CVP 7.0 codec change
Has anyone tried to changed the codec from G.711 to G.729 or vise versa via the registry entry:
HKEY_LOCAL_MACHINE\SOFTWARE\Cisco Systems, Inc.\CVP\Customer\Voice Browser\Config\Codec "g711Ulaw64k"
If not, can someone that is running G.729 on CVP send me the g729 key entry?
Best regards and thanks in advance,
Troy
It's not possible to change the codec on CVP VoiceBrowser to G.729
Using VBAdmin on the CVP server, you will see the following:
>>>>SetCodec /?
Type of codec and rate used in the voice browser when communicating with another VOIP endpoint.
Default: g711Ulaw64k
Valid Values: g711Ulaw64k or g711Alaw64k
System shutdown and startup necessary for changes to take effect.
ShowCodec /?
SetCodec g711Ulaw64k
Note: g.729 is not an option. If you try to set the codec to g.729, you will see the following:
>>>>SetCodec g729r8
The new codec value should be either "g711Ulaw64k" or "g711Alaw64k".
Hope this helps.
Hitesh Patel
Similar Messages
-
When exporting to a lossless file, "None" is selected under Codec. Is there actually 'no' codec used for this export setting, or is it some kind of uncompressed lossless codec with no name? The file size is extremely huge.
thared33 wrote:
I was getting a 2GB file for a few seconds of video, and only a few MB for another lossless codec that I'm sure does use compression.
Correct. There's a way to check in AE if a losless codec changed the footage during rendering: put the original footage in a comp, add the rendered footage and apply the Difference blen mode. The differences will appear.
But it really should be okay with a lossless codec. Think of zipping a long work processor document. The file size is smaller, but when you unzip it,you fully expect every letter, number and punctuation mark to be present an in the proper places. That's a lossles codec. -
MPEG Layer 3 audio files have changed into MP3^^ files,What is ^^
MPEG Layer 3 audio files have changed into MP3^^ files,What is ^^ and why wont the music files play anymore??
I reinstalled OSX on my macbook which previously played all avi files with no difficulties. Now I am having difficulties playing Avi files with quicktime PRO that use the Mpeg layer 3 codec. I have done the following with no success: downloaded flip 4 mac, divx, xvid, 3ivx, Xvid codecs, changed MIME settings, and deleted and reinstalled.
Depending on the specific version and sometimes the ordering in which they were installed, you may have created a conflict among the components here. While most current VidX components should be able to play MPEG Layer-3 audio, v5.1.1 is the only free VidX codec that is bi-directional. If that is not a problem, then I would recommend you remove (or "hide") all components in the main "Library > QuickTime" folder. Next I would determine if one (or more) of the AVI associated components can actually play the specific files mentioned on their own. You do this by installing a compenent to be tested, opening the player with one of the problem movies, and checking for correct audio and video playback. Log your results, close the player, remove the tested component, and move on to the next component to be tested. Once you deterime which component(s) may function correctly, I would then move on to determine which specific components are in conflict by repeating the tests but leaving the components installed after each test. In this case start with a component that is known to work and is the one you most desire to use (if more than one works). If the player stops working, then you have located a potential conflict. Set the offending component aside and continue testing until all components have been tested and are either installed or set aside. The last step is to simply evaluate component alternatives and configure you system for those files most often played. In my case the VidX v6 Pro components did not seem to get along with the 3ivX D4 v4.5 and/or Casio AVI and/or XVIDDelegate component combination. -
HOW TO Developing an Authorization plug-in
#if defined (_WIN32)
#pragma warning(disable : 4996)
BOOL WINAPI DllMain(
HINSTANCE hinstDLL, // handle to DLL module
DWORD fdwReason, // reason for calling function
LPVOID lpReserved ) // reserved
return TRUE;
#endif
How to create here
/*----------------------------------------------------------------------------+
| ___ _ _ |
| / | | | | | |
| / /| | __| | ___ | |__ ___ |
| / /_| |/ _ |/ _ \| _ \ / _ \ |
| / ___ | (_| | (_) | |_) | __/ |
| /_/ |_|\__,_|\___/|____/ \___| |
| |
| |
| ADOBE CONFIDENTIAL |
| __________________ |
| |
| Copyright (c) 2003 - 2010, Adobe Systems Incorporated. |
| All rights reserved. |
| |
| NOTICE: All information contained herein is, and remains the property |
| of Adobe Systems Incorporated and its suppliers, if any. The intellectual |
| and technical concepts contained herein are proprietary to Adobe Systems |
| Incorporated and its suppliers and may be covered by U.S. and Foreign |
| Patents, patents in process, and are protected by trade secret or |
| copyright law. Dissemination of this information or reproduction of this |
| material is strictly forbidden unless prior written permission is |
| obtained from Adobe Systems Incorporated. |
| |
| Adobe Systems Incorporated 415.832.2000 |
| 601 Townsend Street 415.832.2020 fax |
| San Francisco, CA 94103 |
| |
+----------------------------------------------------------------------------*/
#include "StdAfx.h"
#include "FmsAuthAdaptor.h"
#include "FmsAuthActions.h"
#include "FmsMedia.h"
#include <stdio.h>
#include <fcntl.h>
#include <string.h>
#include "hash.h"
#include <sstream>
#if defined (_WIN32)
#pragma warning(disable : 4996)
BOOL WINAPI DllMain(
HINSTANCE hinstDLL, // handle to DLL module
DWORD fdwReason, // reason for calling function
LPVOID lpReserved ) // reserved
return TRUE;
#endif
// Flag to process SWF Verification in this auth sample. A real SWF file
// must be targeted in the SWFVerification code below for the example to work.
static const bool kAuthorizeSwfVerification = false;
class FmsAuthAdaptor : public IFmsAuthAdaptor
public:
FmsAuthAdaptor(IFmsAuthServerContext2* pFmsAuthServerContext)
: m_pFmsAuthServerContext(pFmsAuthServerContext) {}
virtual ~FmsAuthAdaptor() {}
void authorize(IFmsAuthEvent* pAev);
void notify(IFmsAuthEvent* pAev);
void getEvents(I32 aevBitAuth[], I32 aevBitNotf[], unsigned int count);
private:
bool getStats(I64 clientStatsHandle, FmsClientStats& baseStats);
void processStats(IFmsAuthEvent* pAev);
IFmsAuthServerContext2* m_pFmsAuthServerContext;
// Utils
// Note: Do not delete the return value. The return value is a buffer
// allocated in FMS memory space, and FMS will manage the memory.
static char* getStringField(const IFmsAuthEvent* pEv, IFmsAuthEvent::Field prop)
FmsVariant field;
if (pEv->getField(prop, field) == IFmsAuthEvent::S_SUCCESS && field.type == field.kString)
return reinterpret_cast<char*>(field.str);
return 0;
// Note: Do not delete the return value. The return value is a buffer
// allocated in FMS memory space, and FMS will manage the memory.
static U8* getBufferField(const IFmsAuthEvent* pEv, IFmsAuthEvent::Field prop)
FmsVariant field;
if (pEv->getField(prop, field) == IFmsAuthEvent::S_SUCCESS && field.type == field.kBuffer)
return field.buf;
return 0;
static bool getI8Field(const IFmsAuthEvent* pEv, IFmsAuthEvent::Field prop, I8& iValue)
FmsVariant field;
if (pEv->getField(prop, field) == IFmsAuthEvent::S_SUCCESS && field.type == field.kI8)
iValue = field.i8;
return true;
return false;
static bool getI32Field(const IFmsAuthEvent* pEv, IFmsAuthEvent::Field prop, I32& iValue)
FmsVariant field;
if (pEv->getField(prop, field) == IFmsAuthEvent::S_SUCCESS && field.type == field.kI32)
iValue = field.i32;
return true;
return false;
static bool getI64Field(const IFmsAuthEvent* pEv, IFmsAuthEvent::Field prop, I64& iValue)
FmsVariant field;
if (pEv->getField(prop, field) == IFmsAuthEvent::S_SUCCESS && field.type == field.kI64)
iValue = field.i64;
return true;
return false;
static bool getFloatField(const IFmsAuthEvent* pEv, IFmsAuthEvent::Field prop, float& fValue)
FmsVariant field;
if (pEv->getField(prop, field) == IFmsAuthEvent::S_SUCCESS && field.type == field.kFloat)
fValue = field.f;
return true;
return false;
static bool getU16Field(const IFmsAuthEvent* pEv, IFmsAuthEvent::Field prop, U16& iValue)
FmsVariant field;
if (pEv->getField(prop, field) == IFmsAuthEvent::S_SUCCESS && field.type == field.kU16)
iValue = field.u16;
return true;
return false;
static bool setStringField(IFmsAuthEvent* pEv, IFmsAuthEvent::Field prop, char* pValue)
FmsVariant field;
field.setString(reinterpret_cast<I8*>(pValue));
return pEv->setField(prop, field) == IFmsAuthEvent::S_SUCCESS;
static bool setI8Field(IFmsAuthEvent* pEv, IFmsAuthEvent::Field prop, I8 iValue)
FmsVariant field;
field.setI8(iValue);
return pEv->setField(prop, field) == IFmsAuthEvent::S_SUCCESS;
static bool setU8Field(IFmsAuthEvent* pEv, IFmsAuthEvent::Field prop, U8 iValue)
FmsVariant field;
field.setU8(iValue);
return pEv->setField(prop, field) == IFmsAuthEvent::S_SUCCESS;
static bool setI32Field(IFmsAuthEvent* pEv, IFmsAuthEvent::Field prop, I32 iValue)
FmsVariant field;
field.setI32(iValue);
return pEv->setField(prop, field) == IFmsAuthEvent::S_SUCCESS;
static bool setI64Field(IFmsAuthEvent* pEv, IFmsAuthEvent::Field prop, I64 iValue)
FmsVariant field;
field.setI64(iValue);
return pEv->setField(prop, field) == IFmsAuthEvent::S_SUCCESS;
static bool setFloatField(IFmsAuthEvent* pEv, IFmsAuthEvent::Field prop, float fValue)
FmsVariant field;
field.setFloat(fValue);
return pEv->setField(prop, field) == IFmsAuthEvent::S_SUCCESS;
static bool isADPCMSupported(int iAudioCodecs)
return (iAudioCodecs & SUPPORT_SND_ADPCM) != 0;
static bool isVP6Supported(int iVideoCodecs)
int iAllVP6 = ( SUPPORT_VID_VP6ALPHA | SUPPORT_VID_VP6 );
return (iVideoCodecs & iAllVP6) != 0;
static bool isService(int iType)
return (iType & TYPE_SERVICE) != 0;
static bool isAMF3(unsigned char uEncod)
return (uEncod == ENCODE_AMF3);
// This class will process all authorization events
class MyFmsAuthorizeEvent
public:
MyFmsAuthorizeEvent(IFmsAuthEvent* pAev, IFmsAuthServerContext2* pFmsAuthServerContext)
: m_pAev(pAev), m_pFmsAuthServerContext(pFmsAuthServerContext) {}
virtual ~MyFmsAuthorizeEvent() {}
void authorize();
private:
IFmsAuthEvent* m_pAev;
IFmsAuthServerContext2* m_pFmsAuthServerContext;
void MyFmsAuthorizeEvent::authorize()
bool bAuthorized = true; // default authorization state
switch(m_pAev->getType())
case IFmsAuthEvent::E_CONNECT:
// only E_CONNECT allows changes to the following fields:
// F_CLIENT_AUDIO_SAMPLE_ACCESS
// F_CLIENT_AUDIO_SAMPLE_ACCESS_LOCK
// F_CLIENT_READ_ACCESS
// F_CLIENT_READ_ACCESS_LOCK
// F_CLIENT_VIDEO_SAMPLE_ACCESS
// F_CLIENT_VIDEO_SAMPLE_ACCESS_LOCK
// F_CLIENT_WRITE_ACCESS_LOCK
// F_CLIENT_WRITE_ACCESS
I8 iValue;
if (getI8Field(m_pAev, IFmsAuthEvent::F_CLIENT_WRITE_ACCESS, iValue))
setI8Field(m_pAev, IFmsAuthEvent::F_CLIENT_WRITE_ACCESS, iValue);
// redirect connection example
char* pUri = getStringField(m_pAev, IFmsAuthEvent::F_CLIENT_URI);
if (pUri && !strcmp(pUri, "rtmp://localhost/streamtest"))
setStringField(m_pAev, IFmsAuthEvent::F_CLIENT_REDIRECT_URI,
"rtmp://localhost:1935/streamtest");
bAuthorized = false;
// set DiffServ fields based on a client IP
// char* pIp = getStringField(m_pAev, IFmsAuthEvent::F_CLIENT_IP);
// if (pIp && !strcmp(pIp, "192.168.1.1"))
// set the DSCP bits and mask
U8 m_diffServBits = 170;
U8 m_diffServMask = 252;
setU8Field(m_pAev, IFmsAuthEvent::F_CLIENT_DIFFSERV_BITS, m_diffServBits);
setU8Field(m_pAev, IFmsAuthEvent::F_CLIENT_DIFFSERV_MASK, m_diffServMask);
bAuthorized = true;
break;
case IFmsAuthEvent::E_PLAY:
char* pStreamName = getStringField(m_pAev, IFmsAuthEvent::F_STREAM_NAME);
if (pStreamName)
setStringField(m_pAev, IFmsAuthEvent::F_STREAM_NAME, pStreamName);
char* pStreamType = getStringField(m_pAev, IFmsAuthEvent::F_STREAM_TYPE);
if (pStreamType)
setStringField(m_pAev, IFmsAuthEvent::F_STREAM_TYPE, pStreamType);
char* pStreamQuery = getStringField(m_pAev, IFmsAuthEvent::F_STREAM_QUERY);
if (pStreamQuery)
setStringField(m_pAev, IFmsAuthEvent::F_STREAM_QUERY, pStreamQuery);
I8 iValue;
if (getI8Field(m_pAev, IFmsAuthEvent::F_STREAM_RESET, iValue))
// If iValue is 1 (true) the playlist will be reset and the
// stream will be the only stream in the playlist; otherwise
// 0 (false) means the stream will be added to the existing
// playlist.
setI8Field(m_pAev, IFmsAuthEvent::F_STREAM_RESET, iValue);
if (getI8Field(m_pAev, IFmsAuthEvent::F_STREAM_IGNORE, iValue))
// If iValue is 1 (true) the stream timestamps will be ignored;
// otherwise 0 (false) means the timestamps will be handled.
setI8Field(m_pAev, IFmsAuthEvent::F_STREAM_IGNORE, iValue);
char* pStreamTransition = getStringField(m_pAev, IFmsAuthEvent::F_STREAM_TRANSITION);
if (pStreamTransition && strlen(pStreamTransition))
// MBR transition example
if (!strcmp(pStreamTransition, "switch") ||
!strcmp(pStreamTransition, "swap"))
// get the old stream's properties
char* pOldStreamName = getStringField(m_pAev, IFmsAuthEvent::F_OLD_STREAM_NAME);
char* pOldStreamType = getStringField(m_pAev, IFmsAuthEvent::F_OLD_STREAM_TYPE);
char* pOldStreamQuery = getStringField(m_pAev, IFmsAuthEvent::F_OLD_STREAM_QUERY);
// if pOldStream is empty (optional for switch) current stream is in play
// do we really want stream transition?
// no we do not allow transition
// bAuthorized = false;
// now transition will be turned off and old stream continue playing
// break;
// doing nothing will execute transition mode as is
// or you could modify transition by changing transition properties
// set it to 1 to indicate they will be hooking up the stream,
// but that it does not currently exist
setI32Field(m_pAev, IFmsAuthEvent::F_STREAM_LIVE_PUBLISH_PENDING, 1);
// get the offset value if transition is set to offset mode for reconnect
if (!strcmp(pStreamTransition, "resume"))
float fValue;
if (getFloatField(m_pAev, IFmsAuthEvent::F_STREAM_OFFSET, fValue))
float offset = fValue; //offset value in seconds
else
// This is a regular play waiting for approval, which may be converted
// into a play2 command by changing transition properties
break;
case IFmsAuthEvent::E_PUBLISH:
char* pStreamName = getStringField(m_pAev, IFmsAuthEvent::F_STREAM_NAME);
if (pStreamName)
setStringField(m_pAev, IFmsAuthEvent::F_STREAM_NAME, pStreamName);
char* pStreamType = getStringField(m_pAev, IFmsAuthEvent::F_STREAM_TYPE);
if (pStreamType)
setStringField(m_pAev, IFmsAuthEvent::F_STREAM_TYPE, pStreamType);
I32 iValue;
if (getI32Field(m_pAev, IFmsAuthEvent::F_STREAM_PUBLISH_TYPE, iValue))
// publish types:
// 0 : record
// 1 : append
// 2 : appendWithGap
// -1 : live
setI32Field(m_pAev, IFmsAuthEvent::F_STREAM_PUBLISH_TYPE, iValue);
break;
case IFmsAuthEvent::E_FILENAME_TRANSFORM:
I64 iValue;
if (getI64Field(m_pAev, IFmsAuthEvent::F_CLIENT_ID, iValue))
// some fields are not eligible to be modified. The return
// value will be false when trying to modify the F_CLIENT_ID.
bool bSet = setI64Field(m_pAev, IFmsAuthEvent::F_CLIENT_ID, iValue);
char* pStreamName = getStringField(m_pAev, IFmsAuthEvent::F_STREAM_NAME);
if (pStreamName)
// some fields are not eligible to be modified. The return
// value will be false when trying to modify the F_STREAM_NAME.
bool bSet = setStringField(m_pAev, IFmsAuthEvent::F_STREAM_NAME, pStreamName);
char* pStreamPath = getStringField(m_pAev, IFmsAuthEvent::F_STREAM_PATH);
if (pStreamPath)
setStringField(m_pAev, IFmsAuthEvent::F_STREAM_PATH, pStreamPath);
char* pStreamType = getStringField(m_pAev, IFmsAuthEvent::F_STREAM_TYPE);
if (pStreamType)
setStringField(m_pAev, IFmsAuthEvent::F_STREAM_TYPE, pStreamType);
break;
case IFmsAuthEvent::E_PAUSE:
bAuthorized = false; // block all E_PAUSE events.
float fValue;
if (getFloatField(m_pAev, IFmsAuthEvent::F_STREAM_PAUSE_TIME, fValue))
float fPauseTime = fValue; // in seconds
I8 iValue;
if (getI8Field(m_pAev, IFmsAuthEvent::F_STREAM_PAUSE, iValue))
// 1 (true) means PAUSE
// 0 (false) means UNPAUSE
bool boolPause = iValue != 0;
if (getI8Field(m_pAev, IFmsAuthEvent::F_STREAM_PAUSE_TOGGLE, iValue))
// 1 (true) means PAUSE_TOGGLE
// 0 (false) means no PAUSE_TOGGLE was set
bool boolPauseToggle = iValue != 0;
FmsVariant field;
// Notify Action example: An IFmsNofifyAction is created to notify
// server side action script (SSAS) of the E_PAUSE event by calling
// the function name "method" in the script. In this example two
// variables will be passed to "method" by calling addParam(field)
// on the action.
if (m_pAev->getField(IFmsAuthEvent::F_CLIENT_ID, field) == IFmsAuthEvent::S_SUCCESS)
I64 clientId = field.i64;
IFmsNotifyAction* pAction = m_pAev->addNotifyAction("Notified by adaptor");
pAction->setClientId(field);
const char mtd[] = "method";
field.setString(reinterpret_cast<I8*>(const_cast<char*>(mtd)));
pAction->setMethodName(field);
// create and insert a U16 "12345" as the first parameter
field.setU16(12345);
pAction->addParam(field);
// create and insert clientId as a double as the second parameter
field.setDouble((double)clientId);
pAction->addParam(field);
// Note: SSAS does not work with I64 or Buffer variants
// field.setI64(clientId);
// pAction->addParam(field); // incorrect
break;
case IFmsAuthEvent::E_SEEK:
bAuthorized = false; // block all E_SEEK events
float fValue;
if (getFloatField(m_pAev, IFmsAuthEvent::F_STREAM_SEEK_POSITION, fValue))
// Modification of the seek position example:
// fValue + 3; will add 3 seconds to the initial seek posistion
float fSeekTime = fValue; // value in seconds
setFloatField(m_pAev, IFmsAuthEvent::F_STREAM_SEEK_POSITION, fSeekTime);
break;
case IFmsAuthEvent::E_LOADSEGMENT:
// bAuthorized = false; // block all E_LOADSEGMENT events
// E_LOADSEGMENT is a read only event that substitutes E_PLAY on
// FMS Origin servers for recorded streams.
I64 iValue;
if (getI64Field(m_pAev, IFmsAuthEvent::F_SEGMENT_START, iValue))
I64 iStart = iValue; // in bytes
if (getI64Field(m_pAev, IFmsAuthEvent::F_SEGMENT_END, iValue))
I64 iEnd = iValue; // in bytes
break;
case IFmsAuthEvent::E_RECORD:
// bAuthorized = false; // block all E_RECORD events
float fValue;
if (getFloatField(m_pAev, IFmsAuthEvent::F_STREAM_RECORD_MAXSIZE, fValue))
float recMaxSize = fValue; // in kilobytes
setFloatField(m_pAev, IFmsAuthEvent::F_STREAM_RECORD_MAXSIZE, recMaxSize);
if (getFloatField(m_pAev, IFmsAuthEvent::F_STREAM_RECORD_MAXDURATION, fValue))
float recMaxDuration = fValue; // in seconds
setFloatField(m_pAev, IFmsAuthEvent::F_STREAM_RECORD_MAXDURATION, recMaxDuration);
break;
case IFmsAuthEvent::E_SWF_VERIFY:
// SWF Verification example:
// kAuthorizeSwfVerification is assigned false by default. The
// target SWF file must be updated for this to work.
if(kAuthorizeSwfVerification)
I8 swfvVersion = 0;
if(getI8Field(m_pAev, IFmsAuthEvent::F_CLIENT_SWFV_VERSION, swfvVersion))
std::stringstream stream;
stream << "Swf verification version is " << static_cast<int>(swfvVersion);
m_pFmsAuthServerContext->log(stream.str().c_str(), IFmsServerContext::kInformation, false);
I64 swfvDepth;
if(getI64Field(m_pAev, IFmsAuthEvent::F_CLIENT_SWFV_DEPTH, swfvDepth))
I32 swfvTTL;
if(getI32Field(m_pAev, IFmsAuthEvent::F_CLIENT_SWFV_TTL, swfvTTL))
swfvTTL /= 2;
setI32Field(m_pAev, IFmsAuthEvent::F_CLIENT_SWFV_TTL, swfvTTL);
U8 digest[kSHA256DigestLen];
// Target a real SWF file instead of sample.swf
hashSwfFileAtDepth("C:\\sample.swf", swfvDepth, digest);
FmsVariant field;
field.setBuffer(digest, kSHA256DigestLen);
m_pAev->setField(IFmsAuthEvent::F_CLIENT_SWFV_DIGEST, field);
break;
case IFmsAuthEvent::E_APPSTART:
case IFmsAuthEvent::E_APPSTOP:
case IFmsAuthEvent::E_DISCONNECT:
case IFmsAuthEvent::E_STOP:
case IFmsAuthEvent::E_UNPUBLISH:
case IFmsAuthEvent::E_ACTION:
case IFmsAuthEvent::E_CODEC_CHANGE:
case IFmsAuthEvent::E_RECORD_STOP:
case IFmsAuthEvent::E_CLIENT_PAUSE:
case IFmsAuthEvent::E_SWF_VERIFY_COMPLETE:
case IFmsAuthEvent::E_CLIENT_SEEK:
case IFmsAuthEvent::E_START_TRANSMIT:
case IFmsAuthEvent::E_STOP_TRANSMIT:
case IFmsAuthEvent::E_MAXEVENT:
break;
IFmsAuthServerContext2::AuthFailureDesc* desc = NULL;
if(!bAuthorized)
desc = new IFmsAuthServerContext2::AuthFailureDesc("Blocked by auth adaptor",
IFmsAuthServerContext2::kDefaultStatus, -1);
char buf[1024];
const char* const action = bAuthorized ? "approved" : "rejected";
sprintf(buf, "Received authorization type=%d id=%p %s\n", m_pAev->getType(),
m_pAev, action);
// log to the configured FMS log directory. If the third parameter is true,
// also send the log to the system event log.
m_pFmsAuthServerContext->log(buf, IFmsServerContext::kInformation, false);
m_pFmsAuthServerContext->onAuthorize(m_pAev, bAuthorized, desc);
delete desc;
class MyFmsNotifyEvent
public:
MyFmsNotifyEvent(IFmsAuthEvent* pAev, IFmsAuthServerContext2* pFmsAuthServerContext)
: m_pAev(pAev), m_pFmsAuthServerContext(pFmsAuthServerContext) {}
virtual ~MyFmsNotifyEvent() {}
void notify() const;
private:
IFmsAuthEvent* m_pAev;
IFmsAuthServerContext2* m_pFmsAuthServerContext;
void MyFmsNotifyEvent::notify() const
switch(m_pAev->getType())
case IFmsAuthEvent::E_PLAY:
char* pAppName = getStringField(m_pAev, IFmsAuthEvent::F_APP_NAME);
char* pAppInst = getStringField(m_pAev, IFmsAuthEvent::F_APP_INST);
char* pAppUri = getStringField(m_pAev, IFmsAuthEvent::F_APP_URI);
char* pClIp = getStringField(m_pAev, IFmsAuthEvent::F_CLIENT_IP);
char* pClUri = getStringField(m_pAev, IFmsAuthEvent::F_CLIENT_URI);
char* pClNewUri = getStringField(m_pAev, IFmsAuthEvent::F_CLIENT_REDIRECT_URI);
char* pClVhost = getStringField(m_pAev, IFmsAuthEvent::F_CLIENT_VHOST);
char* pClRef = getStringField(m_pAev, IFmsAuthEvent::F_CLIENT_REFERRER);
char* pClPurl = getStringField(m_pAev, IFmsAuthEvent::F_CLIENT_PAGE_URL);
char* pClAgent = getStringField(m_pAev, IFmsAuthEvent::F_CLIENT_USER_AGENT);
char* pClRAccess = getStringField(m_pAev, IFmsAuthEvent::F_CLIENT_READ_ACCESS);
char* pClWAccess = getStringField(m_pAev, IFmsAuthEvent::F_CLIENT_WRITE_ACCESS);
char* pClAudioAccess = getStringField(m_pAev, IFmsAuthEvent::F_CLIENT_AUDIO_SAMPLE_ACCESS);
char* pClVideoAccess = getStringField(m_pAev, IFmsAuthEvent::F_CLIENT_VIDEO_SAMPLE_ACCESS);
char* pClProto = getStringField(m_pAev, IFmsAuthEvent::F_CLIENT_PROTO);
char* pClUstem = getStringField(m_pAev, IFmsAuthEvent::F_CLIENT_URI_STEM);
char* pStreamName = getStringField(m_pAev, IFmsAuthEvent::F_STREAM_NAME);
char* pStreamType = getStringField(m_pAev, IFmsAuthEvent::F_STREAM_TYPE);
char* pStreamQuery = getStringField(m_pAev, IFmsAuthEvent::F_STREAM_QUERY);
char* pStreamPath = getStringField(m_pAev, IFmsAuthEvent::F_STREAM_PATH);
I32 iValue;
if (getI32Field(m_pAev, IFmsAuthEvent::F_CLIENT_AUDIO_CODECS, iValue))
bool bADPCM = isADPCMSupported(iValue);
if (getI32Field(m_pAev, IFmsAuthEvent::F_CLIENT_VIDEO_CODECS, iValue))
bool bVP6 = isVP6Supported(iValue);
if (getI32Field(m_pAev, IFmsAuthEvent::F_CLIENT_TYPE, iValue))
bool bService = isService(iValue);
if (getI32Field(m_pAev, IFmsAuthEvent::F_STREAM_ID, iValue))
I32 iStreamId = iValue;
float fValue;
if (getFloatField(m_pAev, IFmsAuthEvent::F_STREAM_LENGTH, fValue))
float fLength = fValue; // in seconds
if (getFloatField(m_pAev, IFmsAuthEvent::F_STREAM_POSITION, fValue))
float iPosition = fValue; // in seconds
I64 lValue;
if (getI64Field(m_pAev, IFmsAuthEvent::F_CLIENT_ID, lValue))
I64 iClientId = lValue;
I8 sValue;
if (getI8Field(m_pAev, IFmsAuthEvent::F_CLIENT_SECURE, sValue))
bool bSecure = sValue != 0;
if (getI8Field(m_pAev, IFmsAuthEvent::F_CLIENT_AMF_ENCODING, sValue))
bool bAMF3 = isAMF3(sValue);
if (getI8Field(m_pAev, IFmsAuthEvent::F_CLIENT_READ_ACCESS_LOCK, sValue))
bool bRead = sValue != 0;
if (getI8Field(m_pAev, IFmsAuthEvent::F_CLIENT_WRITE_ACCESS_LOCK, sValue))
bool bWrite = sValue != 0;
if (getI8Field(m_pAev, IFmsAuthEvent::F_CLIENT_AUDIO_SAMPLE_ACCESS_LOCK, sValue))
bool bAudioRead = sValue != 0;
if (getI8Field(m_pAev, IFmsAuthEvent::F_CLIENT_VIDEO_SAMPLE_ACCESS_LOCK, sValue))
bool bVideoRead = sValue != 0;
if (getI8Field(m_pAev, IFmsAuthEvent::F_STREAM_RESET, sValue))
bool bReset = sValue != 0;
if (getI8Field(m_pAev, IFmsAuthEvent::F_STREAM_IGNORE, sValue))
bool bIgnore = sValue != 0;
break;
case IFmsAuthEvent::E_SEEK:
float fValue;
if (getFloatField(m_pAev, IFmsAuthEvent::F_STREAM_SEEK_POSITION, fValue))
float fSeekTime = fValue;
// Disconnect Action example: disconnect the client that was
// specified by the E_SEEK notify event
FmsVariant field;
if (m_pAev->getField(IFmsAuthEvent::F_CLIENT_ID, field) == IFmsAuthEvent::S_SUCCESS)
IFmsDisconnectAction* pAction =
const_cast<IFmsAuthEvent*>(m_pAev)->
addDisconnectAction("Seek is not allowed. Blocked by adaptor");
pAction->setClientId(field);
break;
case IFmsAuthEvent::E_CODEC_CHANGE:
char* pAppName = getStringField(m_pAev, IFmsAuthEvent::F_APP_NAME);
char* pAppInst = getStringField(m_pAev, IFmsAuthEvent::F_APP_INST);
char* pAppUri = getStringField(m_pAev, IFmsAuthEvent::F_APP_URI);
char* pClIp = getStringField(m_pAev, IFmsAuthEvent::F_CLIENT_IP);
char* pClUri = getStringField(m_pAev, IFmsAuthEvent::F_CLIENT_URI);
char* pClNewUri = getStringField(m_pAev, IFmsAuthEvent::F_CLIENT_REDIRECT_URI);
char* pClVhost = getStringField(m_pAev, IFmsAuthEvent::F_CLIENT_VHOST);
char* pClRef = getStringField(m_pAev, IFmsAuthEvent::F_CLIENT_REFERRER);
char* pClPurl = getStringField(m_pAev, IFmsAuthEvent::F_CLIENT_PAGE_URL);
char* pClAgent = getStringField(m_pAev, IFmsAuthEvent::F_CLIENT_USER_AGENT);
char* pClRAccess = getStringField(m_pAev, IFmsAuthEvent::F_CLIENT_READ_ACCESS);
char* pClWAccess = getStringField(m_pAev, IFmsAuthEvent::F_CLIENT_WRITE_ACCESS);
char* pClAudioAccess = getStringField(m_pAev, IFmsAuthEvent::F_CLIENT_AUDIO_SAMPLE_ACCESS);
char* pClVideoAccess = getStringField(m_pAev, IFmsAuthEvent::F_CLIENT_VIDEO_SAMPLE_ACCESS);
char* pClProto = getStringField(m_pAev, IFmsAuthEvent::F_CLIENT_PROTO);
char* pClUstem = getStringField(m_pAev, IFmsAuthEvent::F_CLIENT_URI_STEM);
char* pStreamName = getStringField(m_pAev, IFmsAuthEvent::F_STREAM_NAME);
char* pStreamType = getStringField(m_pAev, IFmsAuthEvent::F_STREAM_TYPE);
char* pStreamQuery = getStringField(m_pAev, IFmsAuthEvent::F_STREAM_QUERY);
char* pStreamPath = getStringField(m_pAev, IFmsAuthEvent::F_STREAM_PATH);
U16 fType;
if (getU16Field(m_pAev, IFmsAuthEvent::F_STREAM_CODEC_TYPE, fType))
U16 streamCodecType = fType;
if (streamCodecType == kVIDEO_CODEC)
U16 fValue;
if (getU16Field(m_pAev, IFmsAuthEvent::F_STREAM_CODEC, fValue))
U16 streamCodecValue = fValue;
if (streamCodecValue == VIDEO_CODEC_SORENSON)
// Disconnect Action example: Disallow clients trying
// to publish content with the sorenson video codec.
FmsVariant field;
if (m_pAev->getField(IFmsAuthEvent::F_CLIENT_ID, field) == IFmsAuthEvent::S_SUCCESS)
IFmsDisconnectAction* pAction =
const_cast<IFmsAuthEvent*>(m_pAev)->
addDisconnectAction("Sorenson is not allowed. Blocked by adaptor");
pAction->setClientId(field);
break;
case IFmsAuthEvent::E_RECORD_STOP:
char* pAppName = getStringField(m_pAev, IFmsAuthEvent::F_APP_NAME);
char* pAppInst = getStringField(m_pAev, IFmsAuthEvent::F_APP_INST);
char* pAppUri = getStringField(m_pAev, IFmsAuthEvent::F_APP_URI);
char* pClIp = getStringField(m_pAev, IFmsAuthEvent::F_CLIENT_IP);
char* pClUri = getStringField(m_pAev, IFmsAuthEvent::F_CLIENT_URI);
char* pClNewUri = getStringField(m_pAev, IFmsAuthEvent::F_CLIENT_REDIRECT_URI);
char* pClVhost = getStringField(m_pAev, IFmsAuthEvent::F_CLIENT_VHOST);
char* pClRef = getStringField(m_pAev, IFmsAuthEvent::F_CLIENT_REFERRER);
char* pClPurl = getStringField(m_pAev, IFmsAuthEvent::F_CLIENT_PAGE_URL);
char* pClAgent = getStringField(m_pAev, IFmsAuthEvent::F_CLIENT_USER_AGENT);
char* pClRAccess = getStringField(m_pAev, IFmsAuthEvent::F_CLIENT_READ_ACCESS);
char* pClWAccess = getStringField(m_pAev, IFmsAuthEvent::F_CLIENT_WRITE_ACCESS);
char* pClAudioAccess = getStringField(m_pAev, IFmsAuthEvent::F_CLIENT_AUDIO_SAMPLE_ACCESS);
char* pClVideoAccess = getStringField(m_pAev, IFmsAuthEvent::F_CLIENT_VIDEO_SAMPLE_ACCESS);
char* pClProto = getStringField(m_pAev, IFmsAuthEvent::F_CLIENT_PROTO);
char* pClUstem = getStringField(m_pAev, IFmsAuthEvent::F_CLIENT_URI_STEM);
char* pStreamName = getStringField(m_pAev, IFmsAuthEvent::F_STREAM_NAME);
char* pStreamType = getStringField(m_pAev, IFmsAuthEvent::F_STREAM_TYPE);
char* pStreamQuery = getStringField(m_pAev, IFmsAuthEvent::F_STREAM_QUERY);
char* pStreamPath = getStringField(m_pAev, IFmsAuthEvent::F_STREAM_PATH);
float fValue;
if (getFloatField(m_pAev, IFmsAuthEvent::F_STREAM_RECORD_MAXSIZE, fValue))
float recMaxSize = fValue; // in kilobytes
if (getFloatField(m_pAev, IFmsAuthEvent::F_STREAM_RECORD_MAXDURATION, fValue))
float recMaxDuration = fValue; // in seconds
break;
case IFmsAuthEvent::E_SWF_VERIFY_COMPLETE:
char* pClIp = getStringField(m_pAev, IFmsAuthEvent::F_CLIENT_IP);
I8 version; // version of SWF verification
getI8Field(m_pAev, IFmsAuthEvent::F_CLIENT_SWFV_VERSION, version);
I64 depth; // depth in the SWF file hashed
getI64Field(m_pAev, IFmsAuthEvent::F_CLIENT_SWFV_DEPTH, depth);
I32 ttl; // time to live of the SWF hash provided
getI32Field(m_pAev, IFmsAuthEvent::F_CLIENT_SWFV_TTL, ttl);
// digest provided to match against
U8* buffer = getBufferField(m_pAev, IFmsAuthEvent::F_CLIENT_SWFV_DIGEST);
// result of the attempted match-- see FmsAuthEvents.h enum
// eSWFMatch for the meaning of this field
I32 match;
getI32Field(m_pAev, IFmsAuthEvent::F_CLIENT_SWFV_RESULT, match);
std::stringstream stream;
stream << "swf verification for client: "
<< std::string(pClIp)
<< " is complete, the result is: " << match;
m_pFmsAuthServerContext->log(stream.str().c_str(), IFmsServerContext::kInformation, false);
break;
case IFmsAuthEvent::E_APPSTART:
case IFmsAuthEvent::E_APPSTOP:
case IFmsAuthEvent::E_CONNECT:
case IFmsAuthEvent::E_DISCONNECT:
case IFmsAuthEvent::E_FILENAME_TRANSFORM:
case IFmsAuthEvent::E_STOP:
case IFmsAuthEvent::E_PAUSE:
case IFmsAuthEvent::E_PUBLISH:
case IFmsAuthEvent::E_UNPUBLISH:
case IFmsAuthEvent::E_LOADSEGMENT:
case IFmsAuthEvent::E_ACTION:
case IFmsAuthEvent::E_RECORD:
case IFmsAuthEvent::E_CLIENT_PAUSE:
case IFmsAuthEvent::E_SWF_VERIFY:
case IFmsAuthEvent::E_CLIENT_SEEK:
case IFmsAuthEvent::E_START_TRANSMIT:
case IFmsAuthEvent::E_STOP_TRANSMIT:
case IFmsAuthEvent::E_MAXEVENT:
break;
char buf[1024];
sprintf(buf, "Received notification type=%d id=%p\n", m_pAev->getType(), m_pAev);
// log to the configured FMS log directory. If the third parameter is true,
// also send the log to the system event log.
m_pFmsAuthServerContext->log(buf, IFmsServerContext::kInformation, false);
m_pFmsAuthServerContext->onNotify(m_pAev);
/* All authorization events flow through this wrapper function.
* Note: This sample auth adaptor has MyFmsAppAuthEvent allocated on the
* stack, but time intensive implementations may warrant authorization to
* be allocated on the heap so work may be passed to a thread pool. This
* prevents starvation of the calling FMS threads in custom code that may
* have processing delays (ie database calls, network filesystem access, etc..).
void FmsAuthAdaptor::authorize(IFmsAuthEvent* pAev)
MyFmsAuthorizeEvent(pAev, m_pFmsAuthServerContext).authorize();
/* All notification events flow through this wrapper function.
* Note: This sample auth adaptor has MyFmsNotifyEvent allocated on the
* stack, but time intensive implementations may warrant notifications to
* be allocated on the heap so work may be passed to a thread pool. This
* prevents starvation of the calling FMS threads in custom code that may
* have processing delays (ie database calls, network filesystem access, etc..).
void FmsAuthAdaptor::notify(IFmsAuthEvent* pAev)
processStats(pAev);
MyFmsNotifyEvent(pAev, m_pFmsAuthServerContext).notify();
* Get client statistics.
bool FmsAuthAdaptor::getStats(I64 clientStatsHandle, FmsClientStats& baseStats)
bool bValue= m_pFmsAuthServerContext->getClientStats(clientStatsHandle, baseStats);
return bValue;
* Example obtainting client stats from an E_CONNECT or E_STOP event
void FmsAuthAdaptor::processStats(IFmsAuthEvent* pAev)
I64 statsHandle;
FmsClientStats baseStats;
if (!getI64Field(pAev, IFmsAuthEvent::F_CLIENT_STATS_HANDLE, statsHandle))
return;
char* pAppName = getStringField(pAev, IFmsAuthEvent::F_APP_NAME);
if (pAev->getType() == IFmsAuthEvent::E_CONNECT)
getStats(statsHandle, baseStats);
// log data
char buf[1024];
char hashKey[9];
memset(hashKey, 0, 9);
memcpy(hashKey, &statsHandle, sizeof(statsHandle));
sprintf(buf, "client Stats Handle= %s, bytes_in= %f, bytes_out= %f\n", hashKey,
static_cast<double>(baseStats.bytes_in), static_cast<double>(baseStats.bytes_out));
m_pFmsAuthServerContext->log(buf, IFmsServerContext::kInformation, false);
else if (pAev->getType() == IFmsAuthEvent::E_STOP)
getStats(statsHandle, baseStats);
/* By default, all authorization and notifications events will be sent.
* Call excludeEvents with the bit set to 1, to stop recieving events.
* Note: The events:
* E_APPSTART, E_APPSTOP, E_DISCONNECT, E_STOP, E_UNPUBLISH, E_CODEC_CHANGE
* are excluded by default and are not authorizable.
void FmsAuthAdaptor::getEvents(I32 aevBitAuth[], I32 aevBitNotf[], unsigned int count)
// exclude certain auth events
IFmsAuthEvent::EventType authExcludeEvent[] = { IFmsAuthEvent::E_SEEK };
// set E_SEEK to a non authorizable event
m_pFmsAuthServerContext->excludeEvents(aevBitAuth, count, authExcludeEvent, 1);
// Warning: if E_CODEC_CHANGE event is not excluded, all messages will be
// scanned to detect codec change. Subscribe to this event only as needed.
// Example that excludes certain notify events. (E_PAUSE, E_CODEC_CHANGE)
IFmsAuthEvent::EventType notifyExcludeEvent[] =
{ IFmsAuthEvent::E_PAUSE, IFmsAuthEvent::E_CODEC_CHANGE };
m_pFmsAuthServerContext->excludeEvents(aevBitNotf, count, notifyExcludeEvent, 2);
extern "C" void FCExport FmsCreateAuthAdaptor3(IFmsAuthServerContext2* pAuthServerCtx,
IFmsAuthAdaptor*& pFmsAuthAdaptor, U32& iVersion)
pFmsAuthAdaptor = new FmsAuthAdaptor(pAuthServerCtx);
U32 version = pAuthServerCtx->getVersion();
U16 w2 = LWORD(version);
U16 w1 = HWORD(version);
iVersion = MKLONG(INTERFACE_MINOR_VERSION, INTERFACE_MAJOR_VERSION);
char buf[1024];
char *ptr = buf;
int valueLen = pAuthServerCtx->getConfig("UserKey1", &ptr, sizeof(buf));
if (!valueLen)
valueLen = pAuthServerCtx->getConfig("UserKey2", &ptr, sizeof(buf));
if (!valueLen)
return;
if (valueLen < 0)
// failed to find this key
return;
if (valueLen < 0)
// failed to find this key
return;
// value length is bigger then the buffer size, and a real adaptor should
// allocate valueLen + 1 bytes and call again
extern "C" void FCExport FmsDestroyAuthAdaptor3(IFmsAuthAdaptor* pAuthAdaptor )
delete pAuthAdaptor;There is no API to Acrobat's document compare feature.
It is certainly possible for an experienced plug-in programmer to
create a new compare plug-in. For example, extract text from two PDFs
and compare it. Comparison algorithms have been much studied so should
be findable in academic literature.
Going beyond text comparison would be a major exercise.
Aandi Inston -
[Not really solved] Video with Pidgin is not working
Hi,
I would like to use video chat with Pidgin, but it doesn't work on my computer.
I have a webcam (:D), and it seems that it is correctly recognized because ` vlc v4l2:// ' shows me my face. I'm using XMPP (with a GMail account, but this shouldn't matter) and want to chat with someone using Ubuntu and Pidgin 2.6.6 (she has a webcam which seems to work correctly)
In the "Voice/Video settings" plugin, there are three options in the field Video → Input → Plugins : "default", "Test Input" and "Video4Linux" (and only "default" in the field Device, for the three plugins)
"default" and "Video4Linux" give
(21:03:34) Error with your webcam
(21:03:34) Conference error
(21:03:34) The call has been terminated.
as soon as I try to start a video call, and "Test Input" only shows me a test pattern (but I guess this is the meaning of "Test Input" )
BTW, I wonder why there is only "Video4Linux" and not "Video4Linux2"
The Pidgin's FAQ says this
How do I configure my microphone/webcam?
Currently, the command-line gstreamer-properties program is used. This is only available on GNOME-based systems. On other systems, Pidgin makes its best guess as to which device to use. A plugin is planned to support this functionality on other systems.
But I don't use GNOME, my desktop environment is LXDE.
What could I do to make my webcam work with Pidgin?
Thank you!
Edit:
- my system is up to date
- I have farsight2 and gstreamer0.10-{good,bad,ugly} installed
Fractal
Last edited by Fractal (2010-06-13 15:50:01)Well, the problem is not really solved actually.
Pidgin recognizes my webcam but I still cannot open a video chat.
When I try to start a video chat, the following errors are displayed in the debug window :
(17:38:41) mediamanager: gst pipeline error: Could not encode stream.
(17:38:41) mediamanager: Debug details: gstx264enc.c(737): gst_x264_enc_header_buf (): /GstPipeline:pipeline0/GstBin:conf_0x306c000/FsRtpConference:fsrtpconference1/GstBin:discover_1_97/GstX264Enc:x264enc0:
Unexpected x264 header.
When the other person tries to call me, I have something like that :
(17:43:23) jingle-rtp: received codec: 32: video MPV clock:90000 channels:0
(17:43:23) jingle-rtp: received codec: 26: video JPEG clock:90000 channels:0
(17:43:23) jingle-rtp: received codec: 97: video H263-1998 clock:90000 channels:0
(17:43:23) jingle-rtp: received codec: 34: video H263 clock:90000 channels:0
(17:43:23) jingle-rtp: received codec: 98: video DV clock:90000 channels:0
(17:43:23) jabber: setting param stun-ip for stream using auto-discovered IP: 209.85.229.126
(17:43:23) jabber: setting param stun-port for stream using auto-discovered port: 19302
(17:43:23) mediamanager: Couldn't read fs-element.conf: No such file or directory
(17:43:23) backend-fs2: Couldn't read fs-codec.conf: No such file or directory
(17:43:23) backend-fs2: Creating hash table for sessions
(17:43:23) backend-fs2: connecting pad: success
(17:43:23) backend-fs2: Creating hash table for participants
(17:43:23) media: Creating hash table for sessions
(17:43:23) gtkmedia: state: 0 sid: video-session name: (null)
(17:43:23) jingle-rtp: state-changed: state 0 id: video-session name: (null)
(17:43:23) backend-fs2: got new local candidate: 1
(17:43:23) backend-fs2: got new local candidate: 1
(17:43:23) backend-fs2: farsight-component-state-changed: component: 1 state: GATHERING
(17:43:23) backend-fs2: farsight-component-state-changed: component: 2 state: GATHERING
(17:43:23) jingle-rtp: jingle_rtp_codecs_changed_cb: session_id: video-session jingle_session: 0x212b6a0
(17:43:23) jingle-rtp: jingle_rtp_candidates_prepared_cb
(17:43:23) jingle: jingle_iceudp_finalize
(17:43:23) jingle: jingle_transport_finalize
(17:43:26) backend-fs2: farsight-send-codec-changed: codec: 96: video THEORA clock:90000 channels:0 delivery-method=inline
(17:43:27) jingle-rtp: stream-info: type 1 id: video-session name: (null)
(17:43:27) jingle-rtp: stream-info: type 1 id: (null) name: (null)
(17:43:27) backend-fs2: farsight-component-state-changed: component: 1 state: CONNECTING
(17:43:27) backend-fs2: farsight-component-state-changed: component: 2 state: CONNECTING
(17:43:36) backend-fs2: farsight-component-state-changed: component: 1 state: FAILED
(17:43:36) backend-fs2: farsight-error: 108: Could not establish connection
(17:43:36) backend-fs2: farsight-component-state-changed: component: 2 state: FAILED
It seems that my Pidgin want to encode the flux in H264 and the other wants Theora
Fractal -
I'm upgrading to a new macbook pro mid 2012 with a Solid state drive from a macbook due 2 white that I have a 750 gb HD. My itunes is 95% of that Hard drive I just want to make sure I do not loss any of my itunes info in the move to the hard drive.
So, everything worked out just fine!
I have all my material in the ProRes Codec on a seperate drive now. The project itself was kept or rather duplicated, the bin structure etc stayed the same but is linking to the newly created files. The playback and workflow is fluent. I am happy!
Time info: For about 120 GB of material the codec change process took 25 hours.
Thanks for your help!
Best
*a -
Beginners Guide sound installation -- OSS and ALSA objective features
Beginners Guide sound installation -- OSS and ALSA objective features needed
Hi,
In most wiki pages i just edit the things that i think need editing. Just look at the oss history
In all those cases i didn't really saw a need to ask for permission.. that would kinda destroy the wiki idea.
However i want to change the sound instructions in the beginners guide so i made a copy of the entire guide and the part that i changed there is: http://wiki.archlinux.org/index.php/Use … ling_Sound now i have a slight issue there. As you can see i'm in favor of OSS and because you can spot that it's not objective. Now i would ask 2 things.
1. Could you all post your features of OSS and ALSA
2. Once that's done can i have permission to place that section in the beginners guide? that will also include removing the sound installation from the beginners guide which i already did in my version
My personal reason to do this. i've read this and am since then pro OSS and against the ALSA/PulseAudio combo. i think that combo needs to get out of linux (most notably Fedora and Ubuntu) ASAP. and oss needs to go back into the linux kernel
Thank you for your time,
Markngoonee wrote:
I'm an alsa/pulse user, so I'll give a bit of the 'other side'.
Alsa/pulse
Pros:-
network sound
advanced connection of sink/source (including merging sinks)
bluetooth support!
highly supported by existing apps, either through directly supporting Pulse or through its alsa plugin (you should not need to recompile properly-written apps which do not assume they should write audio data directly to hardware, I think I only needed to recompile mpd on my system)
Cons:-
Setup isn't the easiest. Can't comment vs OSS4 because I haven't tried it
OSS4
Pros:-
Everyone seems to say sound quality is better. I guess that's because they're comparing it with dmix alsa. Use pulse with alsa and you should not notice any difference in sound quality though.
Cons:-
Most apps nowadays default output to alsa. Meaning OSS plays them using an alsa plugin.
USB support is admittedly skimpy.
EDIT: Having read the sound article you referred to, my only comment is that the writer really has it in for Pulse... 3 seconds latency, where'd he get that from? I use pulse for audio recording (when I'm lazy to fire up JACK) and while there IS latency, its definitely in the ms range.
Thanx for the input
Gen2ly wrote:
Gen2ly wrote:...As a side note, do you need libflashsupport here???...
markg85, libflashsupport isn't needed. [1]
pacman -Ql oss | grep flash
If you don't know, please don't put in wiki, this could cause unnecessary problems. As for the mms section:
If your stream sounds ugly in totem like it did with me then you could try to play it with another codec like ffmpeg (mplayer). That "fixed" the issue for me. This will not fix the issue that somehow pops up in gstreamer when playing MMS streams but it will give you the option to play it with good sound quality. Playing it in mplayer is simple:
# mplayer mmsh://yourstreamurl
Could you fix this? ffmpeg is not a codec . Also define ugly, and what is somehow?
markg85 wrote:Thanx a lot for your feedback. i will certainly use it when i make more edits.
As for the things you didn't know. As soon as i fully understand how i can get a microphone working in OSS i will add that to the wiki as well. Unless you already know it.. in that case, feel free to add it.
For the mic, I did get mine going. Can't remember just how I did mine (sorry, think I had to disable one of the inputs), but do remember to prevent it from passing through the speakers had to disable "Misc Microphone".
# ossmix
Selected mixer 0/High Definition Audio ALC888
Known controls are:
jack.green.mode <front|rear|center/LFE|side|pcm4|input> (currently front)
jack.green [<leftvol>:<rightvol>] (currently 29.9:29.9 dB)
jack.green.mute ON|OFF (currently OFF)
jack.black.mode <front|rear|center/LFE|side|pcm4|input> (currently center/LFE)
jack.black [<leftvol>:<rightvol>] (currently 29.9:29.9 dB)
jack.black.mute ON|OFF (currently OFF)
jack.orange.mode <front|rear|center/LFE|side|pcm4|input> (currently rear)
jack.orange [<leftvol>:<rightvol>] (currently 29.9:29.9 dB)
jack.orange.mute ON|OFF (currently OFF)
jack.gray.mode <front|rear|center/LFE|side|pcm4|input> (currently pcm4)
jack.gray [<leftvol>:<rightvol>] (currently 29.9:29.9 dB)
jack.gray.mute ON|OFF (currently OFF)
jack.pink.mode <front|rear|center/LFE|side|pcm4|input> (currently input)
jack.pink [<leftvol>:<rightvol>] (currently 19.9:19.9 dB)
jack.pink.mute ON|OFF (currently OFF)
jack.fp-pink.mode <front|rear|center/LFE|side|pcm4|input> (currently front)
jack.fp-pink [<leftvol>:<rightvol>] (currently 29.9:29.9 dB)
jack.fp-pink.mute ON|OFF (currently OFF)
jack.blue.mode <front|rear|center/LFE|side|pcm4|input> (currently input)
jack.blue [<leftvol>:<rightvol>] (currently 29.9:29.9 dB)
jack.blue.mute ON|OFF (currently OFF)
jack.fp-green.mode <front|rear|center/LFE|side|pcm4|input> (currently front)
jack.fp-green [<leftvol>:<rightvol>] (currently 29.9:29.9 dB)
jack.fp-green.mute ON|OFF (currently OFF)
record.mix.mute.mic1 ON|OFF (currently OFF)
record.mix.mute.fp-mic1 ON|OFF (currently OFF)
record.mix.mute.linein1 ON|OFF (currently OFF)
record.mix.mute.fp-headphone1 ON|OFF (currently OFF)
record.mix.mute.green1 ON|OFF (currently OFF)
record.mix.mute.black1 ON|OFF (currently OFF)
record.mix.mute.orange1 ON|OFF (currently OFF)
record.mix.mute.gray1 ON|OFF (currently OFF)
record.mix.mute.input-mix1 ON|OFF (currently OFF)
record.mix1 [<leftvol>:<rightvol>] (currently 38.9:38.9 dB)
record.mix.mute.mic2 ON|OFF (currently OFF)
record.mix.mute.fp-mic2 ON|OFF (currently OFF)
record.mix.mute.linein2 ON|OFF (currently OFF)
record.mix.mute.fp-headphone2 ON|OFF (currently OFF)
record.mix.mute.green2 ON|OFF (currently OFF)
record.mix.mute.black2 ON|OFF (currently OFF)
record.mix.mute.orange2 ON|OFF (currently OFF)
record.mix.mute.gray2 ON|OFF (currently OFF)
record.mix.mute.input-mix2 ON|OFF (currently OFF)
record.mix2 [<leftvol>:<rightvol>] (currently 4.4:2.9 dB)
misc.mic [<leftvol>:<rightvol>] (currently 0.0:0.0 dB)
misc.fp-mic [<leftvol>:<rightvol>] (currently 46.4:37.4 dB)
misc.linein [<leftvol>:<rightvol>] (currently 38.9:38.9 dB)
misc.fp-headphone [<leftvol>:<rightvol>] (currently 34.4:34.4 dB)
misc.green [<leftvol>:<rightvol>] (currently 34.4:38.9 dB)
misc.black [<leftvol>:<rightvol>] (currently 38.9:38.9 dB)
misc.orange [<leftvol>:<rightvol>] (currently 38.9:38.9 dB)
misc.gray [<leftvol>:<rightvol>] (currently 40.4:41.9 dB)
misc.input-mix <mic|fp-mic|linein> (currently mic)
misc.front-mute ON|OFF (currently OFF)
misc.input-mix-mute1 ON|OFF (currently OFF)
misc.front1 [<leftvol>:<rightvol>] (currently 43.4:43.4 dB)
misc.front2 <front|input-mix> (currently front)
misc.rear-mute ON|OFF (currently OFF)
misc.input-mix-mute2 ON|OFF (currently OFF)
misc.rear1 [<leftvol>:<rightvol>] (currently 4.4:4.4 dB)
misc.rear2 <rear|input-mix> (currently rear)
misc.center/lfe-mute ON|OFF (currently OFF)
misc.input-mix-mute3 ON|OFF (currently OFF)
misc.center/lfe1 [<leftvol>:<rightvol>] (currently 41.9:41.9 dB)
misc.center/lfe2 <center/LFE|input-mix> (currently center/LFE)
misc.side-mute ON|OFF (currently OFF)
misc.input-mix-mute4 ON|OFF (currently OFF)
misc.side1 [<leftvol>:<rightvol>] (currently 35.9:35.9 dB)
misc.side2 <side|input-mix> (currently side)
misc.pcm4-mute ON|OFF (currently OFF)
misc.input-mix-mute5 ON|OFF (currently OFF)
misc.pcm41 [<leftvol>:<rightvol>] (currently 25.4:25.4 dB)
misc.pcm42 <pcm4|input-mix> (currently pcm4)
vmix0-enable ON|OFF (currently ON)
vmix0-rate <decimal value> (currently 48000) (Read-only)
vmix0-channels <Stereo|Multich> (currently Stereo)
vmix0-src <Fast|Low|Medium|High|High+|Production|OFF> (currently Medium)
vmix0-outvol <monovol> (currently 25.0 dB)
vmix0-invol <monovol> (currently 25.0 dB)
vmix0.pcm8 [<leftvol>:<rightvol>] (currently 19.9:19.9 dB) ("knotify4")
vmix0.pcm9 [<leftvol>:<rightvol>] (currently 25.0:25.0 dB)
vmix0.pcm10 [<leftvol>:<rightvol>] (currently 25.0:25.0 dB)
vmix0.pcm11 [<leftvol>:<rightvol>] (currently 25.0:25.0 dB)
For libflashsupport on the same page you linked it clearly states:
# Flash V9 and V10 require libflashsupport to output sound via OSS. Typically a 32-bit version of the library is required.
# Flash V10 has a 64-bit version which requires a 64 bit libflashsupport.
Also i tested it with and without libflashsupport. On archlinux (x64 running here) there most certainly is a need for libflashsupport when you want to have sound in your flash. And yes i tested the archlinux OSS version and the mercurial version (running now) bith need it installed manually! On my pc sound in flash didn't work without it but did with it. So, no not removing from the wiki as it's needed. But i see you removed it for me! please do NOT do that if you didn't even verified it. I use flash 10 x64 and i need it!
As for the ffmpeg "codec" changed it to backend.
And i did get the microphone working near perfect: http://www.4front-tech.com/forum/viewtopic.php?p=13192
Now for some news you all might like.
On my school i need to do an investigation to whatever i want and i'm heavily thinking about investigating the pros/cons of alsa compared to oss (or oss compared to alsa). That investigation will take from monday next week till next mondey till friday 23 of oktober. In that investigation i'm going to do some in depth look of alsa and oss and that will include the usability as well.
Following up on that investigation i will spend another 8 weeks on my school making a volume control application that can be used with alsa and oss and easily expandable with other sound systems. The goal of this is to make one sound application that can manage (in the first place) alsa and oss. oss is going to be implemented and alsa is probably going to be dummy implemented because it's likely way to much for me to implement both.
Before you get to exited, both projects (investigating and making the application) are just made up today and i just don't know if both will get accepted by my school. I asked one teacher and he liked the idea a lot and could potentially have a value for the sound management under linux. Once i do get this started i will involve the community (YOU!) with this since this project can't be done without the community specially the investigation.
And once i start and have something to tell/ask i will blog about it on http://blog.mageprojects.com
edit::
And this idea already got dumped. read more a few posts down or click: http://bbs.archlinux.org/viewtopic.php? … 34#p612634
Last edited by markg85 (2009-09-03 17:51:45) -
Where can I find the error log of adobe premiere CC
Adobe premiere CC is doing very weird things this last month.
When I open adobe I get a really weird setup. When I move everything back in order it's ok again. I save the new order under the same name (overlapping) but when I start adobe again, everything is messed up again.
Than premiere is veeeeeery slowly when I output to adobe media encoder. Very slow in choosing the codec, change settings and Render Que.
I even got a very strange and long error within adobe, but I closed it....
I guess that error was the problem and I need to find it back again.
Does adobe premiere create a log file where I can see that error from today again?
Many thanks!And now I got this error.....
-
Hi Everyone,
I have searched the forums and have found similar problems
but no resolutions.
I've created a module in Captivate 3 and used Articulate
Presenter for delivery. It has been working great for the last few
months. A few days ago, I started having this problem (so did
users).
At 8m38s in the module, there are 2 buttons that popup and
give the option to go back to the table of contents or go to the
next slide. There is a 4 second delay (while the narrator finishes
talking) before the slide pauses and wait for a click. If I click
Next before the 4 second delay ends, everything work great. If I
click Next when the slide has paused, it jumps to the next slide,
but with no audio. The slide displays what it is supposed to (even
closed captioning works) and then fades to white and freezes (even
though there are 3 slides left in the module.
Has anyone else had this problem or know a solution? Any
feedback would be greatly appreciated. Thanks all!
Regards,
CoryI reinstalled OSX on my macbook which previously played all avi files with no difficulties. Now I am having difficulties playing Avi files with quicktime PRO that use the Mpeg layer 3 codec. I have done the following with no success: downloaded flip 4 mac, divx, xvid, 3ivx, Xvid codecs, changed MIME settings, and deleted and reinstalled.
Depending on the specific version and sometimes the ordering in which they were installed, you may have created a conflict among the components here. While most current VidX components should be able to play MPEG Layer-3 audio, v5.1.1 is the only free VidX codec that is bi-directional. If that is not a problem, then I would recommend you remove (or "hide") all components in the main "Library > QuickTime" folder. Next I would determine if one (or more) of the AVI associated components can actually play the specific files mentioned on their own. You do this by installing a compenent to be tested, opening the player with one of the problem movies, and checking for correct audio and video playback. Log your results, close the player, remove the tested component, and move on to the next component to be tested. Once you deterime which component(s) may function correctly, I would then move on to determine which specific components are in conflict by repeating the tests but leaving the components installed after each test. In this case start with a component that is known to work and is the one you most desire to use (if more than one works). If the player stops working, then you have located a potential conflict. Set the offending component aside and continue testing until all components have been tested and are either installed or set aside. The last step is to simply evaluate component alternatives and configure you system for those files most often played. In my case the VidX v6 Pro components did not seem to get along with the 3ivX D4 v4.5 and/or Casio AVI and/or XVIDDelegate component combination. -
running imovie on a powermac w 10.6.8. went through the tutorials. imported my clips, got the hang of basic editing, and created a project.
was pretty happy with it, but needed to shut down and come back to work on it later. when i did come back, the entire project was gone.
how do i get the program to save my work in progress, without finalizing it?
thanksSo, everything worked out just fine!
I have all my material in the ProRes Codec on a seperate drive now. The project itself was kept or rather duplicated, the bin structure etc stayed the same but is linking to the newly created files. The playback and workflow is fluent. I am happy!
Time info: For about 120 GB of material the codec change process took 25 hours.
Thanks for your help!
Best
*a -
X-Fi Xtrememusic and all X-Fi products
I have been corresponding with CL techs via email and telephone re: the connection of my xbox360 to my X-Fi XtremeMusic soundcard via optical cable to produce 5. DD from my analogue speakers (Creative T7700). After doing ALOT of research, many ppl (CL lab techs, other sound card companies, forum members) have stated that this is NOT possible as the decoder of DD lies within the software and NOT the hardware of the X-fi soundcard. Thus, the X-Fi XtremeMusic CAN NOT DECODE 5. DD from an xbox360 and you would need an external decoder like DDTS-00. HOWEVER, I came across several CL techs by email and this is what they told me:
Email :
I understand from your email that you like to know if there is specific driver for X-Fi Xtreme Music sound card. I am sorry for any inconvenience this is causing you. Please let me assist you. For your information there is a built-in decoder on the X-Fi Xtreme Music sound card. Therefore the signal from a game console such as an Xbox can be decoded & output to analog speakers. There is no need for an I/O Dri've. You may use a digital Toslink cable to 3.5 mm plug.
Email 2:
With regard to your email, please be informed that the X-FI Extreme Music sound card is able to decode the 5. DD signal via optical connection to analog speakers. Instead of purchasing the I/O Console, you can get a Tosslink to Minijack adaptor for the connection from the Xbox to Xfi Extreme Music sound card. Please be informed that the decoding will only work on Win XP and not on a Windows Vista PC. In case if you need any clarification, please do not hesitate to contact us.
Email 3:
Prior to getting to the email form on our website, you were asked to try some basic troubleshooting for the issue that you identified. The steps that you were walked through resolve some issues with your product. Some of the information that I provide in this message may be repetiti've but it is still valuable information for troubleshooting your device. Yes it can since the X-Fi XtremeMusic has a built-in decoder.
Email 4:
With regard to your email, i believe that you are enquiring on a sound card which is suitable for Xbox connection so that it can output 5. sound via the analog speakers connected to the sound card. To do that, you would need to purchase a XFi Extreme Music sound card and an Xfi I/O Console.
So this concludes my message. I do not know who is right. I deliberately wrote the names of the authors of the emails above so that other Creative techs can clarify with them. The emails have stated that I can connect use the digital connection from my 360 to the flexijack of the x-fi XtremeMusic and get 5. DD sound but others have stated that I cant since the decoder is in the software. Can someone from CL tech explain to me this discrepancy? Thanks, I appreciate it alot.
Message Edited by turok_t on 0-0-2009 0:58 AMMessage Edited by turok_t on 0-0-2009 02:00 AMMessage Edited by turok_t on 0-0-2009 02:05 AMAdmin Notes: Thanks for letting us know, please understand that I have to remove the name of the Customer Support Advisors for their privacy.
Message Edited by KokChoy-CL on <span class='local-date'> 0-2-2009<span class='local-time'> 02:02 [email protected]turok_t wrote:
...the sound card itself can not decode the signal so its useless.
I wouldn't say that. The card doesn't need to. It's nicer if it appears to do so than if application programs have to be relied on.
Yes, I have heard that previous?sound cards (not just Creative) had built in decoders to the hardware. This was for 2 reasons. ) Privacy and copyright reasons from Dolby Digital themselves. 2) Codecs change very quickly which means that the hardware will be obsolete once a new codec is released. Thus, it is more efficient to embedd the codecs in software itself. This is based on one of the correspondences with a CL tech staff member.
There may have been soundcards that included hardware decoding, but I've only heard of one, which was an early USB-interfaced one by CL which was almost stand-alone in character. I have a vague memory of some card with a DSP chip being programmed for it, but I also have this sense this was vaporware or a baseless claim.
Almost all soundcards punt and say they support decoding but really just include a copy of PowerDVD or similar that actually does it (which CL has started doing recently, unfortunately), but CL has pretty much been unique in including a driver-based decoder which makes the card look like it does do decoding. As far as I can tell, the Xtrememusic is one of those. ? In most cases these support digital-in decoding from an external digital input, but you usually needed the I/O dri've to have the digital input it supported that for. As I said, the formal manual shows explictly how to make an optical connection from a DVD player to the I/O Dri've or I/O console expressly so the card's decoding can be used to hear DD/DTS on the computer's analog speakers.
As far as your comment about hardware becoming obsolete, why haven't I heard about this being a problem for DD/DTS receivers. That's hardware,
isn't it's
Yes, you will need to specify the source if multiple sources are connected to the sound card.
Only if more than one is supported and it is not the default. It is entirely possible that if there is more than one external digital input, only one of them is actually supported as a decoding input.
Well, I think its kinda dumb to put a flexijack that is nonfunctional for digital I/O unless you buy the I/O console or dri've separately. Many consumers, including me, thought that external devices can be directly connected to the flexijack since it is Digital I/O. IMO, this is a bit misleading.
There is only so much real estate on the card's slot cover, and you can't help that some people are apt to jump to unwarranted conclusions. In fact, you can connect external devices to the flexijack directly, in the output direction such as connecting the card to a receiver's coaxial digital input, but you need the Digital I/O Module to do that optically, or to do either form of digital input. You can do so without the I/O console or dri've in addition. The question is not that the flexijack cannot be used for digital input, it's whether the card's decoding feature supports decoding from it, or only from the digital inputs on the dri've/console. It might, but to say it does or doesn't would be an unwarranted conclusion, since I don't have confirmation either way. I also don't know the effect of Vista, if you're using it, but I know that once again drivers had to be rewritten for it, and when that happens, esoteric features may get broken or not work for awhile. (Microsoft seems to be quite incapable of architecting a stable driver environment.)? Plus, drivers rewritten for Vista might force CL to renegotiate the Dolby licensing, which even might be the reason CL has stopped supplying driver-based decoding in some cases.
To me, it doesnt really matter how many external digital inputs are connected or whether you can switch between digital inputs?from multiple input sources. The fact remains that the Xtrememusic currently can not decode 5. DD li've regardless if you have an adaptor (optical>3/5mm), I/O module, I/O console. In other words,?you can find many ways to connect external devices to the sound card, but the sound card cant INTERPRET and DECODE those digital inputs.
It can.
-Dave
[email protected] -
Sync problem converting videos with SU
Hi there. I have owned my Creative Zen Vision:m 30gig player for about 2 weeks now. So far I have really come to love this device. I have learned how to put album art on all my favourite albums with ?Album Art Downloader? and I am using Media Monkey to manage and sync my mp3 files from my pc to my Vision:m. I have even loaded all my photos onto the player and bought the accessories pack so that I can hook it up to my television ? it works perfectly.
My last challenge (so far) was to figure out how to make this video thing work as I have about 00 videos which I have downloaded from Google video via a plugin for Mozilla Firefox called ?Video Downloader ?. The videos from Google video are encoded into something called a .flv file. I have done A LOT of reading here and other places on how to convert the various formats into formats the player can understand. Apparently the program ?Super? is THE thing for converting almost anything into well anything. I tried it and found out I could select ALL of my videos and select a common setting for them all to be encoded into. My first reaction?. sweeeeeet! The videos did not seem to have lost much quality and when I loaded them to the player, it would play them nicely? or so I thought.
It did however slowly but inevitably become clear to me that despite my most fierce attempts of denial? there was a sync problem with the audio. Not a huge annoyance but still an annoyance, something which I knew I couldn?t li've with. I therefore rushed to the program and tried to shuffle the settings a bit, changed the codecs changed the containers but nothing would work. The sound sync problem would still be there, lurking. The problem was also present when converting high quality xvid movies.
This is where my creativity and patience has come to an end. This is where I give up and crawl to you begging for help. I in order for you to help me, I will try to help YOU recreate my scenario.
I have a video, perhaps a little morbid, where the sync problem is very obvious after you have converted it with super, here is the link http://video.google.com/videoplay?docid=-7233445036972827652
I use a plugin for Firefox to download videos from Youtube, Google video, etc. called ?Video Downloader?, here?s the link
https://addons.mozilla.org/firefox/2390/
The settings I use for Super are the following:
Output Container: Avi (For Pocket Pc)
Output Video Codec: Mpeg4
Video Scale Size: 320*240
Frame/Sec: 25
Bitrate: 336
Hi Quality: ?Checked?
Stretch It: ?Checked?
Sampling Freq: 22050
Bitrate kbps: 28
Ok if you manage to download the clip from google video (easy) AND convert with Super into a format fitting for the Zen Vision:m without the sync problem? well my joy will break any ?happy-meter? out there
Ps. I did actually find a way to fix the sync problem, I checked the ?Directshow? for the ffmpeg but the picture was sort of lagging and looking REALLY ugly so that actually created a problem much worse than the previous one.
Thank you for your time and patienceI am not sure I can help you but I do have one pice of information. The sync problems arise from converting to the wrong frame rate. If you can choose the frame rate for the videos you convert to then you can test different rates and find the right rate. Keep in mind that there is currently no way to determine the approiate frome rate without teting it.
[Edit]
BTW is that MS-mpeg4-v or MS-mpeg4-v2?Message Edited by Edward-Nardella on 2-26-200602:08 PM -
Blackmagic Intensity - Crashes during stream every time
Howdy folks. I'm having an immense problem with attempting to stream both in Mac OSX and Windows 7 (same box) with my BlackMagic Intensity Pro.
Everything is updated, and I can stream with the crappy web-based tools. No matter what I do, be it resolution changes, codec changes etc, at around 28 seconds, I get:
"Unexpected error encountered in encoding process"
My Internet connection is 26 Meg down, 6 Meg up, and I am not throttled.
Any ideas? No matter where I've looked, I just can't find an answer to this.
Thanks in advance for any help that you can provide.I have the same card sitting in my PC running osx Snow Leopard 10.6.5 and it works with the HD 1080i50 and composit settings that´s it. Even sound works at least hdmi in from my cam, I haven´t tried the audio in cables yet. I´m casting over at justintv, livestream and blogtv.
Might be if you run 64 bit os you get this for some reason so try 32 bit to check that first. I´ve had problems with fmle and camtwist a few times with my new box (i7 930, asus P6T-SE, ati 5750 videcordhad) had to reinstall them a few times and found out I really had to make sure I repaired permissions. Sometimes it seems to be best to do it with diskutility and other times with pfix, really strange don´t why. Pfix will clear your caches so that might be why it sometimes is like that. You might have updated you chameleon or efi or something. I mean lots of stuff like this and the above here can mess it up for you.
The old comp GA EP45-DS3 worked ok with this card too but not as good as my new.
I use Camtwist FMLE Soundflower and Linein and I can stream "wide" screen with the best soundsettings you get with FMLE. My settings in Camtwist and FMLE depends on what I wanna do. I go for 720*480 in in both Camtwist and FMLE and 720*404 out with a video and desktop + pip cam cast and for a cam cast only with my Panasonic HD cam or my Logitech web cam when I use both Camtwist and FMLE it´s 1280*720 for both in and out. I can cast Youtube videos right of my desktop and even videos from Plex. Soundflower takes care of the sound and if I want to route sound in from external sources I use Linein.
The Livestream Procaster works really good too and they have the best video on the internet so far, the expand video feature they have is the biggest window with the best resolution you can get at this point. -
Lightroom 4 video black thumbnails and playback
On loading I see thumbnails for avi videos for about a second and then they go black. On playing in Lightroom the screen is black but the sound is fine. The videos play correctly in Quicktime and other programs.
Any suggestions on how to fix this?
LR4 Windows 7, Core i7, ATI Mobile Radeon HD5850, 8Gb RAMI used a video conversion program to convert to a different codec. It probably does not need a full codec pack installed, just the coded changed. Google 'codec changer software'. I have not tried any of the free ones so make sure that you make a copy of any file before converting it if the program overwrites the original file.
http://download.cnet.com/windows/video-converters/?tag=bc has some programs -
Exporting ProRes conversion of HDV into Motion 3 darkens image
When I export an HDV clip from FCP into Motion 3, apply Optical Flow frame blending, then export to FCP as an HDV file, the brightness and saturation of the resulting QT HDV file are essentially identical to the original. But when I begin by exporting the original clip to ProRes(HQ) via QT conversion or Compressor, bring the ProRes file into Motion and afterwards export to FCP as HDV, the resulting clip is considerably darker (and maybe more saturated) than the original. In the QT conversion, I'm not applying any brightness or contrast filtering. Any clues about what I'm doing wrong?
Alfred Guzzettihi,
why do you need to convert the codec into motion and then back again into FCP. Why not send to motion from fcp keeping the same format, and then drop thew motion file itself onto the fcp timeline and use it as a clip. ( although the need to render it everytime something moves can be a pain I know). I find that the less codec changes in a work flow the better.
hth
adam
Maybe you are looking for
-
Way can't i do this now? Last year I would turn into creativeLive and watch tutorials on my tv using my Apple TV screening from my iPad . This was great. And allowed me to move around while learning. After Maverick that service disappear, I went into
-
IPad mini backlight issues after digitizer replacement
I recently replaced the digitizer on my 1st gen iPad mini, and after reassembly, I'm havering issues with the backlight. Please have a look at my photo and let me know what the issue may be. Thanks a lot, Dave
-
Font Licensing Restriction Issue
I recently transferred a lot of ttf and otf fonts (originally purchased for and used on a Windows NT machine) to Mac Book Pro. When I try to use the fonts in Abode CS5.5 (In illustrator) I'm told I need a license. Won't let me embed the fonts, for ex
-
Hello, I want to do multiple transformation. i have many xsl files and one xml file. i want to generate single html with diffrent xsl. My question is "is there any case where we can give multiple xsl as a source?" I hope all can understand my problem
-
Each time I synk my iPhone for the appdates, it takes forever, because it goes through denius update for 5,000 songs. I noticed this when I updtaed to iOS5 Is this required? I don't use genious on iphone, can I shot it down, from syncing? Thanks,