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OpenAL Streaming Sound stuff 

from zeq2, but adapted/stripped down for our needs.

Videos work (for me), music does not (yet)
This commit is contained in:
Daniel Gibson 2012-04-22 15:59:18 +00:00
parent 6384b850f1
commit 04fd7f2cda
4 changed files with 211 additions and 52 deletions
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1
src/client/sound/header/local.h
View File

@ -181,6 +181,7 @@ void AL_StopChannel( channel_t *ch );
void AL_PlayChannel( channel_t *ch );
void AL_StopAllChannels( void );
void AL_Update( void );
void AL_RawSamples( int samples, int rate, int width, int channels, byte *data, float volume );
#endif
#endif

242
src/client/sound/snd_al.c
View File

@ -1,8 +1,9 @@
/*
This Source File was taken from the Q2Pro Port.
This Source File - except for the last few functions - see the other copyright
notice further down the - was taken from the Q2Pro Port.
Copyright (C) 2010 skuller.net
2012 Some changes by the Yamagi Quake2 developers
2012 Some changes by the Yamagi Quake II developers
This program is free software; you can redistribute it and/or
modify it under the terms of the GNU General Public License
@ -26,6 +27,7 @@ Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA.
#include "../header/client.h"
#include "header/local.h"
#include "header/qal_api.h"
#include "header/vorbis.h"
// translates from AL coordinate system to quake
#define AL_UnpackVector(v) -v[1],v[2],-v[0]
@ -34,9 +36,17 @@ Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA.
// OpenAL implementation should support at least this number of sources
#define MIN_CHANNELS 16
static ALuint s_srcnums[MAX_CHANNELS];
qboolean streamPlaying;
static ALuint streamSource;
static ALuint s_srcnums[MAX_CHANNELS-1];
static int s_framecount;
// Forward Declarations
static void S_AL_StreamUpdate( void );
static void S_AL_StreamDie( void );
// /Forward Declarations
void AL_SoundInfo( void ) {
Com_Printf( "AL_VENDOR: %s\n", qalGetString( AL_VENDOR ) );
Com_Printf( "AL_RENDERER: %s\n", qalGetString( AL_RENDERER ) );
@ -45,6 +55,18 @@ void AL_SoundInfo( void ) {
Com_Printf( "Number of sources: %d\n", s_numchannels );
}
static void AL_InitStreamSource() {
qalSourcei (streamSource, AL_BUFFER, 0 );
qalSourcei (streamSource, AL_LOOPING, AL_FALSE );
qalSource3f(streamSource, AL_POSITION, 0.0, 0.0, 0.0);
qalSource3f(streamSource, AL_VELOCITY, 0.0, 0.0, 0.0);
qalSource3f(streamSource, AL_DIRECTION, 0.0, 0.0, 0.0);
qalSourcef (streamSource, AL_ROLLOFF_FACTOR, 0.0 );
qalSourcei (streamSource, AL_SOURCE_RELATIVE, AL_TRUE );
// srcList[cursrc].scaleGain = 0.0f; FIXME - something like that?
}
qboolean AL_Init( void ) {
int i;
@ -61,19 +83,29 @@ qboolean AL_Init( void ) {
// generate source names
qalGetError();
for( i = 0; i < MAX_CHANNELS; i++ ) {
qalGenSources( 1, &s_srcnums[i] );
if( qalGetError() != AL_NO_ERROR ) {
break;
}
}
qalGenSources( 1, &streamSource );
if( qalGetError() != AL_NO_ERROR )
{
Com_Printf( "ERROR: Couldn't get a single Source.\n" );
goto fail;
if( i < MIN_CHANNELS ) {
Com_Printf( "ERROR: Required at least %d sources, but got %d.\n", MIN_CHANNELS, i );
goto fail;
} else {
// -1 because we already got one channel for streaming
for( i = 0; i < MAX_CHANNELS - 1; i++ ) {
qalGenSources( 1, &s_srcnums[i] );
if( qalGetError() != AL_NO_ERROR ) {
break;
}
}
if( i < MIN_CHANNELS - 1 ) {
Com_Printf( "ERROR: Required at least %d sources, but got %d.\n", MIN_CHANNELS, i+1 );
goto fail;
}
}
s_numchannels = i;
AL_InitStreamSource();
Com_Printf( "OpenAL initialized.\n" );
return true;
@ -92,7 +124,7 @@ void AL_Shutdown( void ) {
memset( s_srcnums, 0, sizeof( s_srcnums ) );
s_numchannels = 0;
}
S_AL_StreamDie();
QAL_Shutdown();
}
@ -204,6 +236,8 @@ void AL_StopAllChannels( void ) {
continue;
AL_StopChannel( ch );
}
s_rawend = 0;
S_AL_StreamDie();
}
static channel_t *AL_FindLoopingSound( int entnum, sfx_t *sfx ) {
@ -291,34 +325,6 @@ static void AL_IssuePlaysounds( void ) {
break;
S_IssuePlaysound (ps);
}
// TODO: streaming sounds from s_rawsamples, equivalent to code below - see also ioq3's code
#if 0
/* clear the paint buffer */
if ( s_rawend < paintedtime )
{
memset( paintbuffer, 0, ( end - paintedtime ) * sizeof ( portable_samplepair_t ) );
}
else
{
/* copy from the streaming sound source */
int s;
int stop;
stop = ( end < s_rawend ) ? end : s_rawend;
for ( i = paintedtime; i < stop; i++ )
{
s = i & ( MAX_RAW_SAMPLES - 1 );
paintbuffer [ i - paintedtime ] = s_rawsamples [ s ];
}
for ( ; i < end; i++ )
{
paintbuffer [ i - paintedtime ].left = paintbuffer [ i - paintedtime ].right = 0;
}
}
#endif
}
void AL_Update( void ) {
@ -326,13 +332,13 @@ void AL_Update( void ) {
channel_t *ch;
vec_t orientation[6];
/*
/* FIXME
if( !s_active ) {
return;
}
*/
paintedtime = cl.time;
// paintedtime = cl.time; // FIXME: this does *not* make music work.
// set listener parameters
qalListener3f( AL_POSITION, AL_UnpackVector( listener_origin ) );
@ -380,10 +386,156 @@ void AL_Update( void ) {
// add loopsounds
AL_AddLoopSounds ();
// FIXME: About here some ogg vorbis stuff (maybe calling OGG_Stream()
// like in S_Update()) should be done.
// add music
OGG_Stream();
S_AL_StreamUpdate();
AL_IssuePlaysounds();
//paintedtime = <begintime of last pending play>; ??
// evtl paintedtime = soundtime falls paintedtime < soundtime - aber soundtime ist eigentlich DMA spezifisch
// paintedtime = cl.time; FIXME: das allein reicht nicht damit musik funktioniert
}
/*
===========================================================================
* The remaining functions in this file are from zeq2 who seem to have got
* them from ioquake3. Adapted for Yamagi Quake II (e.g. we only need one
* non-spatialized stream)
*
* They came with the following copyright notice (modified to make clear
* that only the rest of this file is affected):
Copyright (C) 1999-2005 Id Software, Inc.
Copyright (C) 2005 Stuart Dalton (badcdev@gmail.com)
2012 Some changes by the Yamagi Quake II developers
The rest of this file is part of Quake III Arena source code.
Quake III Arena source code is free software; you can redistribute it
and/or modify it under the terms of the GNU General Public License as
published by the Free Software Foundation; either version 2 of the License,
or (at your option) any later version.
Quake III Arena source code 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 General Public License for more details.
You should have received a copy of the GNU General Public License
along with Quake III Arena source code; if not, write to the Free Software
Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA
===========================================================================
*/
static void S_AL_StreamDie( void )
{
int numBuffers;
streamPlaying = false;
qalSourceStop(streamSource);
// Un-queue any buffers, and delete them
qalGetSourcei( streamSource, AL_BUFFERS_PROCESSED, &numBuffers );
while( numBuffers-- )
{
ALuint buffer;
qalSourceUnqueueBuffers(streamSource, 1, &buffer);
qalDeleteBuffers(1, &buffer);
}
// S_AL_FreeStreamChannel(stream);
}
static void S_AL_StreamUpdate( void )
{
int numBuffers;
ALint state;
// Un-queue any buffers, and delete them
qalGetSourcei( streamSource, AL_BUFFERS_PROCESSED, &numBuffers );
while( numBuffers-- )
{
ALuint buffer;
qalSourceUnqueueBuffers(streamSource, 1, &buffer);
qalDeleteBuffers(1, &buffer);
}
// Start the streamSource playing if necessary
qalGetSourcei( streamSource, AL_BUFFERS_QUEUED, &numBuffers );
qalGetSourcei(streamSource, AL_SOURCE_STATE, &state);
if(state == AL_STOPPED)
{
streamPlaying = false;
// If there are no buffers queued up, release the streamSource
//if( !numBuffers )
// S_AL_FreeStreamChannel( stream );
}
if( !streamPlaying && numBuffers )
{
qalSourcePlay( streamSource );
streamPlaying = true;
}
}
static ALuint S_AL_Format(int width, int channels)
{
ALuint format = AL_FORMAT_MONO16;
// Work out format
if(width == 1)
{
if(channels == 1)
format = AL_FORMAT_MONO8;
else if(channels == 2)
format = AL_FORMAT_STEREO8;
}
else if(width == 2)
{
if(channels == 1)
format = AL_FORMAT_MONO16;
else if(channels == 2)
format = AL_FORMAT_STEREO16;
}
return format;
}
void AL_RawSamples( int samples, int rate, int width, int channels, byte *data, float volume )
{
ALuint buffer;
ALuint format;
format = S_AL_Format( width, channels );
// Create a buffer, and stuff the data into it
qalGenBuffers(1, &buffer);
qalBufferData(buffer, format, (ALvoid *)data, (samples * width * channels), rate);
// set volume
qalSourcef( streamSource, AL_GAIN, volume );
// Shove the data onto the streamSource
qalSourceQueueBuffers(streamSource, 1, &buffer);
// emulate behavior of S_RawSamples for s_rawend
s_rawend += samples;
/*
// get this buffers speed/frequency
ALint speed = 0;
qalGetBufferi(buffer, AL_FREQUENCY, &speed);
// FIXME: the buffers are leaking I think
float scale = (float) rate / speed;
if( samples > 0 && scale >= 0.0f ) {
s_rawend += samples/scale;
}
*/
}
#endif // USE_OPENAL

18
src/client/sound/snd_dma.c
View File

@ -1101,16 +1101,20 @@ S_RawSamples ( int samples, int rate, int width, int channels, byte *data, float
return;
}
// FIXME!!
if( sound_started == SS_OAL )
return;
if ( s_rawend < paintedtime )
{
s_rawend = paintedtime;
}
scale = (float) rate / dma.speed; // FIXME: dma.speed is not available (0) when using openal
#if USE_OPENAL
if( sound_started == SS_OAL )
{
AL_RawSamples(samples, rate, width, channels, data, volume);
return;
}
#endif
scale = (float) rate / dma.speed;
intVolume = (int) (256 * volume);
if ( ( channels == 2 ) && ( width == 2 ) )
@ -1243,7 +1247,9 @@ S_Update ( vec3_t origin, vec3_t forward, vec3_t right, vec3_t up )
* dma buffer while loading */
if ( cls.disable_screen )
{
S_ClearBuffer();
if (sound_started == SS_DMA ) {
S_ClearBuffer();
}
return;
}

2
src/client/sound/snd_vorbis.c
View File

@ -602,7 +602,7 @@ OGG_Stream ( void )
return;
}
while ( ogg_status == PLAY && paintedtime + MAX_RAW_SAMPLES - 2048 > s_rawend )
while ( ogg_status == PLAY && paintedtime + MAX_RAW_SAMPLES - 2048 > s_rawend )
{
OGG_Read();
}