, "GET_PARAMETER" ) != NULL;
if( p_sys->p_sdp == NULL )
{
client->sendDescribeCommand( continueAfterDESCRIBE );
}
else
{
p_sys->b_error = false;
p_sys->event_rtsp = 1;
}
delete[] result_string;
}
/*****************************************************************************
* Connect: connects to the RTSP server to setup the session DESCRIBE
*****************************************************************************/
static int Connect( demux_t *p_demux )
{
demux_sys_t *p_sys = p_demux->p_sys;
Authenticator authenticator;
char *psz_user = NULL;
char *psz_pwd = NULL;
char *psz_url = NULL;
int i_http_port = 0;
int i_ret = VLC_SUCCESS;
const int i_timeout = var_InheritInteger( p_demux, "ipv4-timeout" );
/* Get the user name and password */
if( p_sys->url.psz_username || p_sys->url.psz_password )
{
/* Create the URL by stripping away the username/password part */
if( p_sys->url.i_port == 0 )
p_sys->url.i_port = 554;
if( asprintf( &psz_url, "rtsp://%s:%d%s",
strempty( p_sys->url.psz_host ),
p_sys->url.i_port,
strempty( p_sys->url.psz_path ) ) == -1 )
return VLC_ENOMEM;
psz_user = strdup( strempty( p_sys->url.psz_username ) );
psz_pwd = strdup( strempty( p_sys->url.psz_password ) );
}
else
{
if( asprintf( &psz_url, "rtsp://%s", p_sys->psz_path ) == -1 )
return VLC_ENOMEM;
psz_user = var_InheritString( p_demux, "rtsp-user" );
psz_pwd = var_InheritString( p_demux, "rtsp-pwd" );
}
createnew:
if( !vlc_object_alive (p_demux) )
{
i_ret = VLC_EGENERIC;
goto bailout;
}
if( var_CreateGetBool( p_demux, "rtsp-http" ) )
i_http_port = var_InheritInteger( p_demux, "rtsp-http-port" );
p_sys->rtsp = new RTSPClientVlc( *p_sys->env, psz_url,
var_InheritInteger( p_demux, "verbose" ) > 1 ? 1 : 0,
"LibVLC/" VERSION, i_http_port, p_sys );
if( !p_sys->rtsp )
{
msg_Err( p_demux, "RTSPClient::createNew failed (%s)",
p_sys->env->getResultMsg() );
i_ret = VLC_EGENERIC;
goto bailout;
}
/* Kasenna enables KeepAlive by analysing the User-Agent string.
* Appending _KA to the string should be enough to enable this feature,
* however, there is a bug where the _KA doesn't get parsed from the
* default User-Agent as created by VLC/Live555 code. This is probably due
* to spaces in the string or the string being too long. Here we override
* the default string with a more compact version.
*/
if( var_InheritBool( p_demux, "rtsp-kasenna" ))
{
p_sys->rtsp->setUserAgentString( "VLC_MEDIA_PLAYER_KA" );
}
describe:
authenticator.setUsernameAndPassword( psz_user, psz_pwd );
p_sys->rtsp->sendOptionsCommand( &continueAfterOPTIONS, &authenticator );
if( !wait_Live555_response( p_demux, i_timeout ) )
{
int i_code = p_sys->i_live555_ret;
if( i_code == 401 )
{
msg_Dbg( p_demux, "authentication failed" );
free( psz_user );
free( psz_pwd );
dialog_Login( p_demux, &psz_user, &psz_pwd,
_("RTSP authentication"), "%s",
_("Please enter a valid login name and a password.") );
if( psz_user != NULL && psz_pwd != NULL )
{
msg_Dbg( p_demux, "retrying with user=%s", psz_user );
goto describe;
}
}
else if( i_code > 0 && i_code != 404 && !var_GetBool( p_demux, "rtsp-http" ) )
{
/* Perhaps a firewall is being annoying. Try HTTP tunneling mode */
msg_Dbg( p_demux, "we will now try HTTP tunneling mode" );
var_SetBool( p_demux, "rtsp-http", true );
if( p_sys->rtsp ) RTSPClient::close( p_sys->rtsp );
p_sys->rtsp = NULL;
goto createnew;
}
else
{
if( i_code == 0 )
msg_Dbg( p_demux, "connection timeout" );
else
{
msg_Dbg( p_demux, "connection error %d", i_code );
if( i_code == 403 )
dialog_Fatal( p_demux, _("RTSP connection failed"),
_("Access to the stream is denied by the server configuration.") );
}
if( p_sys->rtsp ) RTSPClient::close( p_sys->rtsp );
p_sys->rtsp = NULL;
}
i_ret = VLC_EGENERIC;
}
bailout:
/* malloc-ated copy */
free( psz_url );
free( psz_user );
free( psz_pwd );
return i_ret;
}
/*****************************************************************************
* SessionsSetup: prepares the subsessions and does the SETUP
*****************************************************************************/
static int SessionsSetup( demux_t *p_demux )
{
demux_sys_t *p_sys = p_demux->p_sys;
MediaSubsessionIterator *iter = NULL;
MediaSubsession *sub = NULL;
bool b_rtsp_tcp;
int i_client_port;
int i_return = VLC_SUCCESS;
unsigned int i_receive_buffer = 0;
int i_frame_buffer = DEFAULT_FRAME_BUFFER_SIZE;
unsigned const thresh = 200000; /* RTP reorder threshold .2 second (default .1) */
const char *p_sess_lang = NULL;
const char *p_lang;
b_rtsp_tcp = var_CreateGetBool( p_demux, "rtsp-tcp" ) ||
var_GetBool( p_demux, "rtsp-http" );
i_client_port = var_InheritInteger( p_demux, "rtp-client-port" );
/* Create the session from the SDP */
if( !( p_sys->ms = MediaSession::createNew( *p_sys->env, p_sys->p_sdp ) ) )
{
msg_Err( p_demux, "Could not create the RTSP Session: %s",
p_sys->env->getResultMsg() );
return VLC_EGENERIC;
}
if( strcmp( p_sys->p_sdp, "m=" ) != 0 )
{
const char *p_sess_attr_end;
p_sess_attr_end = strstr( p_sys->p_sdp, "\nm=" );
if( !p_sess_attr_end )
p_sess_attr_end = strstr( p_sys->p_sdp, "\rm=" );
p_sess_lang = p_sess_attr_end ? strstr( p_sys->p_sdp, "a=lang:" ) : NULL;
if( p_sess_lang &&
p_sess_lang - p_sys->p_sdp > p_sess_attr_end - p_sys->p_sdp )
p_sess_lang = NULL;
}
/* Initialise each media subsession */
iter = new MediaSubsessionIterator( *p_sys->ms );
while( ( sub = iter->next() ) != NULL )
{
Boolean bInit;
live_track_t *tk;
/* Value taken from mplayer */
if( !strcmp( sub->mediumName(), "audio" ) )
i_receive_buffer = 100000;
else if( !strcmp( sub->mediumName(), "video" ) )
{
int i_var_buf_size = var_InheritInteger( p_demux, "rtsp-frame-buffer-size" );
if( i_var_buf_size > 0 )
i_frame_buffer = i_var_buf_size;
i_receive_buffer = 2000000;
}
else if( !strcmp( sub->mediumName(), "text" ) )
;
else continue;
if( i_client_port != -1 )
{
sub->setClientPortNum( i_client_port );
i_client_port += 2;
}
if( strcasestr( sub->codecName(), "REAL" ) )
{
msg_Info( p_demux, "real codec detected, using real-RTSP instead" );
p_sys->b_real = true; /* This is a problem, we'll handle it later */
continue;
}
if( !strcmp( sub->codecName(), "X-ASF-PF" ) )
bInit = sub->initiate( 0 );
else
bInit = sub->initiate();
if( !bInit )
{
msg_Warn( p_demux, "RTP subsession '%s/%s' failed (%s)",
sub->mediumName(), sub->codecName(),
p_sys->env->getResultMsg() );
}
else
{
if( sub->rtpSource() != NULL )
{
int fd = sub->rtpSource()->RTPgs()->socketNum();
/* Increase the buffer size */
if( i_receive_buffer > 0 )
increaseReceiveBufferTo( *p_sys->env, fd, i_receive_buffer );
/* Increase the RTP reorder timebuffer just a bit */
sub->rtpSource()->setPacketReorderingThresholdTime(thresh);
}
msg_Dbg( p_demux, "RTP subsession '%s/%s'", sub->mediumName(),
sub->codecName() );
/* Issue the SETUP */
if( p_sys->rtsp )
{
p_sys->rtsp->sendSetupCommand( *sub, default_live555_callback, False,
toBool( b_rtsp_tcp ),
toBool( p_sys->b_force_mcast && !b_rtsp_tcp ) );
if( !wait_Live555_response( p_demux ) )
{
/* if we get an unsupported transport error, toggle TCP
* use and try again */
if( p_sys->i_live555_ret == 461 )
p_sys->rtsp->sendSetupCommand( *sub, default_live555_callback, False,
!toBool( b_rtsp_tcp ), False );
if( p_sys->i_live555_ret != 461 || !wait_Live555_response( p_demux ) )
{
msg_Err( p_demux, "SETUP of'%s/%s' failed %s",
sub->mediumName(), sub->codecName(),
p_sys->env->getResultMsg() );
continue;
}
else
{
var_SetBool( p_demux, "rtsp-tcp", true );
b_rtsp_tcp = true;
}
}
}
/* Check if we will receive data from this subsession for
* this track */
if( sub->readSource() == NULL ) continue;
if( !p_sys->b_multicast )
{
/* We need different rollover behaviour for multicast */
p_sys->b_multicast = IsMulticastAddress( sub->connectionEndpointAddress() );
}
tk = (live_track_t*)malloc( sizeof( live_track_t ) );
if( !tk )
{
delete iter;
return VLC_ENOMEM;
}
tk->p_demux = p_demux;
tk->sub = sub;
tk->p_es = NULL;
tk->b_quicktime = false;
tk->b_asf = false;
tk->p_asf_block = NULL;
tk->b_muxed = false;
tk->b_discard_trunc = false;
tk->p_out_muxed = NULL;
tk->waiting = 0;
tk->b_rtcp_sync = false;
tk->i_pts = VLC_TS_INVALID;
tk->f_npt = 0.;
tk->b_selected = true;
tk->i_buffer = i_frame_buffer;
tk->p_buffer = (uint8_t *)malloc( i_frame_buffer );
if( !tk->p_buffer )
{
free( tk );
delete iter;
return VLC_ENOMEM;
}
/* Value taken from mplayer */
if( !strcmp( sub->mediumName(), "audio" ) )
{
es_format_Init( &tk->fmt, AUDIO_ES, VLC_FOURCC('u','n','d','f') );
tk->fmt.audio.i_channels = sub->numChannels();
tk->fmt.audio.i_rate = sub->rtpTimestampFrequency();
if( !strcmp( sub->codecName(), "MPA" ) ||
!strcmp( sub->codecName(), "MPA-ROBUST" ) ||
!strcmp( sub->codecName(), "X-MP3-DRAFT-00" ) )
{
tk->fmt.i_codec = VLC_CODEC_MPGA;
tk->fmt.audio.i_rate = 0;
}
else if( !strcmp( sub->codecName(), "AC3" ) )
{
tk->fmt.i_codec = VLC_CODEC_A52;
tk->fmt.audio.i_rate = 0;
}
else if( !strcmp( sub->codecName(), "L16" ) )
{
tk->fmt.i_codec = VLC_CODEC_S16B;
tk->fmt.audio.i_bitspersample = 16;
}
else if( !strcmp( sub->codecName(), "L20" ) )
{
tk->fmt.i_codec = VLC_CODEC_S20B;
tk->fmt.audio.i_bitspersample = 20;
}
else if( !strcmp( sub->codecName(), "L24" ) )
{
tk->fmt.i_codec = VLC_CODEC_S24B;
tk->fmt.audio.i_bitspersample = 24;
}
else if( !strcmp( sub->codecName(), "L8" ) )
{
tk->fmt.i_codec = VLC_CODEC_U8;
tk->fmt.audio.i_bitspersample = 8;
}
else if( !strcmp( sub->codecName(), "DAT12" ) )
{
tk->fmt.i_codec = VLC_CODEC_DAT12;
tk->fmt.audio.i_bitspersample = 12;
}
else if( !strcmp( sub->codecName(), "PCMU" ) )
{
tk->fmt.i_codec = VLC_CODEC_MULAW;
tk->fmt.audio.i_bitspersample = 8;
}
else if( !strcmp( sub->codecName(), "PCMA" ) )
{
tk->fmt.i_codec = VLC_CODEC_ALAW;
tk->fmt.audio.i_bitspersample = 8;
}
else if( !strncmp( sub->codecName(), "G726", 4 ) )
{
tk->fmt.i_codec = VLC_CODEC_ADPCM_G726;
tk->fmt.audio.i_rate = 8000;
tk->fmt.audio.i_channels = 1;
if( !strcmp( sub->codecName()+5, "40" ) )
tk->fmt.i_bitrate = 40000;
else if( !strcmp( sub->codecName()+5, "32" ) )
tk->fmt.i_bitrate = 32000;
else if( !strcmp( sub->codecName()+5, "24" ) )
tk->fmt.i_bitrate = 24000;
else if( !strcmp( sub->codecName()+5, "16" ) )
tk->fmt.i_bitrate = 16000;
}
else if( !strcmp( sub->codecName(), "AMR" ) )
{
tk->fmt.i_codec = VLC_CODEC_AMR_NB;
}
else if( !strcmp( sub->codecName(), "AMR-WB" ) )
{
tk->fmt.i_codec = VLC_CODEC_AMR_WB;
}
else if( !strcmp( sub->codecName(), "MP4A-LATM" ) )
{
unsigned int i_extra;
uint8_t *p_extra;
tk->fmt.i_codec = VLC_CODEC_MP4A;
if( ( p_extra = parseStreamMuxConfigStr( sub->fmtp_config(),
i_extra ) ) )
{
tk->fmt.i_extra = i_extra;
tk->fmt.p_extra = xmalloc( i_extra );
memcpy( tk->fmt.p_extra, p_extra, i_extra );
delete[] p_extra;
}
/* Because the "faad" decoder does not handle the LATM
* data length field at the start of each returned LATM
* frame, tell the RTP source to omit. */
((MPEG4LATMAudioRTPSource*)sub->rtpSource())->omitLATMDataLengthField();
}
else if( !strcmp( sub->codecName(), "MPEG4-GENERIC" ) )
{
unsigned int i_extra;
uint8_t *p_extra;
tk->fmt.i_codec = VLC_CODEC_MP4A;
if( ( p_extra = parseGeneralConfigStr( sub->fmtp_config(),
i_extra ) ) )
{
tk->fmt.i_extra = i_extra;
tk->fmt.p_extra = xmalloc( i_extra );
memcpy( tk->fmt.p_extra, p_extra, i_extra );
delete[] p_extra;
}
}
else if( !strcmp( sub->codecName(), "X-ASF-PF" ) )
{
tk->b_asf = true;
if( p_sys->p_out_asf == NULL )
p_sys->p_out_asf = stream_DemuxNew( p_demux, "asf",
p_demux->out );
}
else if( !strcmp( sub->codecName(), "X-QT" ) ||
!strcmp( sub->codecName(), "X-QUICKTIME" ) )
{
tk->b_quicktime = true;
}
else if( !strcmp( sub->codecName(), "SPEEX" ) )
{
tk->fmt.i_codec = VLC_FOURCC( 's', 'p', 'x', 'r' );
}
else if( !strcmp( sub->codecName(), "VORBIS" ) )
{
tk->fmt.i_codec = VLC_CODEC_VORBIS;
unsigned int i_extra;
unsigned char *p_extra;
if( ( p_extra=parseVorbisConfigStr( sub->fmtp_config(),
i_extra ) ) )
{
tk->fmt.i_extra = i_extra;
tk->fmt.p_extra = p_extra;
}
else
msg_Warn( p_demux,"Missing or unsupported vorbis header." );
}
else if( !strcmp( sub->codecName(), "OPUS" ) )
{
tk->fmt.i_codec = VLC_CODEC_OPUS;
}
}
else if( !strcmp( sub->mediumName(), "video" ) )
{
es_format_Init( &tk->fmt, VIDEO_ES, VLC_FOURCC('u','n','d','f') );
if( !strcmp( sub->codecName(), "MPV" ) )