rockbox/firmware/target/hosted/sdl/pcm-sdl.c
Michael Sevakis 286a4c5caa Revise the PCM callback system after adding multichannel audio.
Additional status callback is added to pcm_play/rec_data instead of
using a special function to set it. Status includes DMA error
reporting to the status callback. Playback and recording callback
become more alike except playback uses "const void **addr" (because
the data should not be altered) and recording  uses "void **addr".
"const" is put in place throughout where appropriate.

Most changes are fairly trivial. One that should be checked in
particular because it isn't so much is telechips, if anyone cares to
bother. PP5002 is not so trivial either but that tested as working.

Change-Id: I4928d69b3b3be7fb93e259f81635232df9bd1df2
Reviewed-on: http://gerrit.rockbox.org/166
Reviewed-by: Michael Sevakis <jethead71@rockbox.org>
Tested-by: Michael Sevakis <jethead71@rockbox.org>
2012-03-03 07:23:38 +01:00

422 lines
9.5 KiB
C

/***************************************************************************
* __________ __ ___.
* Open \______ \ ____ ____ | | _\_ |__ _______ ___
* Source | _// _ \_/ ___\| |/ /| __ \ / _ \ \/ /
* Jukebox | | ( <_> ) \___| < | \_\ ( <_> > < <
* Firmware |____|_ /\____/ \___ >__|_ \|___ /\____/__/\_ \
* \/ \/ \/ \/ \/
* $Id$
*
* Copyright (C) 2005 by Nick Lanham
* Copyright (C) 2010 by Thomas Martitz
*
* This program 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.
*
* This software is distributed on an "AS IS" basis, WITHOUT WARRANTY OF ANY
* KIND, either express or implied.
*
****************************************************************************/
#include "autoconf.h"
#include <stdlib.h>
#include <stdbool.h>
#include <SDL.h>
#include "config.h"
#include "debug.h"
#include "sound.h"
#include "audiohw.h"
#include "system.h"
#include "panic.h"
#ifdef HAVE_RECORDING
#include "audiohw.h"
#ifdef HAVE_SPDIF_IN
#include "spdif.h"
#endif
#endif
#include "pcm.h"
#include "pcm-internal.h"
#include "pcm_sampr.h"
/*#define LOGF_ENABLE*/
#include "logf.h"
#ifdef DEBUG
#include <stdio.h>
extern bool debug_audio;
#endif
static int sim_volume = 0;
#if CONFIG_CODEC == SWCODEC
static int cvt_status = -1;
static const Uint8* pcm_data;
static size_t pcm_data_size;
static size_t pcm_sample_bytes;
static size_t pcm_channel_bytes;
static struct pcm_udata
{
Uint8 *stream;
Uint32 num_in;
Uint32 num_out;
#ifdef DEBUG
FILE *debug;
#endif
} udata;
static SDL_AudioSpec obtained;
static SDL_AudioCVT cvt;
static int audio_locked = 0;
static SDL_mutex *audio_lock;
void pcm_play_lock(void)
{
if (++audio_locked == 1)
SDL_LockMutex(audio_lock);
}
void pcm_play_unlock(void)
{
if (--audio_locked == 0)
SDL_UnlockMutex(audio_lock);
}
static void pcm_dma_apply_settings_nolock(void)
{
cvt_status = SDL_BuildAudioCVT(&cvt, AUDIO_S16SYS, 2, pcm_sampr,
obtained.format, obtained.channels, obtained.freq);
if (cvt_status < 0) {
cvt.len_ratio = (double)obtained.freq / (double)pcm_sampr;
}
}
void pcm_dma_apply_settings(void)
{
pcm_play_lock();
pcm_dma_apply_settings_nolock();
pcm_play_unlock();
}
void pcm_play_dma_start(const void *addr, size_t size)
{
pcm_dma_apply_settings_nolock();
pcm_data = addr;
pcm_data_size = size;
SDL_PauseAudio(0);
}
void pcm_play_dma_stop(void)
{
SDL_PauseAudio(1);
#ifdef DEBUG
if (udata.debug != NULL) {
fclose(udata.debug);
udata.debug = NULL;
DEBUGF("Audio debug file closed\n");
}
#endif
}
void pcm_play_dma_pause(bool pause)
{
if (pause)
SDL_PauseAudio(1);
else
SDL_PauseAudio(0);
}
size_t pcm_get_bytes_waiting(void)
{
return pcm_data_size;
}
static void write_to_soundcard(struct pcm_udata *udata)
{
#ifdef DEBUG
if (debug_audio && (udata->debug == NULL)) {
udata->debug = fopen("audiodebug.raw", "ab");
DEBUGF("Audio debug file open\n");
}
#endif
if (cvt.needed) {
Uint32 rd = udata->num_in;
Uint32 wr = (double)rd * cvt.len_ratio;
if (wr > udata->num_out) {
wr = udata->num_out;
rd = (double)wr / cvt.len_ratio;
if (rd > udata->num_in)
{
rd = udata->num_in;
wr = (double)rd * cvt.len_ratio;
}
}
if (wr == 0 || rd == 0)
{
udata->num_out = udata->num_in = 0;
return;
}
if (cvt_status > 0) {
cvt.len = rd * pcm_sample_bytes;
cvt.buf = (Uint8 *) malloc(cvt.len * cvt.len_mult);
memcpy(cvt.buf, pcm_data, cvt.len);
SDL_ConvertAudio(&cvt);
SDL_MixAudio(udata->stream, cvt.buf, cvt.len_cvt, sim_volume);
udata->num_in = cvt.len / pcm_sample_bytes;
udata->num_out = cvt.len_cvt / pcm_sample_bytes;
#ifdef DEBUG
if (udata->debug != NULL) {
fwrite(cvt.buf, sizeof(Uint8), cvt.len_cvt, udata->debug);
}
#endif
free(cvt.buf);
}
else {
/* Convert is bad, so do silence */
Uint32 num = wr*obtained.channels;
udata->num_in = rd;
udata->num_out = wr;
switch (pcm_channel_bytes)
{
case 1:
{
Uint8 *stream = udata->stream;
while (num-- > 0)
*stream++ = obtained.silence;
break;
}
case 2:
{
Uint16 *stream = (Uint16 *)udata->stream;
while (num-- > 0)
*stream++ = obtained.silence;
break;
}
}
#ifdef DEBUG
if (udata->debug != NULL) {
fwrite(udata->stream, sizeof(Uint8), wr, udata->debug);
}
#endif
}
} else {
udata->num_in = udata->num_out = MIN(udata->num_in, udata->num_out);
SDL_MixAudio(udata->stream, pcm_data,
udata->num_out * pcm_sample_bytes, sim_volume);
#ifdef DEBUG
if (udata->debug != NULL) {
fwrite(pcm_data, sizeof(Uint8), udata->num_out * pcm_sample_bytes,
udata->debug);
}
#endif
}
}
static void sdl_audio_callback(struct pcm_udata *udata, Uint8 *stream, int len)
{
logf("sdl_audio_callback: len %d, pcm %d\n", len, pcm_data_size);
bool new_buffer = false;
udata->stream = stream;
SDL_LockMutex(audio_lock);
/* Write what we have in the PCM buffer */
if (pcm_data_size > 0)
goto start;
/* Audio card wants more? Get some more then. */
while (len > 0) {
new_buffer = pcm_play_dma_complete_callback(PCM_DMAST_OK,
(const void **)&pcm_data, &pcm_data_size);
if (!new_buffer) {
DEBUGF("sdl_audio_callback: No Data.\n");
break;
}
start:
udata->num_in = pcm_data_size / pcm_sample_bytes;
udata->num_out = len / pcm_sample_bytes;
write_to_soundcard(udata);
udata->num_in *= pcm_sample_bytes;
udata->num_out *= pcm_sample_bytes;
if (new_buffer)
{
new_buffer = false;
pcm_play_dma_status_callback(PCM_DMAST_STARTED);
if ((size_t)len > udata->num_out)
{
int delay = pcm_data_size*250 / pcm_sampr - 1;
if (delay > 0)
{
SDL_UnlockMutex(audio_lock);
SDL_Delay(delay);
SDL_LockMutex(audio_lock);
if (!pcm_is_playing())
break;
}
}
}
pcm_data += udata->num_in;
pcm_data_size -= udata->num_in;
udata->stream += udata->num_out;
len -= udata->num_out;
}
SDL_UnlockMutex(audio_lock);
}
const void * pcm_play_dma_get_peak_buffer(int *count)
{
uintptr_t addr = (uintptr_t)pcm_data;
*count = pcm_data_size / 4;
return (void *)((addr + 2) & ~3);
}
#ifdef HAVE_RECORDING
void pcm_rec_lock(void)
{
}
void pcm_rec_unlock(void)
{
}
void pcm_rec_dma_init(void)
{
}
void pcm_rec_dma_close(void)
{
}
void pcm_rec_dma_start(void *start, size_t size)
{
(void)start;
(void)size;
}
void pcm_rec_dma_stop(void)
{
}
const void * pcm_rec_dma_get_peak_buffer(void)
{
return NULL;
}
void audiohw_set_recvol(int left, int right, int type)
{
(void)left;
(void)right;
(void)type;
}
#ifdef HAVE_SPDIF_IN
unsigned long spdif_measure_frequency(void)
{
return 0;
}
#endif
#endif /* HAVE_RECORDING */
void pcm_play_dma_init(void)
{
if (SDL_InitSubSystem(SDL_INIT_AUDIO))
{
DEBUGF("Could not initialize SDL audio subsystem!\n");
return;
}
audio_lock = SDL_CreateMutex();
if (!audio_lock)
{
panicf("Could not create audio_lock\n");
return;
}
SDL_AudioSpec wanted_spec;
#ifdef DEBUG
udata.debug = NULL;
if (debug_audio) {
udata.debug = fopen("audiodebug.raw", "wb");
DEBUGF("Audio debug file open\n");
}
#endif
/* Set 16-bit stereo audio at 44Khz */
wanted_spec.freq = 44100;
wanted_spec.format = AUDIO_S16SYS;
wanted_spec.channels = 2;
wanted_spec.samples = 2048;
wanted_spec.callback =
(void (SDLCALL *)(void *userdata,
Uint8 *stream, int len))sdl_audio_callback;
wanted_spec.userdata = &udata;
/* Open the audio device and start playing sound! */
if(SDL_OpenAudio(&wanted_spec, &obtained) < 0) {
DEBUGF("Unable to open audio: %s\n", SDL_GetError());
return;
}
switch (obtained.format)
{
case AUDIO_U8:
case AUDIO_S8:
pcm_channel_bytes = 1;
break;
case AUDIO_U16LSB:
case AUDIO_S16LSB:
case AUDIO_U16MSB:
case AUDIO_S16MSB:
pcm_channel_bytes = 2;
break;
default:
DEBUGF("Unknown sample format obtained: %u\n",
(unsigned)obtained.format);
return;
}
pcm_sample_bytes = obtained.channels * pcm_channel_bytes;
pcm_dma_apply_settings_nolock();
}
void pcm_play_dma_postinit(void)
{
}
void pcm_set_mixer_volume(int volume)
{
sim_volume = volume;
}
#endif /* CONFIG_CODEC == SWCODEC */