rockbox/firmware/pcm.c
Michael Sevakis 61d0583384 Greatly reduce volume-change zipper artifacts with SW volume.
Uses a cosine factor to smoothly shift the PCM level from the old level
to the new one over the length of a frame.

Implements indirect calls to PCM scaling function instead of testing
conditions on every callback, cleanly assigning a different call to
do the volume transition. The volume change call then assigns the final
scaling function.

Change-Id: If1004b92a91c5ca766dd0e4014ec274636e8ed26
Reviewed-on: http://gerrit.rockbox.org/763
Reviewed-by: Michael Sevakis <jethead71@rockbox.org>
Tested: Michael Sevakis <jethead71@rockbox.org>
2014-03-08 08:04:41 +01:00

630 lines
15 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_dma_complete_callback
* pcm_play_dma_status_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_dma_complete_callback
* pcm_rec_dma_status_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 */
volatile pcm_play_callback_type
pcm_callback_for_more SHAREDBSS_ATTR = NULL;
/* The registered callback function to inform of DMA status */
volatile pcm_status_callback_type
pcm_play_status_callback 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;
static void pcm_play_data_start_int(const void *addr, size_t size);
static void pcm_play_pause_int(bool play);
void pcm_play_stop_int(void);
#if !defined(HAVE_SW_VOLUME_CONTROL) || defined(PCM_SW_VOLUME_UNBUFFERED)
/** Standard hw volume/unbuffered control functions - otherwise, see
** pcm_sw_volume.c **/
static inline void pcm_play_dma_start_int(const void *addr, size_t size)
{
#ifdef HAVE_SW_VOLUME_CONTROL
/* Smoothed transition might not have happened so sync now */
pcm_sync_pcm_factors();
#endif
pcm_play_dma_start(addr, size);
}
static inline void pcm_play_dma_pause_int(bool pause)
{
if (pause || pcm_get_bytes_waiting() > 0)
{
pcm_play_dma_pause(pause);
}
else
{
logf(" no data");
pcm_play_data_start_int(NULL, 0);
}
}
static inline void pcm_play_dma_stop_int(void)
{
pcm_play_dma_stop();
}
static inline const void * pcm_play_dma_get_peak_buffer_int(int *count)
{
return pcm_play_dma_get_peak_buffer(count);
}
bool pcm_play_dma_complete_callback(enum pcm_dma_status status,
const void **addr, size_t *size)
{
/* Check status callback first if error */
if (status < PCM_DMAST_OK)
status = pcm_play_dma_status_callback(status);
if (status >= PCM_DMAST_OK && pcm_get_more_int(addr, size))
return true;
/* Error, callback missing or no more DMA to do */
pcm_play_stop_int();
return false;
}
#endif /* !HAVE_SW_VOLUME_CONTROL || PCM_SW_VOLUME_UNBUFFERED */
static void pcm_play_data_start_int(const void *addr, size_t size)
{
ALIGN_AUDIOBUF(addr, size);
if ((addr && size) || pcm_get_more_int(&addr, &size))
{
pcm_apply_settings();
logf(" pcm_play_dma_start_int");
pcm_play_dma_start_int(addr, size);
pcm_playing = true;
pcm_paused = false;
}
else
{
/* Force a stop */
logf(" pcm_play_stop_int");
pcm_play_stop_int();
}
}
static void pcm_play_pause_int(bool play)
{
if (play)
pcm_apply_settings();
logf(" pcm_play_dma_pause_int");
pcm_play_dma_pause_int(!play);
pcm_paused = !play && pcm_playing;
}
void pcm_play_stop_int(void)
{
pcm_play_dma_stop_int();
pcm_callback_for_more = NULL;
pcm_play_status_callback = 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 int16_t *p, int count,
struct pcm_peaks *peaks)
{
uint32_t peak_l = 0, peak_r = 0;
const int16_t *pend = p + 2 * count;
do
{
int32_t s;
s = p[0];
if (s < 0)
s = -s;
if ((uint32_t)s > peak_l)
peak_l = s;
s = p[1];
if (s < 0)
s = -s;
if ((uint32_t)s > peak_r)
peak_r = s;
p += 4 * 2; /* Every 4th sample, interleaved */
}
while (p < pend);
peaks->left = peak_l;
peaks->right = 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) / 4;
peaks->tick = tick;
if (active)
{
int framecount = peaks->period*pcm_curr_sampr / HZ;
count = MIN(framecount, count);
if (count > 0)
pcm_peak_peeker(addr, count, peaks);
/* else keep previous peak values */
}
else
{
/* peaks are zero */
peaks->left = peaks->right = 0;
}
}
void pcm_calculate_peaks(int *left, int *right)
{
/* peak data for the global peak values - i.e. what the final output is */
static struct pcm_peaks peaks;
int count;
const void *addr = pcm_play_dma_get_peak_buffer_int(&count);
pcm_do_peak_calculation(&peaks, pcm_playing && !pcm_paused,
addr, count);
if (left)
*left = peaks.left;
if (right)
*right = peaks.right;
}
const void * pcm_get_peak_buffer(int *count)
{
return pcm_play_dma_get_peak_buffer_int(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_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;
}
void pcm_play_data(pcm_play_callback_type get_more,
pcm_status_callback_type status_cb,
const void *start, size_t size)
{
logf("pcm_play_data");
pcm_play_lock();
pcm_callback_for_more = get_more;
pcm_play_status_callback = status_cb;
logf(" pcm_play_data_start_int");
pcm_play_data_start_int(start, size);
pcm_play_unlock();
}
void pcm_play_pause(bool play)
{
logf("pcm_play_pause: %s", play ? "play" : "pause");
pcm_play_lock();
if (play == pcm_paused && pcm_playing)
{
logf(" pcm_play_pause_int");
pcm_play_pause_int(play);
}
pcm_play_unlock();
}
void pcm_play_stop(void)
{
logf("pcm_play_stop");
pcm_play_lock();
if (pcm_playing)
{
logf(" pcm_play_stop_int");
pcm_play_stop_int();
}
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;
}
/* return last-set frequency */
unsigned int pcm_get_frequency(void)
{
return pcm_sampr;
}
/* 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;
}
}
#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;
volatile pcm_status_callback_type
pcm_rec_status_callback 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;
pcm_rec_status_callback = 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 struct pcm_peaks peaks;
if (pcm_recording)
{
const int16_t *peak_addr = pcm_rec_peak_addr;
const int16_t *addr = pcm_rec_dma_get_peak_buffer();
if (addr != NULL)
{
int count = (addr - peak_addr) / 2; /* Interleaved L+R */
if (count > 0)
{
pcm_peak_peeker(peak_addr, count, &peaks);
if (peak_addr == pcm_rec_peak_addr)
pcm_rec_peak_addr = addr;
}
}
/* else keep previous peak values */
}
else
{
peaks.left = peaks.right = 0;
}
if (left)
*left = peaks.left;
if (right)
*right = peaks.right;
}
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,
pcm_status_callback_type status_cb,
void *addr, size_t size)
{
logf("pcm_record_data");
ALIGN_AUDIOBUF(addr, size);
if (!(addr && size))
{
logf(" no buffer");
return;
}
pcm_rec_lock();
pcm_callback_more_ready = more_ready;
pcm_rec_status_callback = status_cb;
/* Need a physical DMA address translation, if not already physical. */
pcm_rec_peak_addr = pcm_rec_dma_addr(addr);
logf(" pcm_rec_dma_start");
pcm_apply_settings();
pcm_rec_dma_start(addr, 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 */
bool pcm_rec_dma_complete_callback(enum pcm_dma_status status,
void **addr, size_t *size)
{
/* Check status callback first if error */
if (status < PCM_DMAST_OK)
status = pcm_rec_dma_status_callback(status);
pcm_rec_callback_type have_more = pcm_callback_more_ready;
if (have_more && status >= PCM_DMAST_OK)
{
/* Call registered callback to obtain next buffer */
have_more(addr, size);
ALIGN_AUDIOBUF(*addr, *size);
if (*addr && *size)
{
/* Need a physical DMA address translation, if not already
* physical. */
pcm_rec_peak_addr = pcm_rec_dma_addr(*addr);
return true;
}
}
/* Error, callback missing or no more DMA to do */
pcm_rec_dma_stop();
pcm_recording_stopped();
return false;
}
#endif /* HAVE_RECORDING */