/*************************************************************************** * __________ __ ___. * 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. */ PLUGIN_HEADER PLUGIN_IRAM_DECLARE; 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 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, 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); PLUGIN_IRAM_INIT(rb); 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; }