Nyaaori/rockbox
1
0
Fork 0
Code Issues Pull requests Projects Releases Packages Wiki Activity
rockbox/firmware/pcm_playback.c
Dave Bryant 27c83252e2 The peak meter on iRiver didn't look right to me and the code seemed to be 
making it a lot more complicated that it needed to be, so I threw in some new
code. This should be easy to adjust for different CPU loads, and I set it up
to use a little less than what was there. If this causes any trouble, please
roll it back.


git-svn-id: svn://svn.rockbox.org/rockbox/trunk@7238 a1c6a512-1295-4272-9138-f99709370657
2005-07-25 05:50:47 +00:00

324 lines
8.6 KiB
C
Raw Blame History

/***************************************************************************
* __________ __ ___.
* Open \______ \ ____ ____ | | _\_ |__ _______ ___
* Source | _// _ \_/ ___\| |/ /| __ \ / _ \ \/ /
* Jukebox | | ( <_> ) \___| < | \_\ ( <_> > < <
* Firmware |____|_ /\____/ \___ >__|_ \|___ /\____/__/\_ \
* \/ \/ \/ \/ \/
* $Id$
*
* Copyright (C) 2005 by Linus Nielsen Feltzing
*
* All files in this archive are subject to the GNU General Public License.
* See the file COPYING in the source tree root for full license agreement.
*
* This software is distributed on an "AS IS" basis, WITHOUT WARRANTY OF ANY
* KIND, either express or implied.
*
****************************************************************************/
#include <stdbool.h>
#include "config.h"
#include "debug.h"
#include "panic.h"
#include <kernel.h>
#ifndef SIMULATOR
#include "cpu.h"
#include "i2c.h"
#include "uda1380.h"
#include "system.h"
#endif
#include "logf.h"
#include <stdio.h>
#include <string.h>
#include <stdarg.h>
#include "pcm_playback.h"
#include "lcd.h"
#include "button.h"
#include "file.h"
#include "buffer.h"
#include "sprintf.h"
#include "button.h"
#include <string.h>
#ifdef HAVE_UDA1380
static bool pcm_playing;
static bool pcm_paused;
static int pcm_freq = 0x6; /* 44.1 is default */
static unsigned char *next_start;
static long next_size;
/* Set up the DMA transfer that kicks in when the audio FIFO gets empty */
static void dma_start(const void *addr, long size)
{
pcm_playing = true;
addr = (void *)((unsigned long)addr & ~3); /* Align data */
size &= ~3; /* Size must be multiple of 4 */
/* Reset the audio FIFO */
IIS2CONFIG = 0x800;
EBU1CONFIG = 0x800;
/* Set up DMA transfer */
SAR0 = ((unsigned long)addr); /* Source address */
DAR0 = (unsigned long)&PDOR3; /* Destination address */
BCR0 = size; /* Bytes to transfer */
/* Enable the FIFO and force one write to it */
IIS2CONFIG = (pcm_freq << 12) | 0x300 | 4 << 2;
/* Also send the audio to S/PDIF */
EBU1CONFIG = (7 << 12) | (3 << 8) | (1 << 5) | (5 << 2);
DCR0 = DMA_INT | DMA_EEXT | DMA_CS | DMA_SINC | DMA_START;
}
/* Stops the DMA transfer and interrupt */
static void dma_stop(void)
{
pcm_playing = false;
DCR0 = 0;
/* Reset the FIFO */
IIS2CONFIG = 0x800;
EBU1CONFIG = 0x800;
next_start = NULL;
next_size = 0;
pcm_paused = false;
}
/*
* This function goes directly into the DMA buffer to calculate the left and
* right peak values. To avoid missing peaks it tries to look forward a full
* refresh period (1/20 sec) although it's always possible that the entire
* period will not be visible. To reduce CPU load it only looks at every
* third sample, and this can be reduced even further if needed (even every
* tenth sample would still be pretty accurate).
*/
#define PEAK_SAMPLES 2205 /* 44100 sample rate / 20 Hz refresh */
#define PEAK_STRIDE 3 /* every 3rd sample is plenty... */
void pcm_calculate_peaks(int *left, int *right)
{
long samples = (BCR0 & 0xffffff) / 4;
short *addr = (short *) (SAR0 & ~3);
if (samples > PEAK_SAMPLES)
samples = PEAK_SAMPLES;
samples /= PEAK_STRIDE;
if (left && right) {
int left_peak = 0, right_peak = 0, value;
while (samples--) {
if ((value = addr [0]) > left_peak)
left_peak = value;
else if (-value > left_peak)
left_peak = -value;
if ((value = addr [PEAK_STRIDE | 1]) > right_peak)
right_peak = value;
else if (-value > right_peak)
right_peak = -value;
addr += PEAK_STRIDE * 2;
}
*left = left_peak;
*right = right_peak;
}
else if (left || right) {
int peak_value = 0, value;
if (right)
addr += (PEAK_STRIDE | 1);
while (samples--) {
if ((value = addr [0]) > peak_value)
peak_value = value;
else if (-value > peak_value)
peak_value = -value;
addr += PEAK_STRIDE * 2;
}
if (left)
*left = peak_value;
else
*right = peak_value;
}
}
/* sets frequency of input to DAC */
void pcm_set_frequency(unsigned int frequency)
{
switch(frequency)
{
case 11025:
pcm_freq = 0x4;
uda1380_set_nsorder(3);
break;
case 22050:
pcm_freq = 0x6;
uda1380_set_nsorder(3);
break;
case 44100:
default:
pcm_freq = 0xC;
uda1380_set_nsorder(5);
break;
}
}
/* the registered callback function to ask for more mp3 data */
static void (*callback_for_more)(unsigned char**, long*) = NULL;
void pcm_play_data(void (*get_more)(unsigned char** start, long* size))
{
unsigned char *start;
long size;
callback_for_more = get_more;
/** FIXME: This is a temporary fix to prevent playback glitches when
* playing the first file. We will just drop the first frame to prevent
* that problem from occurring.
* Some debug data:
* - This problem will occur only when the first file.
* - First frame will be totally corrupt and the song will begin
* from the next frame. But at the next time (when the bug has
* already happened), the song will start from first frame.
* - Dropping some frames directly from (mpa) codec will also
* prevent the problem from happening. So it's unlikely you can
* find the explanation for this bug from this file.
*/
get_more((unsigned char **)&start, (long *)&size); // REMOVE THIS TO TEST
get_more((unsigned char **)&start, (long *)&size);
get_more(&next_start, &next_size);
dma_start(start, size);
uda1380_mute(false);
}
long pcm_get_bytes_waiting(void)
{
return next_size + (BCR0 & 0xffffff);
}
void pcm_play_stop(void)
{
if (pcm_playing) {
uda1380_mute(true);
dma_stop();
}
}
void pcm_play_pause(bool play)
{
if(pcm_paused && play && next_size)
{
logf("unpause");
/* Reset chunk size so dma has enough data to fill the fifo. */
/* This shouldn't be needed anymore. */
//SAR0 = (unsigned long)next_start;
//BCR0 = next_size;
/* Enable the FIFO and force one write to it */
IIS2CONFIG = (pcm_freq << 12) | 0x300 | 4 << 2;
EBU1CONFIG = (7 << 12) | (3 << 8) | (1 << 5) | (5 << 2);
DCR0 |= DMA_EEXT | DMA_START;
uda1380_mute(false);
}
else if(!pcm_paused && !play)
{
logf("pause");
uda1380_mute(true);
/* Disable DMA peripheral request. */
DCR0 &= ~DMA_EEXT;
IIS2CONFIG = 0x800;
EBU1CONFIG = 0x800;
}
pcm_paused = !play;
}
bool pcm_is_paused(void)
{
return pcm_paused;
}
bool pcm_is_playing(void)
{
return pcm_playing;
}
/* DMA0 Interrupt is called when the DMA has finished transfering a chunk */
void DMA0(void) __attribute__ ((interrupt_handler, section(".icode")));
void DMA0(void)
{
int res = DSR0;
DSR0 = 1; /* Clear interrupt */
DCR0 &= ~DMA_EEXT;
/* Stop on error */
if(res & 0x70)
{
pcm_play_stop();
logf("DMA Error:0x%04x", res);
}
else
{
if(next_size)
{
SAR0 = (unsigned long)next_start; /* Source address */
BCR0 = next_size; /* Bytes to transfer */
DCR0 |= DMA_EEXT;
if (callback_for_more)
callback_for_more(&next_start, &next_size);
}
else
{
/* Finished playing */
pcm_play_stop();
logf("DMA No Data:0x%04x", res);
}
}
IPR |= (1<<14); /* Clear pending interrupt request */
}
void pcm_init(void)
{
pcm_playing = false;
pcm_paused = false;
uda1380_init();
BUSMASTER_CTRL = 0x81; /* PARK[1,0]=10 + BCR24BIT */
DIVR0 = 54; /* DMA0 is mapped into vector 54 in system.c */
DMAROUTE = (DMAROUTE & 0xffffff00) | DMA0_REQ_AUDIO_1;
DMACONFIG = 1; /* DMA0Req = PDOR3 */
/* Reset the audio FIFO */
IIS2CONFIG = 0x800;
/* Enable interrupt at level 7, priority 0 */
ICR4 = (ICR4 & 0xffff00ff) | 0x00001c00;
IMR &= ~(1<<14); /* bit 14 is DMA0 */
pcm_set_frequency(44100);
/* Turn on headphone power with audio output muted. */
uda1380_mute(true);
sleep(HZ/4);
uda1380_enable_output(true);
/* Call dma_stop to initialize everything. */
dma_stop();
}
#endif /* HAVE_UDA1380 */
Reference in a new issue View git blame Copy permalink