rockbox/apps/plugins/test_sampr.c
Michael Sevakis e69d567d9e Bring consistency to pcm implementation and samplerate handling. Less low-level duplication. A small test_sampr fix so it works on coldfire again.
git-svn-id: svn://svn.rockbox.org/rockbox/trunk@19400 a1c6a512-1295-4272-9138-f99709370657
2008-12-12 11:01:07 +00:00

342 lines
9.9 KiB
C

/***************************************************************************
* __________ __ ___.
* Open \______ \ ____ ____ | | _\_ |__ _______ ___
* Source | _// _ \_/ ___\| |/ /| __ \ / _ \ \/ /
* Jukebox | | ( <_> ) \___| < | \_\ ( <_> > < <
* Firmware |____|_ /\____/ \___ >__|_ \|___ /\____/__/\_ \
* \/ \/ \/ \/ \/
* $Id$
*
* Copyright (C) 2006 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 "plugin.h"
#include "lib/oldmenuapi.h"
/* This plugin generates a 1kHz tone + noise in order to quickly verify
* hardware samplerate setup is operating correctly.
*
* While switching to different frequencies, the pitch of the tone should
* remain constant whereas the upper harmonics of the noise should vary
* with sample rate.
*/
PLUGIN_HEADER
PLUGIN_IRAM_DECLARE;
const struct plugin_api *rb;
static int hw_freq IDATA_ATTR = HW_FREQ_DEFAULT;
static unsigned long hw_sampr IDATA_ATTR = HW_SAMPR_DEFAULT;
static int gen_thread_stack[DEFAULT_STACK_SIZE/sizeof(int)] IBSS_ATTR;
static bool gen_quit IBSS_ATTR;
static unsigned int gen_thread_id;
#define OUTPUT_CHUNK_COUNT (1 << 1)
#define OUTPUT_CHUNK_MASK (OUTPUT_CHUNK_COUNT-1)
#define OUTPUT_CHUNK_SAMPLES 1152
#define OUTPUT_CHUNK_SIZE (OUTPUT_CHUNK_SAMPLES*sizeof(int16_t)*2)
static uint16_t output_buf[OUTPUT_CHUNK_COUNT][OUTPUT_CHUNK_SAMPLES*2]
__attribute__((aligned(4)));
static int output_head IBSS_ATTR;
static int output_tail IBSS_ATTR;
static int output_step IBSS_ATTR;
static uint32_t gen_phase_step IBSS_ATTR;
static const uint32_t gen_frequency = 1000;
/* fsin shamelessly stolen from signal_gen.c by Thom Johansen (preglow) */
/* Good quality sine calculated by linearly interpolating
* a 128 sample sine table. First harmonic has amplitude of about -84 dB.
* phase has range from 0 to 0xffffffff, representing 0 and
* 2*pi respectively.
* Return value is a signed value from LONG_MIN to LONG_MAX, representing
* -1 and 1 respectively.
*/
static int16_t ICODE_ATTR fsin(uint32_t phase)
{
/* 128 sixteen bit sine samples + guard point */
static const int16_t sinetab[129] ICONST_ATTR =
{
0, 1607, 3211, 4807, 6392, 7961, 9511, 11038,
12539, 14009, 15446, 16845, 18204, 19519, 20787, 22004,
23169, 24278, 25329, 26318, 27244, 28105, 28897, 29621,
30272, 30851, 31356, 31785, 32137, 32412, 32609, 32727,
32767, 32727, 32609, 32412, 32137, 31785, 31356, 30851,
30272, 29621, 28897, 28105, 27244, 26318, 25329, 24278,
23169, 22004, 20787, 19519, 18204, 16845, 15446, 14009,
12539, 11038, 9511, 7961, 6392, 4807, 3211, 1607,
0, -1607, -3211, -4807, -6392, -7961, -9511, -11038,
-12539, -14009, -15446, -16845, -18204, -19519, -20787, -22004,
-23169, -24278, -25329, -26318, -27244, -28105, -28897, -29621,
-30272, -30851, -31356, -31785, -32137, -32412, -32609, -32727,
-32767, -32727, -32609, -32412, -32137, -31785, -31356, -30851,
-30272, -29621, -28897, -28105, -27244, -26318, -25329, -24278,
-23169, -22004, -20787, -19519, -18204, -16845, -15446, -14009,
-12539, -11038, -9511, -7961, -6392, -4807, -3211, -1607,
0,
};
unsigned int pos = phase >> 25;
unsigned short frac = (phase & 0x01ffffff) >> 9;
short diff = sinetab[pos + 1] - sinetab[pos];
return sinetab[pos] + (frac*diff >> 16);
}
/* ISR handler to get next block of data */
static void get_more(unsigned char **start, size_t *size)
{
/* Free previous buffer */
output_head += output_step;
output_step = 0;
*start = (unsigned char *)output_buf[output_head & OUTPUT_CHUNK_MASK];
*size = OUTPUT_CHUNK_SIZE;
/* Keep repeating previous if source runs low */
if (output_head != output_tail)
output_step = 1;
}
static void ICODE_ATTR gen_thread_func(void)
{
uint32_t gen_random = *rb->current_tick;
uint32_t gen_phase = 0;
while (!gen_quit)
{
int16_t *p = output_buf[output_tail & OUTPUT_CHUNK_MASK];
int i = OUTPUT_CHUNK_SAMPLES;
while (output_tail - output_head >= OUTPUT_CHUNK_COUNT)
{
rb->sleep(0);
if (gen_quit)
return;
}
while (--i >= 0)
{
int32_t val = fsin(gen_phase);
int32_t rnd = (int16_t)gen_random;
gen_random = gen_random*0x0019660dL + 0x3c6ef35fL;
val = (rnd + 2*val) / 3;
*p++ = val;
*p++ = val;
gen_phase += gen_phase_step;
}
output_tail++;
rb->yield();
}
}
static void update_gen_step(void)
{
gen_phase_step = 0x100000000ull*gen_frequency / hw_sampr;
}
static void output_clear(void)
{
rb->pcm_play_lock();
rb->memset(output_buf, 0, sizeof (output_buf));
output_head = 0;
output_tail = 0;
rb->pcm_play_unlock();
}
/* Called to switch samplerate on the fly */
static void set_frequency(int index)
{
hw_freq = index;
hw_sampr = rb->hw_freq_sampr[index];
output_clear();
update_gen_step();
rb->pcm_set_frequency(hw_sampr);
rb->pcm_apply_settings();
}
#ifndef HAVE_VOLUME_IN_LIST
static void set_volume(int value)
{
rb->global_settings->volume = value;
rb->sound_set(SOUND_VOLUME, value);
}
static void format_volume(char *buf, size_t len, int value, const char *unit)
{
rb->snprintf(buf, len, "%d %s", rb->sound_val2phys(SOUND_VOLUME, value),
rb->sound_unit(SOUND_VOLUME));
(void)unit;
}
#endif /* HAVE_VOLUME_IN_LIST */
static void play_tone(bool volume_set)
{
static struct opt_items names[HW_NUM_FREQ] =
{
HW_HAVE_96_([HW_FREQ_96] = { "96kHz", -1 },)
HW_HAVE_88_([HW_FREQ_88] = { "88.2kHz", -1 },)
HW_HAVE_64_([HW_FREQ_64] = { "64kHz", -1 },)
HW_HAVE_48_([HW_FREQ_48] = { "48kHz", -1 },)
HW_HAVE_44_([HW_FREQ_44] = { "44.1kHz", -1 },)
HW_HAVE_32_([HW_FREQ_32] = { "32kHz", -1 },)
HW_HAVE_24_([HW_FREQ_24] = { "24kHz", -1 },)
HW_HAVE_22_([HW_FREQ_22] = { "22.05kHz", -1 },)
HW_HAVE_16_([HW_FREQ_16] = { "16kHz", -1 },)
HW_HAVE_12_([HW_FREQ_12] = { "12kHz", -1 },)
HW_HAVE_11_([HW_FREQ_11] = { "11.025kHz", -1 },)
HW_HAVE_8_( [HW_FREQ_8 ] = { "8kHz", -1 },)
};
int freq = hw_freq;
rb->audio_stop();
#if INPUT_SRC_CAPS != 0
/* Select playback */
rb->audio_set_input_source(AUDIO_SRC_PLAYBACK, SRCF_PLAYBACK);
#endif
#ifdef HAVE_ADJUSTABLE_CPU_FREQ
rb->cpu_boost(true);
#endif
rb->pcm_set_frequency(rb->hw_freq_sampr[freq]);
#if INPUT_SRC_CAPS != 0
/* Recordable targets can play back from other sources */
rb->audio_set_output_source(AUDIO_SRC_PLAYBACK);
#endif
gen_quit = false;
output_clear();
update_gen_step();
gen_thread_id = rb->create_thread(gen_thread_func, gen_thread_stack,
sizeof(gen_thread_stack), 0,
"test_sampr generator"
IF_PRIO(, PRIORITY_PLAYBACK)
IF_COP(, CPU));
rb->pcm_play_data(get_more, NULL, 0);
#ifndef HAVE_VOLUME_IN_LIST
if (volume_set)
{
int volume = rb->global_settings->volume;
rb->set_int("Volume", NULL, -1, &volume,
set_volume, 1, rb->sound_min(SOUND_VOLUME),
rb->sound_max(SOUND_VOLUME), format_volume);
}
else
#endif /* HAVE_VOLUME_IN_LIST */
{
rb->set_option("Sample Rate", &freq, INT, names,
HW_NUM_FREQ, set_frequency);
(void)volume_set;
}
gen_quit = true;
rb->thread_wait(gen_thread_id);
rb->pcm_play_stop();
#ifdef HAVE_ADJUSTABLE_CPU_FREQ
rb->cpu_boost(false);
#endif
/* restore default - user of apis is responsible for restoring
default state - normally playback at 44100Hz */
rb->pcm_set_frequency(HW_FREQ_DEFAULT);
}
/* Tests hardware sample rate switching */
/* TODO: needs a volume control */
enum plugin_status plugin_start(const struct plugin_api *api,
const void *parameter)
{
enum
{
__TEST_SAMPR_MENUITEM_FIRST = -1,
#ifndef HAVE_VOLUME_IN_LIST
MENU_VOL_SET,
#endif /* HAVE_VOLUME_IN_LIST */
MENU_SAMPR_SET,
MENU_QUIT,
};
static const struct menu_item items[] =
{
#ifndef HAVE_VOLUME_IN_LIST
[MENU_VOL_SET] =
{ "Set Volume", NULL },
#endif /* HAVE_VOLUME_IN_LIST */
[MENU_SAMPR_SET] =
{ "Set Samplerate", NULL },
[MENU_QUIT] =
{ "Quit", NULL },
};
bool exit = false;
int m;
/* Disable all talking before initializing IRAM */
api->talk_disable(true);
PLUGIN_IRAM_INIT(api);
rb = api;
m = menu_init(rb, items, ARRAYLEN(items),
NULL, NULL, NULL, NULL);
while (!exit)
{
int result = menu_show(m);
switch (result)
{
#ifndef HAVE_VOLUME_IN_LIST
case MENU_VOL_SET:
play_tone(true);
break;
#endif /* HAVE_VOLUME_IN_LIST */
case MENU_SAMPR_SET:
play_tone(false);
break;
case MENU_QUIT:
exit = true;
break;
}
}
menu_exit(m);
rb->talk_disable(false);
return PLUGIN_OK;
(void)parameter;
}