vlc live555源码,该源码摘自v2.1.1的vlc版本
/*****************************************************************************
* live555.cpp : LIVE555 Streaming Media support.
*****************************************************************************
* Copyright (C) 2003-2007 VLC authors and VideoLAN
* $Id: 7a1ab161e062082ab20fd3f9c24b0421c71570e2 $
*
* Authors: Laurent Aimar <[email protected]>
* Derk-Jan Hartman <hartman at videolan. org>
* Derk-Jan Hartman <djhartman at m2x .dot. nl> for M2X
* S茅bastien 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
*****************************************************************************/
/* For inttypes.h
* Note: config.h may include inttypes.h, so make sure we define this option
* early enough. */
#define __STDC_CONSTANT_MACROS 1
#define __STDC_LIMIT_MACROS 1
#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 ? */
}
using namespace std;
/*****************************************************************************
* 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_("RTSP user name")
#define USER_LONGTEXT N_("Sets the username for the connection, " \
"if no username or password are set in the url.")
#define PASS_TEXT N_("RTSP 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" )
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( 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( !vlc_object_alive (p_demux) )
{
free( p_sdp );
goto error;
}
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;
client->sendDescribeCommand( continueAfterDESCRIBE );
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) */
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;
}
/* Initialise each media subsession */
iter = new MediaSubsessionIterator( *p_sys->ms );
while( ( sub = iter->next() ) != NULL )
{
Boolean bInit;
live_track_t *tk;
if( !vlc_object_alive (p_demux) )
{
delete iter;
return VLC_EGENERIC;
}
/* 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' );
if ( tk->fmt.audio.i_rate == 0 )
{
msg_Warn( p_demux,"Using 8kHz as default sample rate." );
tk->fmt.audio.i_rate = 8000;
}
}
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->mediumName(), "video" ) )
{
es_format_Init( &tk->fmt, VIDEO_ES, VLC_FOURCC('u','n','d','f') );
if( !strcmp( sub->codecName(), "MPV" ) )
{
tk->fmt.i_codec = VLC_CODEC_MPGV;
tk->fmt.b_packetized = false;
}
else if( !strcmp( sub->codecName(), "H263" ) ||
!strcmp( sub->codecName(), "H263-1998" ) ||
!strcmp( sub->codecName(), "H263-2000" ) )
{
tk->fmt.i_codec = VLC_CODEC_H263;
}
else if( !strcmp( sub->codecName(), "H261" ) )
{
tk->fmt.i_codec = VLC_CODEC_H261;
}
else if( !strcmp( sub->codecName(), "H264" ) )
{
unsigned int i_extra = 0;
uint8_t *p_extra = NULL;
tk->fmt.i_codec = VLC_CODEC_H264;
tk->fmt.b_packetized = false;
if((p_extra=parseH264ConfigStr( sub->fmtp_spropparametersets(),
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(), "JPEG" ) )
{
tk->fmt.i_codec = VLC_CODEC_MJPG;
}
else if( !strcmp( sub->codecName(), "MP4V-ES" ) )
{
unsigned int i_extra;
uint8_t *p_extra;
tk->fmt.i_codec = VLC_CODEC_MP4V;
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-QT" ) ||
!strcmp( sub->codecName(), "X-QUICKTIME" ) ||
!strcmp( sub->codecName(), "X-QDM" ) ||
!strcmp( sub->codecName(), "X-SV3V-ES" ) ||
!strcmp( sub->codecName(), "X-SORENSONVIDEO" ) )
{
tk->b_quicktime = true;
}
else if( !strcmp( sub->codecName(), "MP2T" ) )
{
tk->b_muxed = true;
tk->p_out_muxed = stream_DemuxNew( p_demux, "ts", p_demux->out );
}
else if( !strcmp( sub->codecName(), "MP2P" ) ||
!strcmp( sub->codecName(), "MP1S" ) )
{
tk->b_muxed = true;
tk->p_out_muxed = stream_DemuxNew( p_demux, "ps",
p_demux->out );
}
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(), "DV" ) )
{
tk->b_muxed = true;
tk->b_discard_trunc = true;
tk->p_out_muxed = stream_DemuxNew( p_demux, "rawdv",
p_demux->out );
}
else if( !strcmp( sub->codecName(), "VP8" ) )
{
tk->fmt.i_codec = VLC_CODEC_VP8;
}
}
else if( !strcmp( sub->mediumName(), "text" ) )
{
es_format_Init( &tk->fmt, SPU_ES, VLC_FOURCC('u','n','d','f') );
if( !strcmp( sub->codecName(), "T140" ) )
{
tk->fmt.i_codec = VLC_CODEC_ITU_T140;
}
}
if( !tk->b_quicktime && !tk->b_muxed && !tk->b_asf )
{
tk->p_es = es_out_Add( p_demux->out, &tk->fmt );
}
if( sub->rtcpInstance() != NULL )
{
sub->rtcpInstance()->setByeHandler( StreamClose, tk );
}
if( tk->p_es || tk->b_quicktime || ( tk->b_muxed && tk->p_out_muxed ) ||
( tk->b_asf && p_sys->p_out_asf ) )
{
TAB_APPEND_CAST( (live_track_t **), p_sys->i_track, p_sys->track, tk );
}
else
{
/* BUG ??? */
msg_Err( p_demux, "unusable RTSP track. this should not happen" );
es_format_Clean( &tk->fmt );
free( tk );
}
}
}
delete iter;
if( p_sys->i_track <= 0 ) i_return = VLC_EGENERIC;
/* Retrieve the starttime if possible */
p_sys->f_npt_start = p_sys->ms->playStartTime();
/* Retrieve the duration if possible */
p_sys->f_npt_length = p_sys->ms->playEndTime();
/* */
msg_Dbg( p_demux, "setup start: %f stop:%f", p_sys->f_npt_start, p_sys->f_npt_length );
/* */
p_sys->b_no_data = true;
p_sys->i_no_data_ti = 0;
return i_return;
}
/*****************************************************************************
* Play: starts the actual playback of the stream
*****************************************************************************/
static int Play( demux_t *p_demux )
{
demux_sys_t *p_sys = p_demux->p_sys;
if( p_sys->rtsp )
{
/* The PLAY */
p_sys->rtsp->sendPlayCommand( *p_sys->ms, default_live555_callback, p_sys->f_npt_start, -1, 1 );
if( !wait_Live555_response(p_demux) )
{
msg_Err( p_demux, "RTSP PLAY failed %s", p_sys->env->getResultMsg() );
return VLC_EGENERIC;
}
/* Retrieve the timeout value and set up a timeout prevention thread */
p_sys->i_timeout = p_sys->rtsp->sessionTimeoutParameter();
if( p_sys->i_timeout <= 0 )
p_sys->i_timeout = 60; /* default value from RFC2326 */
/* start timeout-thread only if GET_PARAMETER is supported by the server */
/* or start it if wmserver dialect, since they don't report that GET_PARAMETER is supported correctly */
if( !p_sys->p_timeout && ( p_sys->b_get_param || var_InheritBool( p_demux, "rtsp-wmserver" ) ) )
{
msg_Dbg( p_demux, "We have a timeout of %d seconds", p_sys->i_timeout );
p_sys->p_timeout = (timeout_thread_t *)malloc( sizeof(timeout_thread_t) );
if( p_sys->p_timeout )
{
memset( p_sys->p_timeout, 0, sizeof(timeout_thread_t) );
p_sys->p_timeout->p_sys = p_demux->p_sys; /* lol, object recursion :D */
if( vlc_clone( &p_sys->p_timeout->handle, TimeoutPrevention,
p_sys->p_timeout, VLC_THREAD_PRIORITY_LOW ) )
{
msg_Err( p_demux, "cannot spawn liveMedia timeout thread" );
free( p_sys->p_timeout );
p_sys->p_timeout = NULL;
}
else
msg_Dbg( p_demux, "spawned timeout thread" );
}
else
msg_Err( p_demux, "cannot spawn liveMedia timeout thread" );
}
}
p_sys->i_pcr = 0;
/* Retrieve the starttime if possible */
p_sys->f_npt_start = p_sys->ms->playStartTime();
if( p_sys->ms->playEndTime() > 0 )
p_sys->f_npt_length = p_sys->ms->playEndTime();
msg_Dbg( p_demux, "play start: %f stop:%f", p_sys->f_npt_start, p_sys->f_npt_length );
return VLC_SUCCESS;
}
/*****************************************************************************
* Demux:
*****************************************************************************/
static int Demux( demux_t *p_demux )
{
demux_sys_t *p_sys = p_demux->p_sys;
TaskToken task;
bool b_send_pcr = true;
int64_t i_pcr = 0;
int i;
/* Check if we need to send the server a Keep-A-Live signal */
if( p_sys->b_timeout_call && p_sys->rtsp && p_sys->ms )
{
char *psz_bye = NULL;
p_sys->rtsp->sendGetParameterCommand( *p_sys->ms, NULL, psz_bye );
p_sys->b_timeout_call = false;
}
for( i = 0; i < p_sys->i_track; i++ )
{
live_track_t *tk = p_sys->track[i];
if( tk->p_es )
{
bool b;
es_out_Control( p_demux->out, ES_OUT_GET_ES_STATE, tk->p_es, &b );
if( !b && tk->b_selected )
{
tk->b_selected = false;
p_sys->rtsp->sendTeardownCommand( *tk->sub, NULL );
}
else if( b && !tk->b_selected)
{
bool b_rtsp_tcp = var_GetBool( p_demux, "rtsp-tcp" ) ||
var_GetBool( p_demux, "rtsp-http" );
p_sys->rtsp->sendSetupCommand( *tk->sub, default_live555_callback, False,
toBool( b_rtsp_tcp ),
toBool( p_sys->b_force_mcast && !b_rtsp_tcp ) );
if( !wait_Live555_response( p_demux ) )
{
msg_Err( p_demux, "SETUP of'%s/%s' failed %s",
tk->sub->mediumName(), tk->sub->codecName(),
p_sys->env->getResultMsg() );
}
else
{
p_sys->rtsp->sendPlayCommand( *tk->sub, default_live555_callback, -1, -1, p_sys->ms->scale() );
if( !wait_Live555_response(p_demux) )
{
msg_Err( p_demux, "RTSP PLAY failed %s", p_sys->env->getResultMsg() );
p_sys->rtsp->sendTeardownCommand( *tk->sub, NULL );
}
else
tk->b_selected = true;
}
if( !tk->b_selected )
es_out_Control( p_demux->out, ES_OUT_SET_ES_STATE, tk->p_es, false );
}
}
if( tk->b_asf || tk->b_muxed )
b_send_pcr = false;
#if 0
if( i_pcr == 0 )
{
i_pcr = tk->i_pts;
}
else if( tk->i_pts != 0 && i_pcr > tk->i_pts )
{
i_pcr = tk->i_pts ;
}
#endif
}
if( p_sys->i_pcr > 0 )
{
if( b_send_pcr )
es_out_Control( p_demux->out, ES_OUT_SET_PCR, 1 + p_sys->i_pcr );
}
/* First warn we want to read data */
p_sys->event_data = 0;
for( i = 0; i < p_sys->i_track; i++ )
{
live_track_t *tk = p_sys->track[i];
if( tk->waiting == 0 )
{
tk->waiting = 1;
tk->sub->readSource()->getNextFrame( tk->p_buffer, tk->i_buffer,
StreamRead, tk, StreamClose, tk );
}
}
/* Create a task that will be called if we wait more than 300ms */
task = p_sys->scheduler->scheduleDelayedTask( 300000, TaskInterruptData, p_demux );
/* Do the read */
p_sys->scheduler->doEventLoop( &p_sys->event_data );
/* remove the task */
p_sys->scheduler->unscheduleDelayedTask( task );
/* Check for gap in pts value */
for( i = 0; i < p_sys->i_track; i++ )
{
live_track_t *tk = p_sys->track[i];
if( !tk->b_muxed && !tk->b_rtcp_sync &&
tk->sub->rtpSource() && tk->sub->rtpSource()->hasBeenSynchronizedUsingRTCP() )
{
msg_Dbg( p_demux, "tk->rtpSource->hasBeenSynchronizedUsingRTCP()" );
es_out_Control( p_demux->out, ES_OUT_RESET_PCR );
tk->b_rtcp_sync = true;
/* reset PCR */
tk->i_pts = VLC_TS_INVALID;
tk->f_npt = 0.;
p_sys->i_pcr = 0;
p_sys->f_npt = 0.;
i_pcr = 0;
}
}
if( p_sys->b_multicast && p_sys->b_no_data &&
( p_sys->i_no_data_ti > 120 ) )
{
/* FIXME Make this configurable
msg_Err( p_demux, "no multicast data received in 36s, aborting" );
return 0;
*/
}
else if( !p_sys->b_multicast && !p_sys->b_paused &&
p_sys->b_no_data && ( p_sys->i_no_data_ti > 34 ) )
{
bool b_rtsp_tcp = var_GetBool( p_demux, "rtsp-tcp" ) ||
var_GetBool( p_demux, "rtsp-http" );
if( !b_rtsp_tcp && p_sys->rtsp && p_sys->ms )
{
msg_Warn( p_demux, "no data received in 10s. Switching to TCP" );
if( RollOverTcp( p_demux ) )
{
msg_Err( p_demux, "TCP rollover failed, aborting" );
return 0;
}
return 1;
}
msg_Err( p_demux, "no data received in 10s, aborting" );
return 0;
}
else if( !p_sys->b_multicast && !p_sys->b_paused &&
( p_sys->i_no_data_ti > 34 ) )
{
/* EOF ? */
msg_Warn( p_demux, "no data received in 10s, eof ?" );
return 0;
}
return p_sys->b_error ? 0 : 1;
}
/*****************************************************************************
* Control:
*****************************************************************************/
static int Control( demux_t *p_demux, int i_query, va_list args )
{
demux_sys_t *p_sys = p_demux->p_sys;
int64_t *pi64, i64;
double *pf, f;
bool *pb, *pb2;
int *pi_int;
switch( i_query )
{
case DEMUX_GET_TIME:
pi64 = (int64_t*)va_arg( args, int64_t * );
if( p_sys->f_npt > 0 )
{
*pi64 = (int64_t)(p_sys->f_npt * 1000000.);
return VLC_SUCCESS;
}
return VLC_EGENERIC;
case DEMUX_GET_LENGTH:
pi64 = (int64_t*)va_arg( args, int64_t * );
if( p_sys->f_npt_length > 0 )
{
double d_length = p_sys->f_npt_length * 1000000.0;
if( d_length >= INT64_MAX )
*pi64 = INT64_MAX;
else
*pi64 = (int64_t)d_length;
return VLC_SUCCESS;
}
return VLC_EGENERIC;
case DEMUX_GET_POSITION:
pf = (double*)va_arg( args, double* );
if( (p_sys->f_npt_length > 0) && (p_sys->f_npt > 0) )
{
*pf = p_sys->f_npt / p_sys->f_npt_length;
return VLC_SUCCESS;
}
return VLC_EGENERIC;
case DEMUX_SET_POSITION:
case DEMUX_SET_TIME:
if( p_sys->rtsp && (p_sys->f_npt_length > 0) )
{
int i;
float time;
if( (i_query == DEMUX_SET_TIME) && (p_sys->f_npt > 0) )
{
i64 = (int64_t)va_arg( args, int64_t );
time = (float)(i64 / 1000000.0); /* in second */
}
else if( i_query == DEMUX_SET_TIME )
return VLC_EGENERIC;
else
{
f = (double)va_arg( args, double );
time = f * p_sys->f_npt_length; /* in second */
}
if( p_sys->b_paused )
{
p_sys->f_seek_request = time;
return VLC_SUCCESS;
}
p_sys->rtsp->sendPauseCommand( *p_sys->ms, default_live555_callback );
if( !wait_Live555_response( p_demux ) )
{
msg_Err( p_demux, "PAUSE before seek failed %s",
p_sys->env->getResultMsg() );
return VLC_EGENERIC;
}
p_sys->rtsp->sendPlayCommand( *p_sys->ms, default_live555_callback, time, -1, 1 );
if( !wait_Live555_response( p_demux ) )
{
msg_Err( p_demux, "seek PLAY failed %s",
p_sys->env->getResultMsg() );
return VLC_EGENERIC;
}
p_sys->i_pcr = 0;
for( i = 0; i < p_sys->i_track; i++ )
{
p_sys->track[i]->b_rtcp_sync = false;
p_sys->track[i]->i_pts = VLC_TS_INVALID;
}
/* Retrieve the starttime if possible */
p_sys->f_npt = p_sys->f_npt_start = p_sys->ms->playStartTime();
/* Retrieve the duration if possible */
if( p_sys->ms->playEndTime() > 0 )
p_sys->f_npt_length = p_sys->ms->playEndTime();
msg_Dbg( p_demux, "seek start: %f stop:%f", p_sys->f_npt_start, p_sys->f_npt_length );
return VLC_SUCCESS;
}
return VLC_EGENERIC;
/* Special for access_demux */
case DEMUX_CAN_PAUSE:
case DEMUX_CAN_SEEK:
pb = (bool*)va_arg( args, bool * );
if( p_sys->rtsp && p_sys->f_npt_length > 0 )
/* Not always true, but will be handled in SET_PAUSE_STATE */
*pb = true;
else
*pb = false;
return VLC_SUCCESS;
case DEMUX_CAN_CONTROL_PACE:
pb = (bool*)va_arg( args, bool * );
#if 1 /* Disable for now until we have a clock synchro algo
* which works with something else than MPEG over UDP */
*pb = false;
#else
*pb = true;
#endif
return VLC_SUCCESS;
case DEMUX_CAN_CONTROL_RATE:
pb = (bool*)va_arg( args, bool * );
pb2 = (bool*)va_arg( args, bool * );
*pb = (p_sys->rtsp != NULL) &&
(p_sys->f_npt_length > 0) &&
( !var_GetBool( p_demux, "rtsp-kasenna" ) ||
!var_GetBool( p_demux, "rtsp-wmserver" ) );
*pb2 = false;
return VLC_SUCCESS;
case DEMUX_SET_RATE:
{
double f_scale, f_old_scale;
if( !p_sys->rtsp || (p_sys->f_npt_length <= 0) ||
var_GetBool( p_demux, "rtsp-kasenna" ) ||
var_GetBool( p_demux, "rtsp-wmserver" ) )
return VLC_EGENERIC;
/* According to RFC 2326 p56 chapter 12.35 a RTSP server that
* supports Scale:
*
* "[...] should try to approximate the viewing rate, but
* may restrict the range of scale values that it supports.
* The response MUST contain the actual scale value chosen
* by the server."
*
* Scale = 1 indicates normal play
* Scale > 1 indicates fast forward
* Scale < 1 && Scale > 0 indicates slow motion
* Scale < 0 value indicates rewind
*/
pi_int = (int*)va_arg( args, int * );
f_scale = (double)INPUT_RATE_DEFAULT / (*pi_int);
f_old_scale = p_sys->ms->scale();
/* Passing -1 for the start and end time will mean liveMedia won't
* create a Range: section for the RTSP message. The server should
* pick up from the current position */
p_sys->rtsp->sendPlayCommand( *p_sys->ms, default_live555_callback, -1, -1, f_scale );
if( !wait_Live555_response( p_demux ) )
{
msg_Err( p_demux, "PLAY with Scale %0.2f failed %s", f_scale,
p_sys->env->getResultMsg() );
return VLC_EGENERIC;
}
if( p_sys->ms->scale() == f_old_scale )
{
msg_Err( p_demux, "no scale change using old Scale %0.2f",
p_sys->ms->scale() );
return VLC_EGENERIC;
}
/* ReSync the stream */
p_sys->f_npt_start = 0;
p_sys->i_pcr = 0;
p_sys->f_npt = 0.0;
*pi_int = (int)( INPUT_RATE_DEFAULT / p_sys->ms->scale() );
msg_Dbg( p_demux, "PLAY with new Scale %0.2f (%d)", p_sys->ms->scale(), (*pi_int) );
return VLC_SUCCESS;
}
case DEMUX_SET_PAUSE_STATE:
{
bool b_pause = (bool)va_arg( args, int );
if( p_sys->rtsp == NULL )
return VLC_EGENERIC;
if( b_pause == p_sys->b_paused )
return VLC_SUCCESS;
if( b_pause )
p_sys->rtsp->sendPauseCommand( *p_sys->ms, default_live555_callback );
else
p_sys->rtsp->sendPlayCommand( *p_sys->ms, default_live555_callback, p_sys->f_seek_request,
-1.0f, p_sys->ms->scale() );
if( !wait_Live555_response( p_demux ) )
{
msg_Err( p_demux, "PLAY or PAUSE failed %s", p_sys->env->getResultMsg() );
return VLC_EGENERIC;
}
p_sys->f_seek_request = -1;
p_sys->b_paused = b_pause;
/* When we Pause, we'll need the TimeoutPrevention thread to
* handle sending the "Keep Alive" message to the server.
* Unfortunately Live555 isn't thread safe and so can't
* do this normally while the main Demux thread is handling
* a live stream. We end up with the Timeout thread blocking
* waiting for a response from the server. So when we PAUSE
* we set a flag that the TimeoutPrevention function will check
* and if it's set, it will trigger the GET_PARAMETER message */
if( p_sys->b_paused && p_sys->p_timeout != NULL )
p_sys->p_timeout->b_handle_keep_alive = true;
else if( !p_sys->b_paused && p_sys->p_timeout != NULL )
p_sys->p_timeout->b_handle_keep_alive = false;
if( !p_sys->b_paused )
{
for( int i = 0; i < p_sys->i_track; i++ )
{
live_track_t *tk = p_sys->track[i];
tk->b_rtcp_sync = false;
tk->i_pts = VLC_TS_INVALID;
p_sys->i_pcr = 0;
es_out_Control( p_demux->out, ES_OUT_RESET_PCR );
}
}
/* Reset data received counter */
p_sys->i_no_data_ti = 0;
/* Retrieve the starttime if possible */
p_sys->f_npt_start = p_sys->ms->playStartTime();
/* Retrieve the duration if possible */
if( p_sys->ms->playEndTime() )
p_sys->f_npt_length = p_sys->ms->playEndTime();
msg_Dbg( p_demux, "pause start: %f stop:%f", p_sys->f_npt_start, p_sys->f_npt_length );
return VLC_SUCCESS;
}
case DEMUX_GET_TITLE_INFO:
case DEMUX_SET_TITLE:
case DEMUX_SET_SEEKPOINT:
return VLC_EGENERIC;
case DEMUX_GET_PTS_DELAY:
pi64 = (int64_t*)va_arg( args, int64_t * );
*pi64 = INT64_C(1000)
* var_InheritInteger( p_demux, "network-caching" );
return VLC_SUCCESS;
default:
return VLC_EGENERIC;
}
}
/*****************************************************************************
* RollOverTcp: reopen the rtsp into TCP mode
* XXX: ugly, a lot of code are duplicated from Open()
* This should REALLY be fixed
*****************************************************************************/
static int RollOverTcp( demux_t *p_demux )
{
demux_sys_t *p_sys = p_demux->p_sys;
int i, i_return;
var_SetBool( p_demux, "rtsp-tcp", true );
/* We close the old RTSP session */
p_sys->rtsp->sendTeardownCommand( *p_sys->ms, NULL );
Medium::close( p_sys->ms );
RTSPClient::close( p_sys->rtsp );
for( 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 );
if( tk->p_es ) es_out_Del( p_demux->out, tk->p_es );
if( tk->p_asf_block ) block_Release( tk->p_asf_block );
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 );
p_sys->ms = NULL;
p_sys->rtsp = NULL;
p_sys->b_no_data = true;
p_sys->i_no_data_ti = 0;
p_sys->p_out_asf = NULL;
/* Reopen rtsp client */
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" );
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( ( i_return = Play( p_demux ) ) != VLC_SUCCESS )
goto error;
return VLC_SUCCESS;
error:
return VLC_EGENERIC;
}
/*****************************************************************************
*
*****************************************************************************/
static block_t *StreamParseAsf( demux_t *p_demux, live_track_t *tk,
bool b_marker,
const uint8_t *p_data, unsigned i_size )
{
const unsigned i_packet_size = p_demux->p_sys->asfh.i_min_data_packet_size;
block_t *p_list = NULL;
while( i_size >= 4 )
{
unsigned i_flags = p_data[0];
unsigned i_length_offset = (p_data[1] << 16) |
(p_data[2] << 8) |
(p_data[3] );
bool b_length = i_flags & 0x40;
bool b_relative_ts = i_flags & 0x20;
bool b_duration = i_flags & 0x10;
bool b_location_id = i_flags & 0x08;
//msg_Dbg( p_demux, "ASF: marker=%d size=%d : %c=%d id=%d",
// b_marker, i_size, b_length ? 'L' : 'O', i_length_offset );
unsigned i_header_size = 4;
if( b_relative_ts )
i_header_size += 4;
if( b_duration )
i_header_size += 4;
if( b_location_id )
i_header_size += 4;
if( i_header_size > i_size )
{
msg_Warn( p_demux, "Invalid header size" );
break;
}
/* XXX
* When b_length is true, the streams I found do not seems to respect
* the documentation.
* From them, I have failed to find which choice between '__MIN()' or
* 'i_length_offset - i_header_size' is the right one.
*/
unsigned i_payload;
if( b_length )
i_payload = __MIN( i_length_offset, i_size - i_header_size);
else
i_payload = i_size - i_header_size;
if( !tk->p_asf_block )
{
tk->p_asf_block = block_Alloc( i_packet_size );
if( !tk->p_asf_block )
break;
tk->p_asf_block->i_buffer = 0;
}
unsigned i_offset = b_length ? 0 : i_length_offset;
if( i_offset == tk->p_asf_block->i_buffer && i_offset + i_payload <= i_packet_size )
{
memcpy( &tk->p_asf_block->p_buffer[i_offset], &p_data[i_header_size], i_payload );
tk->p_asf_block->i_buffer += i_payload;
if( b_marker )
{
/* We have a complete packet */
tk->p_asf_block->i_buffer = i_packet_size;
block_ChainAppend( &p_list, tk->p_asf_block );
tk->p_asf_block = NULL;
}
}
else
{
/* Reset on broken stream */
msg_Err( p_demux, "Broken packet detected (%d vs %zu or %d + %d vs %d)",
i_offset, tk->p_asf_block->i_buffer, i_offset, i_payload, i_packet_size);
tk->p_asf_block->i_buffer = 0;
}
/* */
p_data += i_header_size + i_payload;
i_size -= i_header_size + i_payload;
}
return p_list;
}
/*****************************************************************************
*
*****************************************************************************/
static void StreamRead( void *p_private, unsigned int i_size,
unsigned int i_truncated_bytes, struct timeval pts,
unsigned int duration )
{
VLC_UNUSED( duration );
live_track_t *tk = (live_track_t*)p_private;
demux_t *p_demux = tk->p_demux;
demux_sys_t *p_sys = p_demux->p_sys;
block_t *p_block;
//msg_Dbg( p_demux, "pts: %d", pts.tv_sec );
int64_t i_pts = (int64_t)pts.tv_sec * INT64_C(1000000) +
(int64_t)pts.tv_usec;
/* XXX Beurk beurk beurk Avoid having negative value XXX */
i_pts &= INT64_C(0x00ffffffffffffff);
/* Retrieve NPT for this pts */
tk->f_npt = tk->sub->getNormalPlayTime(pts);
if( tk->b_quicktime && tk->p_es == NULL )
{
QuickTimeGenericRTPSource *qtRTPSource =
(QuickTimeGenericRTPSource*)tk->sub->rtpSource();
QuickTimeGenericRTPSource::QTState &qtState = qtRTPSource->qtState;
uint8_t *sdAtom = (uint8_t*)&qtState.sdAtom[4];
/* Get codec informations from the quicktime atoms :
* http://developer.apple.com/quicktime/icefloe/dispatch026.html */
if( tk->fmt.i_cat == VIDEO_ES ) {
if( qtState.sdAtomSize < 16 + 32 )
{
/* invalid */
p_sys->event_data = 0xff;
tk->waiting = 0;
return;
}
tk->fmt.i_codec = VLC_FOURCC(sdAtom[0],sdAtom[1],sdAtom[2],sdAtom[3]);
tk->fmt.video.i_width = (sdAtom[28] << 8) | sdAtom[29];
tk->fmt.video.i_height = (sdAtom[30] << 8) | sdAtom[31];
if( tk->fmt.i_codec == VLC_FOURCC('a', 'v', 'c', '1') )
{
uint8_t *pos = (uint8_t*)qtRTPSource->qtState.sdAtom + 86;
uint8_t *endpos = (uint8_t*)qtRTPSource->qtState.sdAtom
+ qtRTPSource->qtState.sdAtomSize;
while (pos+8 < endpos) {
unsigned int atomLength = pos[0]<<24 | pos[1]<<16 | pos[2]<<8 | pos[3];
if( atomLength == 0 || atomLength > (unsigned int)(endpos-pos)) break;
if( memcmp(pos+4, "avcC", 4) == 0 &&
atomLength > 8 &&
atomLength <= INT_MAX )
{
tk->fmt.i_extra = atomLength-8;
tk->fmt.p_extra = xmalloc( tk->fmt.i_extra );
memcpy(tk->fmt.p_extra, pos+8, atomLength-8);
break;
}
pos += atomLength;
}
}
else
{
tk->fmt.i_extra = qtState.sdAtomSize - 16;
tk->fmt.p_extra = xmalloc( tk->fmt.i_extra );
memcpy( tk->fmt.p_extra, &sdAtom[12], tk->fmt.i_extra );
}
}
else {
if( qtState.sdAtomSize < 24 )
{
/* invalid */
p_sys->event_data = 0xff;
tk->waiting = 0;
return;
}
tk->fmt.i_codec = VLC_FOURCC(sdAtom[0],sdAtom[1],sdAtom[2],sdAtom[3]);
tk->fmt.audio.i_bitspersample = (sdAtom[22] << 8) | sdAtom[23];
}
tk->p_es = es_out_Add( p_demux->out, &tk->fmt );
}
#if 0
fprintf( stderr, "StreamRead size=%d pts=%lld\n",
i_size,
pts.tv_sec * 1000000LL + pts.tv_usec );
#endif
/* grow buffer if it looks like buffer is too small, but don't eat
* up all the memory on strange streams */
if( i_truncated_bytes > 0 )
{
if( tk->i_buffer < 2000000 )
{
void *p_tmp;
msg_Dbg( p_demux, "lost %d bytes", i_truncated_bytes );
msg_Dbg( p_demux, "increasing buffer size to %d", tk->i_buffer * 2 );
p_tmp = realloc( tk->p_buffer, tk->i_buffer * 2 );
if( p_tmp == NULL )
{
msg_Warn( p_demux, "realloc failed" );
}
else
{
tk->p_buffer = (uint8_t*)p_tmp;
tk->i_buffer *= 2;
}
}
if( tk->b_discard_trunc )
{
p_sys->event_data = 0xff;
tk->waiting = 0;
return;
}
}
assert( i_size <= tk->i_buffer );
if( tk->fmt.i_codec == VLC_CODEC_AMR_NB ||
tk->fmt.i_codec == VLC_CODEC_AMR_WB )
{
AMRAudioSource *amrSource = (AMRAudioSource*)tk->sub->readSource();
p_block = block_Alloc( i_size + 1 );
p_block->p_buffer[0] = amrSource->lastFrameHeader();
memcpy( p_block->p_buffer + 1, tk->p_buffer, i_size );
}
else if( tk->fmt.i_codec == VLC_CODEC_H261 )
{
H261VideoRTPSource *h261Source = (H261VideoRTPSource*)tk->sub->rtpSource();
uint32_t header = h261Source->lastSpecialHeader();
p_block = block_Alloc( i_size + 4 );
memcpy( p_block->p_buffer, &header, 4 );
memcpy( p_block->p_buffer + 4, tk->p_buffer, i_size );
if( tk->sub->rtpSource()->curPacketMarkerBit() )
p_block->i_flags |= BLOCK_FLAG_END_OF_FRAME;
}
else if( tk->fmt.i_codec == VLC_CODEC_H264 )
{
if( (tk->p_buffer[0] & 0x1f) >= 24 )
msg_Warn( p_demux, "unsupported NAL type for H264" );
/* Normal NAL type */
p_block = block_Alloc( i_size + 4 );
p_block->p_buffer[0] = 0x00;
p_block->p_buffer[1] = 0x00;
p_block->p_buffer[2] = 0x00;
p_block->p_buffer[3] = 0x01;
memcpy( &p_block->p_buffer[4], tk->p_buffer, i_size );
}
else if( tk->b_asf )
{
p_block = StreamParseAsf( p_demux, tk,
tk->sub->rtpSource()->curPacketMarkerBit(),
tk->p_buffer, i_size );
}
else
{
p_block = block_Alloc( i_size );
memcpy( p_block->p_buffer, tk->p_buffer, i_size );
}
if( p_sys->i_pcr < i_pts )
{
p_sys->i_pcr = i_pts;
}
/* Update our global npt value */
if( tk->f_npt > 0 &&
( tk->f_npt < p_sys->f_npt_length || p_sys->f_npt_length <= 0 ) )
p_sys->f_npt = tk->f_npt;
if( p_block )
{
if( !tk->b_muxed && !tk->b_asf )
{
if( i_pts != tk->i_pts )
p_block->i_pts = VLC_TS_0 + i_pts;
/*FIXME: for h264 you should check that packetization-mode=1 in sdp-file */
p_block->i_dts = ( tk->fmt.i_codec == VLC_CODEC_MPGV ) ? VLC_TS_INVALID : (VLC_TS_0 + i_pts);
}
if( tk->b_muxed )
stream_DemuxSend( tk->p_out_muxed, p_block );
else if( tk->b_asf )
stream_DemuxSend( p_sys->p_out_asf, p_block );
else
es_out_Send( p_demux->out, tk->p_es, p_block );
}
/* warn that's ok */
p_sys->event_data = 0xff;
/* we have read data */
tk->waiting = 0;
p_demux->p_sys->b_no_data = false;
p_demux->p_sys->i_no_data_ti = 0;
if( i_pts > 0 && !tk->b_muxed )
{
tk->i_pts = i_pts;
}
}
/*****************************************************************************
*
*****************************************************************************/
static void StreamClose( void *p_private )
{
live_track_t *tk = (live_track_t*)p_private;
demux_t *p_demux = tk->p_demux;
demux_sys_t *p_sys = p_demux->p_sys;
tk->b_selected = false;
p_sys->event_rtsp = 0xff;
p_sys->event_data = 0xff;
if( tk->p_es )
es_out_Control( p_demux->out, ES_OUT_SET_ES_STATE, tk->p_es, false );
int nb_tracks = 0;
for( int i = 0; i < p_sys->i_track; i++ )
{
if( p_sys->track[i]->b_selected )
nb_tracks++;
}
msg_Dbg( p_demux, "RTSP track Close, %d track remaining", nb_tracks );
if( !nb_tracks )
p_sys->b_error = true;
}
/*****************************************************************************
*
*****************************************************************************/
static void TaskInterruptRTSP( void *p_private )
{
demux_t *p_demux = (demux_t*)p_private;
/* Avoid lock */
p_demux->p_sys->event_rtsp = 0xff;
}
static void TaskInterruptData( void *p_private )
{
demux_t *p_demux = (demux_t*)p_private;
p_demux->p_sys->i_no_data_ti++;
/* Avoid lock */
p_demux->p_sys->event_data = 0xff;
}
/*****************************************************************************
*
*****************************************************************************/
VLC_NORETURN
static void* TimeoutPrevention( void *p_data )
{
timeout_thread_t *p_timeout = (timeout_thread_t *)p_data;
for( ;; )
{
/* Voodoo (= no) thread safety here! *Ahem* */
if( p_timeout->b_handle_keep_alive )
{
char *psz_bye = NULL;
int canc = vlc_savecancel ();
p_timeout->p_sys->rtsp->sendGetParameterCommand( *p_timeout->p_sys->ms, NULL, psz_bye );
vlc_restorecancel (canc);
}
p_timeout->p_sys->b_timeout_call = !p_timeout->b_handle_keep_alive;
msleep (((int64_t)p_timeout->p_sys->i_timeout - 2) * CLOCK_FREQ);
}
assert(0); /* dead code */
}
/*****************************************************************************
*
*****************************************************************************/
static int ParseASF( demux_t *p_demux )
{
demux_sys_t *p_sys = p_demux->p_sys;
const char *psz_marker = "a=pgmpu:data:application/vnd.ms.wms-hdr.asfv1;base64,";
char *psz_asf = strcasestr( p_sys->p_sdp, psz_marker );
char *psz_end;
block_t *p_header;
/* Parse the asf header */
if( psz_asf == NULL )
return VLC_EGENERIC;
psz_asf += strlen( psz_marker );
psz_asf = strdup( psz_asf ); /* Duplicate it */
psz_end = strchr( psz_asf, '\n' );
while( psz_end > psz_asf && ( *psz_end == '\n' || *psz_end == '\r' ) )
*psz_end-- = '\0';
if( psz_asf >= psz_end )
{
free( psz_asf );
return VLC_EGENERIC;
}
/* Always smaller */
p_header = block_Alloc( psz_end - psz_asf );
p_header->i_buffer = vlc_b64_decode_binary_to_buffer( p_header->p_buffer,
p_header->i_buffer, psz_asf );
//msg_Dbg( p_demux, "Size=%d Hdrb64=%s", p_header->i_buffer, psz_asf );
if( p_header->i_buffer <= 0 )
{
free( psz_asf );
return VLC_EGENERIC;
}
/* Parse it to get packet size */
asf_HeaderParse( &p_sys->asfh, p_header->p_buffer, p_header->i_buffer );
/* Send it to demuxer */
stream_DemuxSend( p_sys->p_out_asf, p_header );
free( psz_asf );
return VLC_SUCCESS;
}
static unsigned char* parseH264ConfigStr( char const* configStr,
unsigned int& configSize )
{
char *dup, *psz;
size_t i_records = 1;
configSize = 0;
if( configStr == NULL || *configStr == '\0' )
return NULL;
psz = dup = strdup( configStr );
/* Count the number of commas */
for( psz = dup; *psz != '\0'; ++psz )
{
if( *psz == ',')
{
++i_records;
*psz = '\0';
}
}
size_t configMax = 5*strlen(dup);
unsigned char *cfg = new unsigned char[configMax];
psz = dup;
for( size_t i = 0; i < i_records; ++i )
{
cfg[configSize++] = 0x00;
cfg[configSize++] = 0x00;
cfg[configSize++] = 0x00;
cfg[configSize++] = 0x01;
configSize += vlc_b64_decode_binary_to_buffer( cfg+configSize,
configMax-configSize, psz );
psz += strlen(psz)+1;
}
free( dup );
return cfg;
}
static uint8_t *parseVorbisConfigStr( char const* configStr,
unsigned int& configSize )
{
configSize = 0;
if( configStr == NULL || *configStr == '\0' )
return NULL;
#if LIVEMEDIA_LIBRARY_VERSION_INT >= 1332115200 // 2012.03.20
unsigned char *p_cfg = base64Decode( configStr, configSize );
#else
char* configStr_dup = strdup( configStr );
unsigned char *p_cfg = base64Decode( configStr_dup, configSize );
free( configStr_dup );
#endif
uint8_t *p_extra = NULL;
/* skip header count, ident number and length (cf. RFC 5215) */
const unsigned int headerSkip = 9;
if( configSize > headerSkip && ((uint8_t*)p_cfg)[3] == 1 )
{
configSize -= headerSkip;
p_extra = (uint8_t*)xmalloc( configSize );
memcpy( p_extra, p_cfg+headerSkip, configSize );
}
delete[] p_cfg;
return p_extra;
}