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rockbox/apps/plugins/test_sampr.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

324 lines
9.5 KiB
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/***************************************************************************
* __________ __ ___.
* 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"
/* 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.
*/
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(const void **start, size_t *size)
{
/* Free previous buffer */
output_head += output_step;
output_step = 0;
*start = 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 const char *format_volume(char *buf, size_t len, int value,
const char *unit)
{
(void)unit;
rb->snprintf(buf, len, "%d %s", rb->sound_val2phys(SOUND_VOLUME, value),
rb->sound_unit(SOUND_VOLUME));
return buf;
}
#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, 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 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,
};
MENUITEM_STRINGLIST(menu, "Test Sampr Menu", NULL,
#ifndef HAVE_VOLUME_IN_LIST
"Set Volume",
#endif /* HAVE_VOLUME_IN_LIST */
"Set Samplerate", "Quit");
bool exit = false;
int selected = 0;
/* Disable all talking before initializing IRAM */
rb->talk_disable(true);
while (!exit)
{
int result = rb->do_menu(&menu, &selected, NULL, false);
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;
}
}
rb->talk_disable(false);
return PLUGIN_OK;
(void)parameter;
}
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