audiomgr.c
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/******************************************************************************
* audiomgr.c
*
* This code implements the audio manager. This provides a low level
* interface to the audio library, and manages the routines needed to
* create the audio task.
*
* At the begining of each video retrace, the scheduler sends a message
* that wakes up the audio thread, which calls alAudioFrame to build an
* audio task. Once this task is built, it is sent to the scheduler, that
* will then send the task to the rsp for execution.
*
* Copyright 1993, Silicon Graphics, Inc.
* All Rights Reserved.
*
* This is UNPUBLISHED PROPRIETARY SOURCE CODE of Silicon Graphics,
* Inc.; the contents of this file may not be disclosed to third
* parties, copied or duplicated in any form, in whole or in part,
* without the prior written permission of Silicon Graphics, Inc.
*
* RESTRICTED RIGHTS LEGEND:
* Use, duplication or disclosure by the Government is subject to
* restrictions as set forth in subdivision (c)(1)(ii) of the Rights
* in Technical Data and Computer Software clause at DFARS
* 252.227-7013, and/or in similar or successor clauses in the FAR,
* DOD or NASA FAR Supplement. Unpublished - rights reserved under the
* Copyright Laws of the United States.
*****************************************************************************/
/*---------------------------------------------------------------------*
Copyright (C) 1998 Nintendo. (Originated by SGI)
$RCSfile: audiomgr.c,v $
$Revision: 1.1.1.1 $
$Date: 2002/05/02 03:27:33 $
*---------------------------------------------------------------------*/
#include <ultralog.h>
#include <assert.h>
#include "audio.h"
#include "simple.h"
/**** type define's for structures unique to audiomgr ****/
typedef union {
struct {
short type;
} gen;
struct {
short type;
struct AudioInfo_s *info;
} done;
OSScMsg app;
} AudioMsg;
typedef struct AudioInfo_s {
short *data; /* Output data pointer */
short frameSamples; /* # of samples synthesized in this frame */
OSScTask task; /* scheduler structure */
AudioMsg msg; /* completion message */
} AudioInfo;
typedef struct {
Acmd *ACMDList[NUM_ACMD_LISTS];
AudioInfo *audioInfo[NUM_OUTPUT_BUFFERS];
OSThread thread;
OSMesgQueue audioFrameMsgQ;
OSMesg audioFrameMsgBuf[MAX_MESGS];
OSMesgQueue audioReplyMsgQ;
OSMesg audioReplyMsgBuf[MAX_MESGS];
ALGlobals g;
} AMAudioMgr;
typedef struct
{
ALLink node;
u32 startAddr;
u32 lastFrame;
char *ptr;
} AMDMABuffer;
typedef struct
{
u8 initialized;
AMDMABuffer *firstUsed;
AMDMABuffer *firstFree;
} AMDMAState;
/**** audio manager globals ****/
extern OSSched sc;
extern OSMesgQueue *sched_cmdQ;
AMAudioMgr __am;
static u64 audioStack[AUDIO_STACKSIZE/sizeof(u64)];
AMDMAState dmaState;
AMDMABuffer dmaBuffs[NUM_DMA_BUFFERS];
u32 audFrameCt = 0;
u32 nextDMA = 0;
u32 curAcmdList = 0;
u32 minFrameSize;
u32 frameSize;
u32 maxFrameSize;
u32 maxRSPCmds;
/** Queues and storage for use with audio DMA's ****/
OSIoMesg audDMAIOMesgBuf[NUM_DMA_MESSAGES];
OSMesgQueue audDMAMessageQ;
OSMesg audDMAMessageBuf[NUM_DMA_MESSAGES];
/**** private routines ****/
static void __amMain(void *arg);
static s32 __amDMA(s32 addr, s32 len, void *state);
static ALDMAproc __amDmaNew(AMDMAState **state);
static u32 __amHandleFrameMsg(AudioInfo *, AudioInfo *);
static void __amHandleDoneMsg(AudioInfo *);
static void __clearAudioDMA(void);
/******************************************************************************
* Audio Manager API
*****************************************************************************/
void amCreateAudioMgr(ALSynConfig *c, OSPri pri, amConfig *amc)
{
u32 i;
f32 fsize;
dmaState.initialized = FALSE; /* Reset this before the first call to __amDmaNew */
c->dmaproc = __amDmaNew;
c->outputRate = osAiSetFrequency(amc->outputRate);
/*
* Calculate the frame sample parameters from the
* video field rate and the output rate
*/
fsize = (f32) amc->framesPerField * c->outputRate / (f32) 60;
frameSize = (s32) fsize;
if (frameSize < fsize)
frameSize++;
if (frameSize & 0xf)
frameSize = (frameSize & ~0xf) + 0x10;
minFrameSize = frameSize - 16;
maxFrameSize = frameSize + EXTRA_SAMPLES + 16;
alInit(&__am.g, c);
dmaBuffs[0].node.prev = 0;
dmaBuffs[0].node.next = 0;
for (i=0; i<NUM_DMA_BUFFERS-1; i++)
{
alLink((ALLink*)&dmaBuffs[i+1],(ALLink*)&dmaBuffs[i]);
dmaBuffs[i].ptr = alHeapAlloc(c->heap, 1, DMA_BUFFER_LENGTH);
}
/* last buffer already linked, but still needs buffer */
dmaBuffs[i].ptr = alHeapAlloc(c->heap, 1, DMA_BUFFER_LENGTH);
for(i=0;i<NUM_ACMD_LISTS;i++)
__am.ACMDList[i] = (Acmd*)alHeapAlloc(c->heap, 1,
amc->maxACMDSize * sizeof(Acmd));
maxRSPCmds = amc->maxACMDSize;
/**** initialize the done messages ****/
for (i = 0; i < NUM_OUTPUT_BUFFERS; i++)
{
__am.audioInfo[i] = (AudioInfo *)alHeapAlloc(c->heap, 1,
sizeof(AudioInfo));
__am.audioInfo[i]->msg.done.type = OS_SC_DONE_MSG;
__am.audioInfo[i]->msg.done.info = __am.audioInfo[i];
__am.audioInfo[i]->data = alHeapAlloc(c->heap, 1, 4*maxFrameSize);
}
osCreateMesgQueue(&__am.audioReplyMsgQ, __am.audioReplyMsgBuf, MAX_MESGS);
osCreateMesgQueue(&__am.audioFrameMsgQ, __am.audioFrameMsgBuf, MAX_MESGS);
osCreateMesgQueue(&audDMAMessageQ, audDMAMessageBuf, NUM_DMA_MESSAGES);
osCreateThread(&__am.thread, 3, __amMain, 0,
(void *)(audioStack+AUDIO_STACKSIZE/sizeof(u64)), pri);
osStartThread(&__am.thread);
}
/******************************************************************************
*
* Audio Manager implementation. This thread wakes up at every retrace,
* and builds an audio task, which it returns to the scheduler, who then
* is responsible for its finally execution on the RSP. Once the task has
* finished execution, the scheduler sends back a message saying the task
* is complete. The audio is triple buffered because the switching to a new
* audio buffer does not occur exactly at the gfx swapbuffer time. With
* 3 buffers you ensure that the program does not destroy data before it is
* played.
*
*****************************************************************************/
static void __amMain(void *arg)
{
u32 validTask;
u32 done = 0;
AudioMsg *msg;
AudioInfo *lastInfo = 0;
OSScClient client;
osScAddClient(&sc, &client, &__am.audioFrameMsgQ);
while (!done)
{
(void) osRecvMesg(&__am.audioFrameMsgQ, (OSMesg *)&msg, OS_MESG_BLOCK);
switch (msg->gen.type)
{
case (OS_SC_RETRACE_MSG):
validTask = __amHandleFrameMsg(__am.audioInfo[audFrameCt % 3],
lastInfo);
if(validTask)
{
/* wait for done message */
osRecvMesg(&__am.audioReplyMsgQ, (OSMesg *)&msg,
OS_MESG_BLOCK);
__amHandleDoneMsg(msg->done.info);
lastInfo = msg->done.info;
}
break;
case (OS_SC_PRE_NMI_MSG):
/* what should we really do here? quit? ramp down volume? */
break;
case (QUIT_MSG):
done = 1;
break;
default:
break;
}
}
alClose(&__am.g);
}
/******************************************************************************
*
* __amHandleFrameMsg. First, clear the past audio dma's, then calculate
* the number of samples you will need for this frame. This value varies
* due to the fact that audio is synchronised off of the video interupt
* which can have a small amount of jitter in it. Varying the number of
* samples slightly will allow you to stay in synch with the video. This
* is an advantageous thing to do, since if you are in synch with the
* video, you will have fewer graphics yields. After you've calculated
* the number of frames needed, call alAudioFrame, which will call all
* of the synthesizer's players (sequence player and sound player) to
* generate the audio task list. If you get a valid task list back, put
* it in a task structure and send a message to the scheduler to let it
* know that the next frame of audio is ready for processing.
*
*****************************************************************************/
static u32 __amHandleFrameMsg(AudioInfo *info, AudioInfo *lastInfo)
{
s16 *audioPtr;
Acmd *cmdp;
s32 cmdLen;
int samplesLeft = 0;
OSScTask *t;
__clearAudioDMA(); /* call once a frame, before doing alAudioFrame */
audioPtr = (s16 *) osVirtualToPhysical(info->data);
if (lastInfo)
osAiSetNextBuffer(lastInfo->data, lastInfo->frameSamples<<2);
/* calculate how many samples needed for this frame to keep the DAC full */
/* this will vary slightly frame to frame, must recalculate every frame */
samplesLeft = osAiGetLength() >> 2; /* divide by four, to convert bytes */
/* to stereo 16 bit samples */
info->frameSamples = 16 + (frameSize - samplesLeft + EXTRA_SAMPLES)& ~0xf;
if(info->frameSamples < minFrameSize)
info->frameSamples = minFrameSize;
cmdp = alAudioFrame(__am.ACMDList[curAcmdList], &cmdLen, audioPtr,
info->frameSamples);
assert(cmdLen <= maxRSPCmds);
if(cmdLen == 0) /* no task produced, return zero to show no valid task */
return 0;
t = &info->task;
t->next = 0; /* paranoia */
t->msgQ = &__am.audioReplyMsgQ; /* reply to when finished */
t->msg = (OSMesg)&info->msg; /* reply with this message */
t->flags = OS_SC_NEEDS_RSP;
t->list.t.data_ptr = (u64 *) __am.ACMDList[curAcmdList];
t->list.t.data_size = (cmdp - __am.ACMDList[curAcmdList]) * sizeof(Acmd);
t->list.t.type = M_AUDTASK;
t->list.t.ucode_boot = (u64 *)rspbootTextStart;
t->list.t.ucode_boot_size =
((int) rspbootTextEnd - (int) rspbootTextStart);
t->list.t.flags = OS_TASK_DP_WAIT;
t->list.t.ucode = (u64 *) aspMainTextStart;
t->list.t.ucode_data = (u64 *) aspMainDataStart;
t->list.t.ucode_data_size = SP_UCODE_DATA_SIZE;
t->list.t.dram_stack = (u64 *) NULL;
t->list.t.dram_stack_size = 0;
t->list.t.output_buff = (u64 *) NULL;
t->list.t.output_buff_size = 0;
t->list.t.yield_data_ptr = NULL;
t->list.t.yield_data_size = 0;
osSendMesg(sched_cmdQ, (OSMesg) t, OS_MESG_BLOCK);
curAcmdList ^= 1; /* swap which acmd list you use each frame */
return 1;
}
/******************************************************************************
*
* __amHandleDoneMsg. Really just debugging info in this frame. Checks
* to make sure we completed before we were out of samples.
*
*****************************************************************************/
static void __amHandleDoneMsg(AudioInfo *info)
{
s32 samplesLeft;
static int firstTime = 1;
samplesLeft = osAiGetLength()>>2;
if (samplesLeft == 0 && !firstTime)
{
#ifndef _FINALROM
PRINTF("audio: ai out of samples\n");
#endif
firstTime = 0;
}
}
/******************************************************************************
*
* __amDMA This routine handles the dma'ing of samples from rom to ram.
* First it checks the current buffers to see if the samples needed are
* already in place. Because buffers are linked sequentially by the
* addresses where the samples are on rom, it doesn't need to check all
* of them, only up to the address that it needs. If it finds one, it
* returns the address of that buffer. If it doesn't find the samples
* that it needs, it will initiate a DMA of the samples that it needs.
* In either case, it updates the lastFrame variable, to indicate that
* this buffer was last used in this frame. This is important for the
* __clearAudioDMA routine.
*
*****************************************************************************/
s32 __amDMA(s32 addr, s32 len, void *state)
{
void *foundBuffer;
s32 delta, addrEnd, buffEnd;
AMDMABuffer *dmaPtr, *lastDmaPtr;
lastDmaPtr = 0;
dmaPtr = dmaState.firstUsed;
addrEnd = addr+len;
/* first check to see if a currently existing buffer contains the
sample that you need. */
while(dmaPtr)
{
buffEnd = dmaPtr->startAddr + DMA_BUFFER_LENGTH;
if(dmaPtr->startAddr > addr) /* since buffers are ordered */
break; /* abort if past possible */
else if(addrEnd <= buffEnd) /* yes, found a buffer with samples */
{
dmaPtr->lastFrame = audFrameCt; /* mark it used */
foundBuffer = dmaPtr->ptr + addr - dmaPtr->startAddr;
return (int) osVirtualToPhysical(foundBuffer);
}
lastDmaPtr = dmaPtr;
dmaPtr = (AMDMABuffer*)dmaPtr->node.next;
}
/* get here, and you didn't find a buffer, so dma a new one */
/* get a buffer from the free list */
dmaPtr = dmaState.firstFree;
/*
* if you get here and dmaPtr is null, send back the a bogus
* pointer, it's better than nothing
*/
if(!dmaPtr)
return osVirtualToPhysical(dmaState.firstUsed);
dmaState.firstFree = (AMDMABuffer*)dmaPtr->node.next;
alUnlink((ALLink*)dmaPtr);
/* add it to the used list */
if(lastDmaPtr) /* if you have other dmabuffers used, add this one */
{ /* to the list, after the last one checked above */
alLink((ALLink*)dmaPtr,(ALLink*)lastDmaPtr);
}
else if(dmaState.firstUsed) /* if this buffer is before any others */
{ /* jam at begining of list */
lastDmaPtr = dmaState.firstUsed;
dmaState.firstUsed = dmaPtr;
dmaPtr->node.next = (ALLink*)lastDmaPtr;
dmaPtr->node.prev = 0;
lastDmaPtr->node.prev = (ALLink*)dmaPtr;
}
else /* no buffers in list, this is the first one */
{
dmaState.firstUsed = dmaPtr;
dmaPtr->node.next = 0;
dmaPtr->node.prev = 0;
}
foundBuffer = dmaPtr->ptr;
delta = addr & 0x1;
addr -= delta;
dmaPtr->startAddr = addr;
dmaPtr->lastFrame = audFrameCt; /* mark it */
audDMAIOMesgBuf[nextDMA].hdr.pri = OS_MESG_PRI_NORMAL;
audDMAIOMesgBuf[nextDMA].hdr.retQueue = &audDMAMessageQ;
audDMAIOMesgBuf[nextDMA].dramAddr = foundBuffer;
audDMAIOMesgBuf[nextDMA].devAddr = (u32)addr;
audDMAIOMesgBuf[nextDMA].size = DMA_BUFFER_LENGTH;
osEPiStartDma(handler, &audDMAIOMesgBuf[nextDMA++], OS_READ);
return (int) osVirtualToPhysical(foundBuffer) + delta;
}
/******************************************************************************
*
* __amDmaNew. Initialize the dma buffers and return the address of the
* procedure that will be used to dma the samples from rom to ram. This
* routine will be called once for each physical voice that is created.
* In this case, because we know where all the buffers are, and since
* they are not attached to a specific voice, we will only really do any
* initialization the first time. After that we just return the address
* to the dma routine.
*
*****************************************************************************/
ALDMAproc __amDmaNew(AMDMAState **state)
{
int i;
if(!dmaState.initialized) /* only do this once */
{
dmaState.firstUsed = 0;
dmaState.firstFree = &dmaBuffs[0];
dmaState.initialized = 1;
}
*state = &dmaState; /* state is never used in this case */
return __amDMA;
}
/******************************************************************************
*
* __clearAudioDMA. Routine to move dma buffers back to the unused list.
* First clear out your dma messageQ. Then check each buffer to see when
* it was last used. If that was more than FRAME_LAG frames ago, move it
* back to the unused list.
*
*****************************************************************************/
static void __clearAudioDMA(void)
{
u32 i;
OSIoMesg *iomsg;
AMDMABuffer *dmaPtr,*nextPtr;
/*
* Don't block here. If dma's aren't complete, you've had an audio
* overrun. (Bad news, but go for it anyway, and try and recover.
*/
for (i=0; i<nextDMA; i++)
{
if (osRecvMesg(&audDMAMessageQ,(OSMesg *)&iomsg,OS_MESG_NOBLOCK) == -1)
#ifndef _FINALROM
PRINTF("Dma not done\n");
#else
;
#endif
#ifndef _FINALROM
if (logging)
osLogEvent(log, 17, 2, iomsg->devAddr, iomsg->size);
#endif
}
dmaPtr = dmaState.firstUsed;
while(dmaPtr)
{
nextPtr = (AMDMABuffer*)dmaPtr->node.next;
/* remove old dma's from list */
/* Can change FRAME_LAG value. Should be at least one. */
/* Larger values mean more buffers needed, but fewer DMA's */
if(dmaPtr->lastFrame + FRAME_LAG < audFrameCt)
{
if(dmaState.firstUsed == dmaPtr)
dmaState.firstUsed = (AMDMABuffer*)dmaPtr->node.next;
alUnlink((ALLink*)dmaPtr);
if(dmaState.firstFree)
alLink((ALLink*)dmaPtr,(ALLink*)dmaState.firstFree);
else
{
dmaState.firstFree = dmaPtr;
dmaPtr->node.next = 0;
dmaPtr->node.prev = 0;
}
}
dmaPtr = nextPtr;
}
nextDMA = 0; /* reset */
audFrameCt++;
}