rockbox/firmware/pcm.c
Michael Sevakis e42a3194de AS3525v1/v2:
Fix problems with volume of recorded material by converting 14-bit samples to
16-bit. Remove duplicate samples from recorded data and support proper
samplerate since ADC runs 1/2 the codec clock. Support monitoring mono on both
output channels by feeding data manually to I2SOUT under the right conditions.

DMA is no longer used for recording since frames must be processed as described
above but it does allow full-duplex audio.

Miscellaneous change includes a proper constant (HW_SAMPR_DEFAULT) to reset the
hardware samplerate when recording is closed. PP5024 and AS3525 have different
default recording rates (22kHz and 44kHz respectively) but both have half-speed
ADC.


git-svn-id: svn://svn.rockbox.org/rockbox/trunk@31180 a1c6a512-1295-4272-9138-f99709370657
2011-12-08 19:20:00 +00:00

621 lines
14 KiB
C

/***************************************************************************
* __________ __ ___.
* Open \______ \ ____ ____ | | _\_ |__ _______ ___
* Source | _// _ \_/ ___\| |/ /| __ \ / _ \ \/ /
* Jukebox | | ( <_> ) \___| < | \_\ ( <_> > < <
* Firmware |____|_ /\____/ \___ >__|_ \|___ /\____/__/\_ \
* \/ \/ \/ \/ \/
* $Id$
*
* Copyright (C) 2007 by Michael Sevakis
*
* 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 <stdlib.h>
#include "system.h"
#include "kernel.h"
/* Define LOGF_ENABLE to enable logf output in this file */
/*#define LOGF_ENABLE*/
#include "logf.h"
#include "audio.h"
#include "sound.h"
#include "general.h"
#include "pcm-internal.h"
#include "pcm_mixer.h"
/**
* Aspects implemented in the target-specific portion:
*
* ==Playback==
* Public -
* pcm_postinit
* pcm_get_bytes_waiting
* pcm_play_lock
* pcm_play_unlock
* Semi-private -
* pcm_play_get_more_callback
* pcm_play_dma_init
* pcm_play_dma_postinit
* pcm_play_dma_start
* pcm_play_dma_stop
* pcm_play_dma_pause
* pcm_play_dma_get_peak_buffer
* Data Read/Written within TSP -
* pcm_sampr (R)
* pcm_fsel (R)
* pcm_curr_sampr (R)
* pcm_playing (R)
* pcm_paused (R)
*
* ==Playback/Recording==
* Public -
* pcm_dma_addr
* Semi-private -
* pcm_dma_apply_settings
*
* ==Recording==
* Public -
* pcm_rec_lock
* pcm_rec_unlock
* Semi-private -
* pcm_rec_more_ready_callback
* pcm_rec_dma_init
* pcm_rec_dma_close
* pcm_rec_dma_start
* pcm_rec_dma_stop
* pcm_rec_dma_get_peak_buffer
* Data Read/Written within TSP -
* pcm_recording (R)
*
* States are set _after_ the target's pcm driver is called so that it may
* know from whence the state is changed. One exception is init.
*
*/
/* 'true' when all stages of pcm initialization have completed */
static bool pcm_is_ready = false;
/* the registered callback function to ask for more mp3 data */
static pcm_play_callback_type pcm_callback_for_more SHAREDBSS_ATTR = NULL;
void (* pcm_play_dma_started)(void) SHAREDBSS_ATTR = NULL;
/* PCM playback state */
volatile bool pcm_playing SHAREDBSS_ATTR = false;
/* PCM paused state. paused implies playing */
volatile bool pcm_paused SHAREDBSS_ATTR = false;
/* samplerate of currently playing audio - undefined if stopped */
unsigned long pcm_curr_sampr SHAREDBSS_ATTR = 0;
/* samplerate waiting to be set */
unsigned long pcm_sampr SHAREDBSS_ATTR = HW_SAMPR_DEFAULT;
/* samplerate frequency selection index */
int pcm_fsel SHAREDBSS_ATTR = HW_FREQ_DEFAULT;
/* peak data for the global peak values - i.e. what the final output is */
static struct pcm_peaks global_peaks;
/* Called internally by functions to reset the state */
static void pcm_play_stopped(void)
{
pcm_callback_for_more = NULL;
pcm_play_dma_started = NULL;
pcm_paused = false;
pcm_playing = false;
}
static void pcm_wait_for_init(void)
{
while (!pcm_is_ready)
sleep(0);
}
/**
* Perform peak calculation on a buffer of packed 16-bit samples.
*
* Used for recording and playback.
*/
static void pcm_peak_peeker(const int32_t *addr, int count, uint16_t peaks[2])
{
int peak_l = 0, peak_r = 0;
const int32_t * const end = addr + count;
do
{
int32_t value = *addr;
int ch;
#ifdef ROCKBOX_BIG_ENDIAN
ch = value >> 16;
#else
ch = (int16_t)value;
#endif
if (ch < 0)
ch = -ch;
if (ch > peak_l)
peak_l = ch;
#ifdef ROCKBOX_BIG_ENDIAN
ch = (int16_t)value;
#else
ch = value >> 16;
#endif
if (ch < 0)
ch = -ch;
if (ch > peak_r)
peak_r = ch;
addr += 4;
}
while (addr < end);
peaks[0] = peak_l;
peaks[1] = peak_r;
}
void pcm_do_peak_calculation(struct pcm_peaks *peaks, bool active,
const void *addr, int count)
{
long tick = current_tick;
/* Peak no farther ahead than expected period to avoid overcalculation */
long period = tick - peaks->tick;
/* Keep reasonable limits on period */
if (period < 1)
period = 1;
else if (period > HZ/5)
period = HZ/5;
peaks->period = (3*peaks->period + period) >> 2;
peaks->tick = tick;
if (active)
{
int framecount = peaks->period*pcm_curr_sampr / HZ;
count = MIN(framecount, count);
if (count > 0)
pcm_peak_peeker((int32_t *)addr, count, peaks->val);
/* else keep previous peak values */
}
else
{
/* peaks are zero */
peaks->val[0] = peaks->val[1] = 0;
}
}
void pcm_calculate_peaks(int *left, int *right)
{
int count;
const void *addr = pcm_play_dma_get_peak_buffer(&count);
pcm_do_peak_calculation(&global_peaks, pcm_playing && !pcm_paused,
addr, count);
if (left)
*left = global_peaks.val[0];
if (right)
*right = global_peaks.val[1];
}
const void* pcm_get_peak_buffer(int * count)
{
return pcm_play_dma_get_peak_buffer(count);
}
bool pcm_is_playing(void)
{
return pcm_playing;
}
bool pcm_is_paused(void)
{
return pcm_paused;
}
/****************************************************************************
* Functions that do not require targeted implementation but only a targeted
* interface
*/
/* This should only be called at startup before any audio playback or
recording is attempted */
void pcm_init(void)
{
logf("pcm_init");
pcm_play_stopped();
pcm_set_frequency(HW_SAMPR_DEFAULT);
logf(" pcm_play_dma_init");
pcm_play_dma_init();
}
/* Finish delayed init */
void pcm_postinit(void)
{
logf("pcm_postinit");
logf(" pcm_play_dma_postinit");
pcm_play_dma_postinit();
pcm_is_ready = true;
}
bool pcm_is_initialized(void)
{
return pcm_is_ready;
}
/* Common code to pcm_play_data and pcm_play_pause */
static void pcm_play_data_start(unsigned char *start, size_t size)
{
ALIGN_AUDIOBUF(start, size);
if (!(start && size))
{
pcm_play_callback_type get_more = pcm_callback_for_more;
size = 0;
if (get_more)
{
logf(" get_more");
get_more(&start, &size);
ALIGN_AUDIOBUF(start, size);
}
}
if (start && size)
{
logf(" pcm_play_dma_start");
pcm_apply_settings();
pcm_play_dma_start(start, size);
pcm_playing = true;
pcm_paused = false;
return;
}
/* Force a stop */
logf(" pcm_play_dma_stop");
pcm_play_dma_stop();
pcm_play_stopped();
}
void pcm_play_data(pcm_play_callback_type get_more,
unsigned char *start, size_t size)
{
logf("pcm_play_data");
pcm_play_lock();
pcm_callback_for_more = get_more;
logf(" pcm_play_data_start");
pcm_play_data_start(start, size);
pcm_play_unlock();
}
void pcm_play_get_more_callback(void **start, size_t *size)
{
pcm_play_callback_type get_more = pcm_callback_for_more;
*size = 0;
if (get_more && start)
{
/* Call registered callback */
get_more((unsigned char **)start, size);
ALIGN_AUDIOBUF(*start, *size);
if (*start && *size)
return;
}
/* Error, callback missing or no more DMA to do */
pcm_play_dma_stop();
pcm_play_stopped();
}
void pcm_play_pause(bool play)
{
logf("pcm_play_pause: %s", play ? "play" : "pause");
pcm_play_lock();
if (play == pcm_paused && pcm_playing)
{
if (!play)
{
logf(" pcm_play_dma_pause");
pcm_play_dma_pause(true);
pcm_paused = true;
}
else if (pcm_get_bytes_waiting() > 0)
{
logf(" pcm_play_dma_pause");
pcm_apply_settings();
pcm_play_dma_pause(false);
pcm_paused = false;
}
else
{
logf(" pcm_play_dma_start: no data");
pcm_play_data_start(NULL, 0);
}
}
else
{
logf(" no change");
}
pcm_play_unlock();
}
void pcm_play_stop(void)
{
logf("pcm_play_stop");
pcm_play_lock();
if (pcm_playing)
{
logf(" pcm_play_dma_stop");
pcm_play_dma_stop();
pcm_play_stopped();
}
else
{
logf(" not playing");
}
pcm_play_unlock();
}
/**/
/* set frequency next frequency used by the audio hardware -
* what pcm_apply_settings will set */
void pcm_set_frequency(unsigned int samplerate)
{
logf("pcm_set_frequency");
int index;
#ifdef CONFIG_SAMPR_TYPES
unsigned int type = samplerate & SAMPR_TYPE_MASK;
samplerate &= ~SAMPR_TYPE_MASK;
/* For now, supported targets have direct conversion when configured with
* CONFIG_SAMPR_TYPES.
* Some hypothetical target with independent rates would need slightly
* different handling throughout this source. */
samplerate = pcm_sampr_to_hw_sampr(samplerate, type);
#endif /* CONFIG_SAMPR_TYPES */
index = round_value_to_list32(samplerate, hw_freq_sampr,
HW_NUM_FREQ, false);
if (samplerate != hw_freq_sampr[index])
index = HW_FREQ_DEFAULT; /* Invalid = default */
pcm_sampr = hw_freq_sampr[index];
pcm_fsel = index;
}
/* apply pcm settings to the hardware */
void pcm_apply_settings(void)
{
logf("pcm_apply_settings");
pcm_wait_for_init();
if (pcm_sampr != pcm_curr_sampr)
{
logf(" pcm_dma_apply_settings");
pcm_dma_apply_settings();
pcm_curr_sampr = pcm_sampr;
}
}
/* register callback to buffer more data */
void pcm_play_set_dma_started_callback(void (* callback)(void))
{
pcm_play_dma_started = callback;
}
#ifdef HAVE_RECORDING
/** Low level pcm recording apis **/
/* Next start for recording peaks */
static const void * volatile pcm_rec_peak_addr SHAREDBSS_ATTR = NULL;
/* the registered callback function for when more data is available */
static volatile pcm_rec_callback_type
pcm_callback_more_ready SHAREDBSS_ATTR = NULL;
/* DMA transfer in is currently active */
volatile bool pcm_recording SHAREDBSS_ATTR = false;
/* Called internally by functions to reset the state */
static void pcm_recording_stopped(void)
{
pcm_recording = false;
pcm_callback_more_ready = NULL;
}
/**
* Return recording peaks - From the end of the last peak up to
* current write position.
*/
void pcm_calculate_rec_peaks(int *left, int *right)
{
static uint16_t peaks[2];
if (pcm_recording)
{
const void *peak_addr = pcm_rec_peak_addr;
const void *addr = pcm_rec_dma_get_peak_buffer();
if (addr != NULL)
{
int count = (int)(((intptr_t)addr - (intptr_t)peak_addr) >> 2);
if (count > 0)
{
pcm_peak_peeker((int32_t *)peak_addr, count, peaks);
if (peak_addr == pcm_rec_peak_addr)
pcm_rec_peak_addr = addr;
}
}
/* else keep previous peak values */
}
else
{
peaks[0] = peaks[1] = 0;
}
if (left)
*left = peaks[0];
if (right)
*right = peaks[1];
} /* pcm_calculate_rec_peaks */
bool pcm_is_recording(void)
{
return pcm_recording;
}
/****************************************************************************
* Functions that do not require targeted implementation but only a targeted
* interface
*/
void pcm_init_recording(void)
{
logf("pcm_init_recording");
pcm_wait_for_init();
/* Stop the beasty before attempting recording */
mixer_reset();
/* Recording init is locked unlike general pcm init since this is not
* just a one-time event at startup and it should and must be safe by
* now. */
pcm_rec_lock();
logf(" pcm_rec_dma_init");
pcm_recording_stopped();
pcm_rec_dma_init();
pcm_rec_unlock();
}
void pcm_close_recording(void)
{
logf("pcm_close_recording");
pcm_rec_lock();
if (pcm_recording)
{
logf(" pcm_rec_dma_stop");
pcm_rec_dma_stop();
pcm_recording_stopped();
}
logf(" pcm_rec_dma_close");
pcm_rec_dma_close();
pcm_rec_unlock();
}
void pcm_record_data(pcm_rec_callback_type more_ready,
void *start, size_t size)
{
logf("pcm_record_data");
ALIGN_AUDIOBUF(start, size);
if (!(start && size))
{
logf(" no buffer");
return;
}
pcm_rec_lock();
pcm_callback_more_ready = more_ready;
#ifdef HAVE_PCM_REC_DMA_ADDRESS
/* Need a physical DMA address translation, if not already physical. */
pcm_rec_peak_addr = pcm_dma_addr(start);
#else
pcm_rec_peak_addr = start;
#endif
logf(" pcm_rec_dma_start");
pcm_apply_settings();
pcm_rec_dma_start(start, size);
pcm_recording = true;
pcm_rec_unlock();
} /* pcm_record_data */
void pcm_stop_recording(void)
{
logf("pcm_stop_recording");
pcm_rec_lock();
if (pcm_recording)
{
logf(" pcm_rec_dma_stop");
pcm_rec_dma_stop();
pcm_recording_stopped();
}
pcm_rec_unlock();
} /* pcm_stop_recording */
void pcm_rec_more_ready_callback(int status, void **start, size_t *size)
{
pcm_rec_callback_type have_more = pcm_callback_more_ready;
*size = 0;
if (have_more && start)
{
have_more(status, start, size);
ALIGN_AUDIOBUF(*start, *size);
if (*start && *size)
{
#ifdef HAVE_PCM_REC_DMA_ADDRESS
/* Need a physical DMA address translation, if not already
* physical. */
pcm_rec_peak_addr = pcm_dma_addr(*start);
#else
pcm_rec_peak_addr = *start;
#endif
return;
}
}
/* Error, callback missing or no more DMA to do */
pcm_rec_dma_stop();
pcm_recording_stopped();
}
#endif /* HAVE_RECORDING */