[directmusic] Re: Compression: ADPCM?
- From: "justin" <jlove@xxxxxxx>
- To: <directmusic@xxxxxxxxxxxxx>
- Date: Sat, 15 Mar 2003 17:08:26 -0800
Encapsulation in this context means hiding complexity--like the
cMusicManager class that MS provides in dmutil. Here is the code for a
DirectX-based Ogg Vorbis player, it should be easy to convert to play
Ogg Vorbis files in whatever application you planned. The code isn't
mine and I can't remember where I got it from but we were able to use it
to stream Ogg Vorbis files into our game...
Justin Love
University of Victoria
/*
This is a simple Ogg Vorbis player class for DirectX 8. Why Ogg Vorbis?
Because MP3 is evil ;) You have to pay Fraunhofer whenever you code an
encoder / decoder. Ogg Vorbis on the other hand is really free.
You'll need the Ogg Vorbis Sdk which you can obtained at
http://www.vorbis.com/download_win.psp.
Link the following files to your project: vorbisfile_static.lib,
vorbis_static.lib, ogg_static, dsound.lib. Also be sure your include
paths contain the SDK.
To use the OggPlayer Class simply do the following:
OggPlayer op;
op.InitDirectSound(hwnd);
op.OpenOgg("somemusic.ogg");
op.Play();
while(op.IsPlaying());
op.Close();
You may use this code for whatever you want. :)
*/
/**************************************************************
* OggPlayer.h
***************************************************************/
#include <windows.h> // from your fave include dir ;)
#include <mmreg.h>
#include <dsound.h> // from the directx 8 sdk
#include <vorbis/codec.h> // from the vorbis sdk
#include <vorbis/vorbisfile.h> // also :)
#define BUFSIZE 65536*10 // buffer length
class OggPlayer
{
protected:
bool bInitialized; // initialized?
bool bFileOpened; // have we opened an ogg yet?
bool bReleaseDS; // release ds by ourselves?
LPDIRECTSOUND8
pDS; // the directsound 8 object
LPDIRECTSOUNDBUFFER
pDSB; // the buffer
OggVorbis_File
vf; // for the vorbisfile interface
int nLastSection, // which half of the buffer
are/were
nCurSection; // we playing?
bool bAlmostDone; // only one half of the buffer
to play
bool bDone; // done playing
bool bLoop; // loop?
public:
OggPlayer();
~OggPlayer();
bool // initialize dsound ..
InitDirectSound( HWND hWnd );
inline void // .. or use already initialized
UseDirectSound( LPDIRECTSOUND8 _pDS )
{
pDS = _pDS;
}
bool // this opens an oggvorbis for
playing
OpenOgg( char *filename );
void // and this one closes it :)
Close();
void // play it again sam
Play(bool loop = false);
void // stop it
Stop();
void // be sure to call this from
time to time
Update();
inline bool IsPlaying()
{
return !bDone;
}
};
/****************************************************
* OggPlayer.cpp
****************************************************/
#include "oggplayer.h"
OggPlayer::OggPlayer()
{
bFileOpened = false;
bInitialized = false;
bReleaseDS = false;
pDS = NULL;
pDSB = NULL;
bLoop = false;
bDone = false;
bAlmostDone = false;
}
OggPlayer::~OggPlayer()
{
if (bFileOpened)
Close();
if (bReleaseDS && pDS)
pDS->Release();
}
bool OggPlayer::InitDirectSound( HWND hWnd )
{
HRESULT hr;
if (FAILED(hr = DirectSoundCreate8(NULL, &pDS, NULL)))
return bInitialized = false;
pDS->Initialize(NULL);
pDS->SetCooperativeLevel( hWnd, DSSCL_PRIORITY );
bReleaseDS = true;
return bInitialized = true;
}
bool OggPlayer::OpenOgg( char *filename )
{
if (!bInitialized)
return false;
if (bFileOpened)
Close();
FILE *f;
f = fopen(filename, "rb");
if (!f) return false;
ov_open(f, &vf, NULL, 0);
// ok now the tricky part
// the vorbis_info struct keeps the most of the interesting format
info
vorbis_info *vi = ov_info(&vf,-1);
// set the wave format
WAVEFORMATEX wfm;
memset(&wfm, 0, sizeof(wfm));
wfm.cbSize = sizeof(wfm);
wfm.nChannels = vi->channels;
wfm.wBitsPerSample = 16; // ogg vorbis is always
16 bit
wfm.nSamplesPerSec = vi->rate;
wfm.nAvgBytesPerSec = wfm.nSamplesPerSec*wfm.nChannels*2;
wfm.nBlockAlign = 2*wfm.nChannels;
wfm.wFormatTag = 1;
// set up the buffer
DSBUFFERDESC desc;
desc.dwSize = sizeof(desc);
desc.dwFlags = 0;
desc.lpwfxFormat = &wfm;
desc.dwReserved = 0;
desc.dwBufferBytes = BUFSIZE*2;
pDS->CreateSoundBuffer(&desc, &pDSB, NULL );
// fill the buffer
DWORD pos = 0;
int sec = 0;
int ret = 1;
DWORD size = BUFSIZE*2;
char *buf;
pDSB->Lock(0, size, (LPVOID*)&buf, &size, NULL, NULL,
DSBLOCK_ENTIREBUFFER);
// now read in the bits
while(ret && pos<size)
{
ret = ov_read(&vf, buf+pos, size-pos, 0, 2, 1, &sec);
pos += ret;
}
pDSB->Unlock( buf, size, NULL, NULL );
nCurSection =
nLastSection = 0;
return bFileOpened = true;
}
void OggPlayer::Close()
{
bFileOpened = false;
if (pDSB)
pDSB->Release();
}
void OggPlayer::Play(bool loop)
{
if (!bInitialized)
return;
if (!bFileOpened)
return;
// play looping because we will fill the buffer
pDSB->Play(0,0,DSBPLAY_LOOPING);
bLoop = loop;
bDone = false;
bAlmostDone = false;
}
void OggPlayer::Stop()
{
if (!bInitialized)
return;
if (!bFileOpened)
return;
pDSB->Stop();
}
void OggPlayer::Update()
{
DWORD pos;
pDSB->GetCurrentPosition(&pos, NULL);
nCurSection = pos<BUFSIZE?0:1;
// section changed?
if (nCurSection != nLastSection)
{
if (bDone && !bLoop)
Stop();
// gotta use this trick 'cause otherwise there wont be played
all bits
if (bAlmostDone && !bLoop)
bDone = true;
DWORD size = BUFSIZE;
char *buf;
// fill the section we just left
pDSB->Lock( nLastSection*BUFSIZE, size, (LPVOID*)&buf, &size,
NULL, NULL, 0 );
DWORD pos = 0;
int sec = 0;
int ret = 1;
while(ret && pos<size)
{
ret = ov_read(&vf, buf+pos, size-pos, 0, 2, 1, &sec);
pos += ret;
}
// reached the and?
if (!ret && bLoop)
{
// we are looping so restart from the beginning
// NOTE: sound with sizes smaller than BUFSIZE may be cut
off
ret = 1;
ov_pcm_seek(&vf, 0);
while(ret && pos<size)
{
ret = ov_read(&vf, buf+pos, size-pos, 0, 2, 1, &sec);
pos += ret;
}
}
else if (!ret && !(bLoop))
{
// not looping so fill the rest with 0
while(pos<size)
*(buf+pos)=0; pos ++;
// and say that after the current section no other sectin
follows
bAlmostDone = true;
}
pDSB->Unlock( buf, size, NULL, NULL );
nLastSection = nCurSection;
}
}
-----Original Message-----
From: directmusic-bounce@xxxxxxxxxxxxx
[mailto:directmusic-bounce@xxxxxxxxxxxxx] On Behalf Of Paul Stroud
Sent: March 13, 2003 3:53 AM
To: directmusic@xxxxxxxxxxxxx
Subject: [directmusic] Re: Compression: ADPCM?
Hi Justin,
I have no idea what encapsulation means, but it sounds good. I've heard
of
Ogg before - does the encapsulation install the codec, because it's for
a
showreel
Cheers
Paul
> -----Original Message-----
> From: directmusic-bounce@xxxxxxxxxxxxx
> [mailto:directmusic-bounce@xxxxxxxxxxxxx]On Behalf Of justin
> Sent: 12 March 2003 17:49
> To: directmusic@xxxxxxxxxxxxx
> Subject: [directmusic] Re: Compression: ADPCM?
>
>
>
> I used Ogg Vorbis (similar compression and quality to MP3, except it
is
> open source and license-free) for a game last year. If you are
> interested I can post a nice encapsulation we used.
>
> Justin Love
> University of Victoria
>
> -----Original Message-----
> From: directmusic-bounce@xxxxxxxxxxxxx
> [mailto:directmusic-bounce@xxxxxxxxxxxxx] On Behalf Of Bjorn Lynne
> Sent: March 12, 2003 5:39 AM
> To: directmusic@xxxxxxxxxxxxx
> Subject: [directmusic] Re: Compression: ADPCM?
>
>
> Hi Paul,
>
> > Anyone know if there is anything else that will compress my wavs
into
> ADPCM
> > format other than DMP?
> >
> > I've heard that this compression can sound a bit grainy or slightly
> harsh,
> > but an external editor, like wavelab, might be able to render my
files
> a
> > touch more elegantly.
>
> You can save out ADPCM from SoundForge, but don't expect any miracles.
> ADPCM
> compression sounds great with *most* sounds, but a few sounds will
sound
> very gritty and harsh, no matter which program performed the
> compression.
>
> > I was tempted to go the mp3 route, but this 2.1 second thing sound
> like a
> > real pain - is it DMP that adds the glitch on play back? If you
encode
> in
> an
> > external editor and replace the runtime files, it must be in the
play
> back
> > that the glitch occurs.
>
> I'm not sure if you are talking about the little bit of silence added
to
> the
> beginning and end of the mp3 file, making it unsuitable for sounds
that
> need
> to start immediately, or looping sounds... but this is a weakness of
the
> mp3
> file format. I investigated this a lot a couple of years ago
(including
> communicating directly with one of the guys at Frauhofer who made the
> format), and there is apparently no way to create mp3 files that don't
> have
> this little bit of silence at the beginning and end... making mp3
> unsuitable
> for looping or low-latency playback.
>
> Actually, there is ONE program that does seem to make it possible:
> Flash. If
> you import a WAV file into Flash and set the compression to "mp3
> compression" inside Flash, then you can save out your Flash movie,
which
> will include an embedded mp3, which *will* loop smoothly in Flash
> Player.
> I'm guessing that this is because Flash has compressed to mp3 itself,
it
> knows exactly how much silence has been added to the beginning and
end,
> and
> is able to skip these parts when it plays back.
>
> > Is any of these compression formats more CPU friendly on decode?
>
> I believe mp3 is a great deal more CPU intensive than ADPCM. Somebody
> correct me if I'm wrong.
>
> Cheers,
> --
> Bjorn Lynne - Composer, Producer, Sound Designer
> Main music site: www.lynnemusic.com
>
>
>
>
>
>
>
>
>
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