diff --git a/apps/lang/bulgarian.lang b/apps/lang/bulgarian.lang index 76b058e6f9..06f75d50a4 100644 --- a/apps/lang/bulgarian.lang +++ b/apps/lang/bulgarian.lang @@ -12507,6 +12507,20 @@ *: "Средна побитова скорост" + + id: LANG_PLAYTIME_ERROR + desc: playing time screen + user: core + + *: "Error while gathering info" + + + *: "Грешка при събирането на информация" + + + *: "Грешка при събирането на информация" + + id: LANG_PLAYING_TIME desc: onplay menu @@ -14672,7 +14686,7 @@ user: core *: "Press LEFT to cancel." - android,hifietma*: "Press BACK to cancel." + android,hifietma*,zenvision: "Press BACK to cancel." cowond2,creativezenxfi2,ibassodx50,ibassodx90,mrobe500,ondavx747: "Press POWER to cancel." ihifi760,ihifi960: "Double tap RETURN to cancel." ihifi770,ihifi770c,ihifi800: "Press HOME to cancel." @@ -14687,7 +14701,7 @@ *: "Натиснете LEFT за отказ." - android,hifietma*: "Натиснете BACK за отказ." + android,hifietma*,zenvision: "Натиснете BACK за отказ." cowond2,creativezenxfi2,ibassodx50,ibassodx90,mrobe500,ondavx747: "Натиснете POWER за отказ." ihifi760,ihifi960: "Натиснете двукратно RETURN за отказ." ihifi770,ihifi770c,ihifi800: "Натиснете HOME за отказ." @@ -14702,7 +14716,7 @@ *: "Натиснете LEFT за отказ." - android,hifietma*: "Натиснете BACK за отказ." + android,hifietma*,zenvision: "Натиснете BACK за отказ." cowond2,creativezenxfi2,ibassodx50,ibassodx90,mrobe500,ondavx747: "Натиснете POWER за отказ." ihifi760,ihifi960: "Натиснете двукратно RETURN за отказ." ihifi770,ihifi770c,ihifi800: "Натиснете HOME за отказ." @@ -15197,13 +15211,13 @@ id: LANG_VOICED_DATE_FORMAT - desc: format string for how dates will be read back. Y == 4-digit year (grouped), y == 4-digit year (numeric), A == month name, m == numeric month, d == numeric day. For example, for 2021-01-05, "AdY" will be voiced as "January 5 twenty twenty-one" and "dmy" will be voiced as "5 1 two thousand twenty one + desc: format string for how dates will be read back. Y == 4-digit year, A == month name, m == numeric month, d == numeric day. For example, "AdY" will read "January 21 2021" user: core *: "dAY" - *: "dAy" + *: "~dAY" *: "" @@ -15818,6 +15832,20 @@ *: "Браузър по подразбиране" + + id: LANG_AMAZE_MENU + desc: Amaze game + user: core + + *: "Amaze Main Menu" + + + *: "Главно меню на Amaze" + + + *: "Главно меню на Amaze" + + id: LANG_SET_MAZE_SIZE desc: Maze size in Amaze game @@ -16070,6 +16098,34 @@ *: "Настройки на Mik mod" + + id: LANG_MIKMOD_MENU + desc: mikmod plugin + user: core + + *: "Mikmod Menu" + + + *: "Меню на Mikmod" + + + *: "Меню на Mik mod" + + + + id: LANG_CHESSBOX_MENU + desc: chessbox plugin + user: core + + *: "Chessbox Menu" + + + *: "Меню на Chessbox" + + + *: "Меню на Chess box" + + id: VOICE_INVALID_VOICE_FILE desc: played if the voice file fails to load @@ -16931,48 +16987,3 @@ usbdac: "USB ЦАП е активен" - - id: LANG_ANNOUNCE_STATUS - desc: announnnce_status plugin - user: core - - *: "Announce Status" - - - *: "Съобщаване на състоянието" - - - *: "Съобщаване на състоянието" - - - - id: LANG_USE_LED_INDICATORS - desc: LED indicators setting - user: core - - *: none - general_purpose_led: "Use LED indicators" - - - *: none - general_purpose_led: "Използване на светодиодни индикатори" - - - *: none - general_purpose_led: "Използване на светодиодни индикатори" - - - - id: LANG_KEEP_DIRECTORY - desc: file browser setting - user: core - - *: "Always remember last folder" - - - *: "Винаги помни последната папка" - - - *: "Винаги помни последната папка" - - diff --git a/apps/lang/deutsch.lang b/apps/lang/deutsch.lang index b006e5d3c4..8a8f5ae5a1 100644 --- a/apps/lang/deutsch.lang +++ b/apps/lang/deutsch.lang @@ -14690,7 +14690,7 @@ user: core *: "Press LEFT to cancel." - android,hifietma*: "Press BACK to cancel." + android,hifietma*,zenvision: "Press BACK to cancel." cowond2,creativezenxfi2,ibassodx50,ibassodx90,mrobe500,ondavx747: "Press POWER to cancel." ihifi760,ihifi960: "Double tap RETURN to cancel." ihifi770,ihifi770c,ihifi800: "Press HOME to cancel." @@ -14705,7 +14705,7 @@ *: "Zum Abbrechen LINKS drücken." - android,hifietma*,vibe500: "Zum Abbrechen ZURÜCK drücken." + android,hifietma*,vibe500,zenvision: "Zum Abbrechen ZURÜCK drücken." cowond2,creativezenxfi2,ibassodx50,ibassodx90,mrobe500,ondavx747: "Zum Abbrechen POWER drücken." ihifi760,ihifi960: "Zum Abbrechen ZURÜCK doppeldrücken." ihifi770,ihifi770c,ihifi800: "Zum Abbrechen STARTMENÜ drücken." @@ -14719,7 +14719,7 @@ *: "Zum Abbrechen LINKS drücken." - android,hifietma*,vibe500: "Zum Abbrechen ZURÜCK drücken." + android,hifietma*,vibe500,zenvision: "Zum Abbrechen ZURÜCK drücken." cowond2,creativezenxfi2,ibassodx50,ibassodx90,mrobe500,ondavx747: "Zum Abbrechen POWER drücken." ihifi760,ihifi960: "Zum Abbrechen ZURÜCK doppeldrücken." ihifi770,ihifi770c,ihifi800: "Zum Abbrechen STARTMENÜ drücken." @@ -15213,7 +15213,7 @@ id: LANG_VOICED_DATE_FORMAT - desc: format string for how dates will be read back. Y == 4-digit year (grouped), y == 4-digit year (numeric), A == month name, m == numeric month, d == numeric day. For example, for 2021-01-05, "AdY" will be voiced as "January 5 twenty twenty-one" and "dmy" will be voiced as "5 1 two thousand twenty one + desc: format string for how dates will be read back. Y == 4-digit year, A == month name, m == numeric month, d == numeric day. For example, "AdY" will read "January 21 2021" user: core *: "dAY" @@ -16978,17 +16978,3 @@ *: "Status mitteilen" - - id: LANG_KEEP_DIRECTORY - desc: file browser setting - user: core - - *: "Always remember last folder" - - - *: "Zuletzt verwendeten Ordner immer merken" - - - *: "Zuletzt verwendeten Ordner immer merken" - - diff --git a/apps/plugin.c b/apps/plugin.c index 2c0e2e778d..d15f88c376 100644 --- a/apps/plugin.c +++ b/apps/plugin.c @@ -644,6 +644,7 @@ static const struct plugin_api rockbox_api = { #endif &audio_master_sampr_list[0], &hw_freq_sampr[0], + pcm_apply_settings, pcm_play_lock, pcm_play_unlock, pcm_current_sink_caps, diff --git a/apps/plugin.h b/apps/plugin.h index 2dcbdbe39f..69cf7659c9 100644 --- a/apps/plugin.h +++ b/apps/plugin.h @@ -179,7 +179,7 @@ int plugin_open(const char *plugin, const char *parameter); * when this happens please take the opportunity to sort in * any new functions "waiting" at the end of the list. */ -#define PLUGIN_API_VERSION 280 +#define PLUGIN_API_VERSION 279 /* 239 Marks the removal of ARCHOS HWCODEC and CHARCELL */ @@ -743,6 +743,7 @@ struct plugin_api { #endif const unsigned long *audio_master_sampr_list; const unsigned long *hw_freq_sampr; + void (*pcm_apply_settings)(void); void (*pcm_play_lock)(void); void (*pcm_play_unlock)(void); const struct pcm_sink_caps* (*pcm_current_sink_caps)(void); diff --git a/apps/plugins/doom/i_sound.c b/apps/plugins/doom/i_sound.c index 3581b0e301..68fb8bdc6c 100644 --- a/apps/plugins/doom/i_sound.c +++ b/apps/plugins/doom/i_sound.c @@ -462,7 +462,7 @@ void get_more(const void** start, size_t* size) I_UpdateSound(); // Force sound update *start = mixbuffer; - *size = MIXBUFFERSIZE*sizeof(short); + *size = SAMPLECOUNT*2*sizeof(short); } diff --git a/apps/plugins/lua/include_lua/pcm.lua b/apps/plugins/lua/include_lua/pcm.lua index 9652c5e3a7..23f8eb308e 100644 --- a/apps/plugins/lua/include_lua/pcm.lua +++ b/apps/plugins/lua/include_lua/pcm.lua @@ -24,5 +24,6 @@ -- [[ conversion to old style pcm_ functions ]] if not rb.pcm then rb.splash(rb.HZ, "No Support!") return nil end +rb.pcm_apply_settings = function() rb.pcm("apply_settings") end rb.pcm_play_lock = function() rb.pcm("play_lock") end rb.pcm_play_unlock = function() rb.pcm("play_unlock") end diff --git a/apps/plugins/lua/rocklib.c b/apps/plugins/lua/rocklib.c index 4d77afedc0..0ee104e7fd 100644 --- a/apps/plugins/lua/rocklib.c +++ b/apps/plugins/lua/rocklib.c @@ -549,10 +549,10 @@ RB_WRAP(sound) RB_WRAP(pcm) { - enum e_pcm {PCM_PLAYLOCK = 0, PCM_PLAYUNLOCK, + enum e_pcm {PCM_APPLYSETTINGS = 0, PCM_PLAYLOCK, PCM_PLAYUNLOCK, PCM_ECOUNT}; - const char *pcm_option[] = {"play_lock", "play_unlock", + const char *pcm_option[] = {"apply_settings", "play_lock", "play_unlock", NULL}; lua_pushnil(L); /*push nil so options w/o return have something to return */ @@ -560,6 +560,9 @@ RB_WRAP(pcm) int option = luaL_checkoption (L, 1, NULL, pcm_option); switch(option) { + case PCM_APPLYSETTINGS: + rb->pcm_apply_settings(); + break; case PCM_PLAYLOCK: rb->pcm_play_lock(); break; diff --git a/apps/plugins/pdbox/PDa/src/s_audio_rockbox.c b/apps/plugins/pdbox/PDa/src/s_audio_rockbox.c index 5f82f6d393..17ef90b1d4 100644 --- a/apps/plugins/pdbox/PDa/src/s_audio_rockbox.c +++ b/apps/plugins/pdbox/PDa/src/s_audio_rockbox.c @@ -64,6 +64,7 @@ void rockbox_open_audio(int rate) /* Set sample rate of the audio buffer. */ rb->mixer_set_frequency(rate); + rb->pcm_apply_settings(); /* Initialize output buffer. */ for(i = 0; i < OUTBUFSIZE; i++) @@ -85,6 +86,7 @@ void rockbox_close_audio(void) /* Restore default sampling rate. */ rb->mixer_set_frequency(HW_SAMPR_DEFAULT); + rb->pcm_apply_settings(); } /* Rockbox audio callback. */ diff --git a/apps/plugins/test_sampr.c b/apps/plugins/test_sampr.c index c73e131072..10571a2d98 100644 --- a/apps/plugins/test_sampr.c +++ b/apps/plugins/test_sampr.c @@ -168,6 +168,7 @@ static void set_frequency(int index) update_gen_step(); rb->mixer_set_frequency(hw_sampr); + rb->pcm_apply_settings(); } #ifndef HAVE_VOLUME_IN_LIST diff --git a/apps/plugins/xrick/system/syssnd_rockbox.c b/apps/plugins/xrick/system/syssnd_rockbox.c index afd0766863..dd522cd6f7 100644 --- a/apps/plugins/xrick/system/syssnd_rockbox.c +++ b/apps/plugins/xrick/system/syssnd_rockbox.c @@ -253,6 +253,7 @@ bool syssnd_init(void) #endif rb->mixer_set_frequency(HW_FREQ_44); + rb->pcm_apply_settings(); rb->memset(channels, 0, sizeof(channels)); rb->memset(mixBuffers, 0, sizeof(mixBuffers)); @@ -285,6 +286,7 @@ void syssnd_shutdown(void) /* Restore default sampling rate. */ rb->mixer_set_frequency(HW_SAMPR_DEFAULT); + rb->pcm_apply_settings(); rb->talk_disable(false); diff --git a/apps/recorder/pcm_record.c b/apps/recorder/pcm_record.c index 0d5581563f..01f0e604dd 100644 --- a/apps/recorder/pcm_record.c +++ b/apps/recorder/pcm_record.c @@ -616,10 +616,6 @@ static void init_state(void) record_status = RECORD_STOPPED; } -/* To avoid having to pull in all of pcm-internal.h */ -void pcm_set_frequency(unsigned int samplerate); -void pcm_apply_settings(void); - /* Set hardware samplerate and save it */ static void update_samplerate_config(unsigned long sampr) { @@ -880,7 +876,7 @@ copy_buffer_mono_lr(void *dst, const void *src, size_t src_size) int16_t *d = (int16_t*) dst; int16_t const *s = (int16_t const*) src; ssize_t copy_size = src_size; - + /* mono = (L + R) / 2 */ while(copy_size > 0) { *d++ = ((int32_t)s[0] + (int32_t)s[1] + 1) >> 1; diff --git a/docs/PLUGIN_API b/docs/PLUGIN_API index 65bf9d4c5b..8fdc10f9eb 100644 --- a/docs/PLUGIN_API +++ b/docs/PLUGIN_API @@ -1754,6 +1754,10 @@ void pcmbuf_set_low_latency(bool state) \param state \description +void pcm_apply_settings(void) + \group sound + \description + void pcm_calculate_rec_peaks(int *left, int *right) \group sound \conditions (defined(HAVE_RECORDING)) diff --git a/firmware/SOURCES b/firmware/SOURCES index 73804c24e9..8ff0c20a9c 100644 --- a/firmware/SOURCES +++ b/firmware/SOURCES @@ -1942,6 +1942,7 @@ target/arm/stm32/debug-stm32h7.c #endif target/arm/stm32/gpio-stm32h7.c target/arm/stm32/i2c-stm32h7.c +target/arm/stm32/pcm-stm32h7.c target/arm/stm32/sdmmc-stm32h7.c target/arm/stm32/spi-stm32h7.c target/arm/stm32/system-stm32h7.c @@ -1958,9 +1959,6 @@ target/arm/stm32/echoplayer/button-echoplayer.c target/arm/stm32/echoplayer/clock-echoplayer.c target/arm/stm32/echoplayer/i2c-echoplayer.c target/arm/stm32/echoplayer/lcd-echoplayer.c -#ifndef BOOTLOADER -target/arm/stm32/echoplayer/pcm-echoplayer.c -#endif target/arm/stm32/echoplayer/power-echoplayer.c target/arm/stm32/echoplayer/sdmmc-echoplayer.c target/arm/stm32/echoplayer/system-echoplayer.c diff --git a/firmware/export/config/echor1.h b/firmware/export/config/echor1.h index dd8ad30f66..7727f03da7 100644 --- a/firmware/export/config/echor1.h +++ b/firmware/export/config/echor1.h @@ -51,14 +51,9 @@ /* Codec / audio hardware defines */ #define HW_SAMPR_CAPS SAMPR_CAP_ALL_96 // FIXME: check this section #define HAVE_ECHOPLAYER_CODEC -#define HAVE_AIC310X #define HAVE_SW_TONE_CONTROLS #define HAVE_SW_VOLUME_CONTROL -#ifndef SIMULATOR -#define PCM_NATIVE_BITDEPTH 32 -#endif - /* Button defines */ #define CONFIG_KEYPAD ECHO_R1_PAD #define HAVE_HEADPHONE_DETECTION diff --git a/firmware/export/echoplayer_codec.h b/firmware/export/echoplayer_codec.h index 4f3d1e0f49..3be3e9c015 100644 --- a/firmware/export/echoplayer_codec.h +++ b/firmware/export/echoplayer_codec.h @@ -21,13 +21,9 @@ #ifndef __ECHOPLAYER_CODEC_H__ #define __ECHOPLAYER_CODEC_H__ -#include - /* -79 to 0 dB in 0.5 dB steps; software volume control * is used because the hardware volume controls "click" * when changing the volume */ AUDIOHW_SETTING(VOLUME, "dB", 1, 5, -790, 0, -200); -void audiohw_mute(bool mute); - #endif /* __ECHOPLAYER_CODEC_H__ */ diff --git a/firmware/export/pcm-internal.h b/firmware/export/pcm-internal.h index 4b4890d35a..4e9cb58c91 100644 --- a/firmware/export/pcm-internal.h +++ b/firmware/export/pcm-internal.h @@ -71,20 +71,6 @@ void pcm_sync_pcm_factors(void); ({ (start) = (void *)(((uintptr_t)(start) + 3) & ~3); \ (size) &= ~3; }) -/* Internal PCM API calls for playback */ -void pcm_play_data(pcm_play_callback_type get_more, - pcm_status_callback_type status_cb, - const void *start, size_t size); - -void pcm_play_stop(void); -void pcm_play_stop_int(void); /* requires PCM lock held */ -bool pcm_is_playing(void); - -void pcm_set_frequency(unsigned int samplerate); -unsigned int pcm_get_frequency(void); -/* apply settings to hardware immediately */ -void pcm_apply_settings(void); - void pcm_do_peak_calculation(struct pcm_peaks *peaks, bool active, const void *addr, int count); @@ -139,6 +125,7 @@ pcm_play_dma_status_callback(enum pcm_dma_status status) #if defined(HAVE_SW_VOLUME_CONTROL) && !defined(PCM_SW_VOLUME_UNBUFFERED) void pcm_play_dma_start_int(const void *addr, size_t size); void pcm_play_dma_stop_int(void); +void pcm_play_stop_int(void); #endif /* HAVE_SW_VOLUME_CONTROL && !PCM_SW_VOLUME_UNBUFFERED */ /* Called by the bottom layer ISR when more data is needed. Returns true diff --git a/firmware/export/pcm.h b/firmware/export/pcm.h index 5f1509ad97..8c5170ac0e 100644 --- a/firmware/export/pcm.h +++ b/firmware/export/pcm.h @@ -44,7 +44,7 @@ typedef void (*pcm_play_callback_type)(const void **start, size_t *size); /* Typedef for registered status callback */ typedef enum pcm_dma_status (*pcm_status_callback_type)(enum pcm_dma_status status); -/* set the pcm frequency - use values in hw_sampr_list +/* set the pcm frequency - use values in hw_sampr_list * when CONFIG_SAMPR_TYPES is #defined, or-in SAMPR_TYPE_* fields with * frequency value. SAMPR_TYPE_PLAY is 0 and the default if none is * specified. */ @@ -54,6 +54,15 @@ unsigned int pcm_sampr_type_rec_to_play(unsigned int samplerate); #endif #endif /* CONFIG_SAMPR_TYPES */ +/* set next frequency to be used */ +void pcm_set_frequency(unsigned int samplerate); +/* return last-set frequency */ +unsigned int pcm_get_frequency(void); +/* apply settings to hardware immediately */ +void pcm_apply_settings(void); + +/** RAW PCM playback routines **/ + /* Reenterable locks for locking and unlocking the playback interrupt */ void pcm_play_lock(void); void pcm_play_unlock(void); @@ -68,6 +77,11 @@ const struct pcm_sink_caps* pcm_sink_caps(enum pcm_sink_ids sink); /* shortcut for plugins */ const struct pcm_sink_caps* pcm_current_sink_caps(void); +/* This is for playing "raw" PCM data */ +void pcm_play_data(pcm_play_callback_type get_more, + pcm_status_callback_type status_cb, + const void *start, size_t size); + /* Kept internally for global PCM and used by mixer's verion of peak calculation */ struct pcm_peaks @@ -78,6 +92,9 @@ struct pcm_peaks long tick; /* Last tick called */ }; +void pcm_play_stop(void); +bool pcm_is_playing(void); + #ifdef HAVE_RECORDING /** RAW PCM recording routines **/ diff --git a/firmware/pcm.c b/firmware/pcm.c index 0cee855fcc..c19ed8e070 100644 --- a/firmware/pcm.c +++ b/firmware/pcm.c @@ -94,6 +94,8 @@ volatile pcm_status_callback_type /* PCM playback state */ volatile bool pcm_playing SHAREDBSS_ATTR = false; +void pcm_play_stop_int(void); + struct pcm_sink* pcm_get_current_sink(void) { return sinks[cur_sink]; diff --git a/firmware/pcm_mixer.c b/firmware/pcm_mixer.c index d36560f149..ca48e3ba10 100644 --- a/firmware/pcm_mixer.c +++ b/firmware/pcm_mixer.c @@ -475,9 +475,6 @@ void mixer_set_frequency(unsigned int samplerate) mix_frame_size = 1; mix_frame_size *= MIX_FRAME_SAMPLES * 4; - - if (pcm_is_initialized()) - pcm_apply_settings(); } /* Get output samplerate */ diff --git a/firmware/target/arm/stm32/debug-stm32h7.c b/firmware/target/arm/stm32/debug-stm32h7.c index f8cd7200c3..99a3849b41 100644 --- a/firmware/target/arm/stm32/debug-stm32h7.c +++ b/firmware/target/arm/stm32/debug-stm32h7.c @@ -86,68 +86,11 @@ static bool swd_menu(void) return simplelist_show_list(&info); } -#if defined(ECHO_R1) -extern volatile int pcm_sai_xrun_count; -extern volatile int pcm_dma_teif_count; -extern volatile int pcm_dma_dmeif_count; -extern volatile int pcm_dma_feif_count; - -struct pcmdebug_counter -{ - const char *name; - volatile int *value; -}; - -static const struct pcmdebug_counter pcmdebug_counters[] = -{ - {"SAI xrun count", &pcm_sai_xrun_count}, - {"DMA TEIF count", &pcm_dma_teif_count}, - {"DMA DMEIF count", &pcm_dma_dmeif_count}, - {"DMA FEIF count", &pcm_dma_feif_count}, -}; - -static int pcmdebug_menu_action_cb(int action, struct gui_synclist *lists) -{ - if (action == ACTION_STD_OK) - { - int item = gui_synclist_get_sel_pos(lists); - const struct pcmdebug_counter *counter = &pcmdebug_counters[item]; - - *counter->value = 0; - action = ACTION_REDRAW; - } - - if (action == ACTION_NONE) - action = ACTION_REDRAW; - - return action; -} - -static const char *pcmdebug_menu_get_name(int item, void *data, char *buf, size_t bufsz) -{ - const struct pcmdebug_counter *counter = &pcmdebug_counters[item]; - (void)data; - - snprintf(buf, bufsz, "%s: %d", counter->name, *counter->value); - return buf; -} - -static bool pcm_debug_menu(void) -{ - struct simplelist_info info; - simplelist_info_init(&info, "PCM debug", ARRAYLEN(pcmdebug_counters), NULL); - info.action_callback = pcmdebug_menu_action_cb; - info.get_name = pcmdebug_menu_get_name; - return simplelist_show_list(&info); -} -#endif - /* Menu definition */ static const struct { const char *name; bool (*function) (void); } menuitems[] = { - {"PCM debug", pcm_debug_menu}, {"SWD/JTAG", swd_menu}, }; diff --git a/firmware/target/arm/stm32/echoplayer/audiohw-echoplayer.c b/firmware/target/arm/stm32/echoplayer/audiohw-echoplayer.c index 2fb913481e..cf336ee41e 100644 --- a/firmware/target/arm/stm32/echoplayer/audiohw-echoplayer.c +++ b/firmware/target/arm/stm32/echoplayer/audiohw-echoplayer.c @@ -19,151 +19,21 @@ * ****************************************************************************/ #include "audiohw.h" -#include "kernel.h" -#include "panic.h" -#include "aic310x.h" -#include "pcm_sw_volume.h" -#include "power-echoplayer.h" -#include "gpio-stm32h7.h" -#include "i2c-echoplayer.h" - -static int tlv_read_multiple(uint8_t reg, uint8_t *data, size_t count) -{ - return stm32_i2c_read_mem(&i2c1_ctl, AIC310X_I2C_ADDR, reg, data, count); -} - -static int tlv_write_multiple(uint8_t reg, const uint8_t *data, size_t count) -{ - return stm32_i2c_write_mem(&i2c1_ctl, AIC310X_I2C_ADDR, reg, data, count); -} - -static struct aic310x tlv_codec = { - .read_multiple = tlv_read_multiple, - .write_multiple = tlv_write_multiple, -}; - -struct codec_reg_data -{ - uint8_t reg; - uint8_t val; -}; - -static const struct codec_reg_data codec_init_seq[] = { - /* Set output driver configuration to stereo psuedo-differential */ - { .reg = AIC310X_HEADSET_DETECT_B, .val = 0x08 }, - - /* Clock input is CODEC_CLKIN (bypass codec PLL) */ - { .reg = AIC310X_CLOCK, .val = 0x01 }, - - /* Route left ch. -> left DAC, right ch -> right DAC */ - { .reg = AIC310X_DATA_PATH, .val = 0x0A }, - - /* Power on DACs, set HPLCOM = VCM, set HPRCOM = VCM, - * enable short circuit protection */ - { .reg = AIC310X_HPOUT_DRIVER_CTRL, .val = 0x0C }, - { .reg = AIC310X_DAC_POWER_CTRL, .val = 0xD0 }, - - /* Unmute the DACs */ - { .reg = AIC310X_LEFT_DAC_DIGITAL_VOLUME, .val = 0x00 }, - { .reg = AIC310X_RIGHT_DAC_DIGITAL_VOLUME, .val = 0x00 }, - - /* Route DACs to output drivers */ - { .reg = AIC310X_DAC_L1_TO_HPLOUT_VOLUME, .val = 0x80 }, - { .reg = AIC310X_DAC_R1_TO_HPROUT_VOLUME, .val = 0x80 }, - - /* Route DACs to line out port */ - { .reg = AIC310X_DAC_L1_TO_LEFT_LO_VOLUME, .val = 0x80 }, - { .reg = AIC310X_DAC_R1_TO_RIGHT_LO_VOLUME, .val = 0x80 }, - - /* Power on headphone output drivers */ - { .reg = AIC310X_HPLCOM_LEVEL_CONTROL, .val = 0x01 }, - { .reg = AIC310X_HPRCOM_LEVEL_CONTROL, .val = 0x01 }, - { .reg = AIC310X_HPLOUT_LEVEL_CONTROL, .val = 0x01 }, - { .reg = AIC310X_HPROUT_LEVEL_CONTROL, .val = 0x01 }, - - /* Power up line out drivers */ - { .reg = AIC310X_RIGHT_LO_LEVEL_CONTROL, .val = 0x01 }, - { .reg = AIC310X_LEFT_LO_LEVEL_CONTROL, .val = 0x01 }, -}; - -static void write_aic310x_seq(const struct codec_reg_data *seq, size_t count) -{ - while (count > 0) - { - if (aic310x_write(&tlv_codec, seq->reg, seq->val)) - panicf("aic310x wr fail: %02x", (unsigned int)seq->reg); - - seq++; - count--; - } -} - -static void wait_aic310x_active(void) -{ - long end_tick = current_tick + HZ; - while (TIME_BEFORE(current_tick, end_tick)) - { - uint8_t tmp; - int err = tlv_read_multiple(AIC310X_PAGE_SELECT, &tmp, 1); - if (err == 0) - return; - - sleep(1); - } - - panicf("aic310x init timeout"); -} void audiohw_init(void) { - /* initialize driver */ - aic310x_init(&tlv_codec); - - /* bring 1.8v regulator up */ - echoplayer_enable_1v8_regulator(true); - - /* apply AVDD/DRVDD and DVDD */ - gpio_set_level(GPIO_CODEC_AVDD_EN, 0); - gpio_set_level(GPIO_CODEC_DVDD_EN, 0); - - /* delay for power stabilization */ - sleep(HZ/100); - - /* bring codec out of reset */ - gpio_set_level(GPIO_CODEC_RESET, 1); } void audiohw_postinit(void) { - wait_aic310x_active(); - write_aic310x_seq(codec_init_seq, ARRAYLEN(codec_init_seq)); } void audiohw_close(void) { - /* apply reset */ - gpio_set_level(GPIO_CODEC_RESET, 0); - - /* remove power */ - gpio_set_level(GPIO_CODEC_AVDD_EN, 1); - gpio_set_level(GPIO_CODEC_DVDD_EN, 1); - - /* disable regulator */ - echoplayer_enable_1v8_regulator(false); -} - -void audiohw_mute(bool mute) -{ - uint8_t wr_val = (mute ? 0x01 : 0x09); - - /* Mute and unmute at the output drivers */ - aic310x_write(&tlv_codec, AIC310X_HPLOUT_LEVEL_CONTROL, wr_val); - aic310x_write(&tlv_codec, AIC310X_HPROUT_LEVEL_CONTROL, wr_val); - aic310x_write(&tlv_codec, AIC310X_LEFT_LO_LEVEL_CONTROL, wr_val); - aic310x_write(&tlv_codec, AIC310X_RIGHT_LO_LEVEL_CONTROL, wr_val); } void audiohw_set_volume(int vol_l, int vol_r) { - pcm_set_master_volume(vol_l, vol_r); + (void)vol_l; + (void)vol_r; } diff --git a/firmware/target/arm/stm32/echoplayer/clock-echoplayer.c b/firmware/target/arm/stm32/echoplayer/clock-echoplayer.c index 9f1b158902..f83314a2e6 100644 --- a/firmware/target/arm/stm32/echoplayer/clock-echoplayer.c +++ b/firmware/target/arm/stm32/echoplayer/clock-echoplayer.c @@ -52,7 +52,6 @@ INIT_ATTR static void init_pll(void) /* * Use HSE/4 input for PLL1 * Use HSE/16 input for PLL3 - * PLL2 reserved for audio; configured in target PCM code */ reg_writef(RCC_PLLCKSELR, PLLSRC_V(HSE), @@ -165,7 +164,7 @@ INIT_ATTR static void init_lse(void) INIT_ATTR static void init_periph_clock(void) { reg_writef(RCC_D1CCIPR, SDMMCSEL_V(PLL1Q)); - reg_writef(RCC_D2CCIP1R, SAI1SEL_V(PLL2P), SPI45SEL_V(HSE)); + reg_writef(RCC_D2CCIP1R, SPI45SEL_V(HSE)); reg_writef(RCC_D2CCIP2R, I2C123SEL_V(HSI)); /* Enable AXI SRAM in sleep mode to allow DMA'ing out of it */ diff --git a/firmware/target/arm/stm32/echoplayer/pcm-echoplayer.c b/firmware/target/arm/stm32/echoplayer/pcm-echoplayer.c deleted file mode 100644 index c1d457ef8d..0000000000 --- a/firmware/target/arm/stm32/echoplayer/pcm-echoplayer.c +++ /dev/null @@ -1,381 +0,0 @@ -/*************************************************************************** - * __________ __ ___. - * Open \______ \ ____ ____ | | _\_ |__ _______ ___ - * Source | _// _ \_/ ___\| |/ /| __ \ / _ \ \/ / - * Jukebox | | ( <_> ) \___| < | \_\ ( <_> > < < - * Firmware |____|_ /\____/ \___ >__|_ \|___ /\____/__/\_ \ - * \/ \/ \/ \/ \/ - * $Id$ - * - * Copyright (C) 2026 Aidan MacDonald - * - * 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 "pcm.h" -#include "pcm_sampr.h" -#include "pcm-internal.h" -#include "audiohw.h" -#include "panic.h" -#include "nvic-arm.h" -#include "dmamux-stm32h743.h" -#include "regs/stm32h743/rcc.h" -#include "regs/stm32h743/sai.h" -#include "regs/stm32h743/dma.h" -#include "regs/stm32h743/dmamux.h" -#include - -/* Configurable bit depth */ -#if PCM_NATIVE_BITDEPTH == 32 -# define PCM_NATIVE_SAMPLE_SIZE 4 -# define DMA_PSIZE_VAL BV_DMA_CHN_CR_PSIZE_32BIT -# define SAI_DSIZE_VAL BV_SAI_SUBBLOCK_CR1_DS_32BIT -# define SAI_FIFO_THRESH BV_SAI_SUBBLOCK_CR2_FTH_QUARTER -# define SAI_FRL_VAL 64 -#elif PCM_NATIVE_BITDEPTH == 24 -# define PCM_NATIVE_SAMPLE_SIZE 4 -# define DMA_PSIZE_VAL BV_DMA_CHN_CR_PSIZE_32BIT -# define SAI_DSIZE_VAL BV_SAI_SUBBLOCK_CR1_DS_24BIT -# define SAI_FIFO_THRESH BV_SAI_SUBBLOCK_CR2_FTH_QUARTER -# define SAI_FRL_VAL 64 -#elif PCM_NATIVE_BITDEPTH == 16 -# define PCM_NATIVE_SAMPLE_SIZE 2 -# define DMA_PSIZE_VAL BV_DMA_CHN_CR_PSIZE_16BIT -# define SAI_DSIZE_VAL BV_SAI_SUBBLOCK_CR1_DS_16BIT -# define SAI_FIFO_THRESH BV_SAI_SUBBLOCK_CR2_FTH_HALF -# define SAI_FRL_VAL 32 -#else -# error "Unsupported PCM bit depth" -#endif - -static const uintptr_t sai1 = ITA_SAI1; -static const uintptr_t sai1a = sai1 + ITO_SAI_A; -static const uintptr_t dmamux1 = ITA_DMAMUX1; -static const uintptr_t dma1 = ITA_DMA1; -static const uintptr_t dma1_ch0 = dma1 + ITO_DMA_CHN(0); - -static atomic_size_t play_lock; -static volatile int play_active; - -static int pcm_last_freq = -1; - -volatile int pcm_sai_xrun_count; -volatile int pcm_dma_teif_count; -volatile int pcm_dma_dmeif_count; -volatile int pcm_dma_feif_count; - -static void play_dma_start(const void *addr, size_t size) -{ - commit_dcache_range(addr, size); - - /* - * The number of transfers is defined by PSIZE, which may - * be 16-bit or 32-bit depending on PCM native sample size. - * Each FIFO write transfers 1 sample, so we want to set - * NDT = size / PCM_NATIVE_SAMPLE_SIZE. - * - * When MSIZE == 32-bit and PSIZE == 16-bit, the DMA will - * take each 32-bit word from memory and write the lower/upper - * halfwords separately to the FIFO. The number of reads from - * memory in this case is NDT/2, so we're still transferring - * the correct amount of data. - */ - reg_varl(dma1_ch0, DMA_CHN_NDTR) = size / PCM_NATIVE_SAMPLE_SIZE; - reg_varl(dma1_ch0, DMA_CHN_M0AR) = (uintptr_t)addr; - reg_writelf(dma1_ch0, DMA_CHN_CR, EN(1)); - - pcm_play_dma_status_callback(PCM_DMAST_STARTED); -} - -static void sai_init(void) -{ - /* Enable SAI1 and DMA1 */ - reg_writef(RCC_APB2ENR, SAI1EN(1)); - reg_writef(RCC_AHB1ENR, DMA1EN(1)); - - /* Link DMA1_CH0 to SAI1A */ - reg_writelf(dmamux1, DMAMUX_CR(0), DMAREQ_ID(DMAMUX1_REQ_SAI1A_DMA)); - - /* Configure DMA1 channel 0 */ - reg_writelf(dma1_ch0, DMA_CHN_CR, - MBURST_V(INCR4), /* 32-bit x 4 burst = 16 bytes */ - PBURST_V(INCR4), /* (16|32)-bit x 4 burst = 8-16 bytes */ - TRBUFF(0), /* bufferable mode not used for SAI */ - DBM(0), /* double buffer mode off */ - PL_V(VERYHIGH), /* highest priority */ - MSIZE_V(32BIT), /* read 32-bit words from memory */ - PSIZE(DMA_PSIZE_VAL), /* set according to PCM bit depth */ - MINC(1), /* increment memory address */ - PINC(0), /* don't increment peripheral address */ - CIRC(0), /* circular mode off */ - DIR_V(MEM_TO_PERIPH), /* read from memory, write to SAI */ - PFCTRL_V(DMA), /* DMA is flow controller */ - TCIE(1), /* transfer complete interrupt */ - TEIE(1), /* transfer error interrupt */ - DMEIE(1)); /* direct mode error interrupt */ - reg_writelf(dma1_ch0, DMA_CHN_FCR, - FEIE(1), /* fifo error interrupt */ - DMDIS(1), /* enable fifo mode */ - FTH_V(FULL)); /* fifo threshold = 4 words */ - - /* Set peripheral address here since it's constant */ - reg_varl(dma1_ch0, DMA_CHN_PAR) = - (uintptr_t)reg_ptrl(sai1a, SAI_SUBBLOCK_DR); - - /* Configure SAI for playback */ - reg_writelf(sai1a, SAI_SUBBLOCK_CR1, - OSR_V(256FS), /* MCLK is 256 x Fs */ - NOMCK(0), /* MCLK enabled */ - DMAEN(0), /* DMA request disabled */ - SAIEN(0), /* SAI disabled */ - OUTDRIV(0), /* drive outputs when SAIEN=1 */ - MONO(0), /* stereo mode */ - SYNCEN_V(ASYNC), /* no sync with other SAIs */ - CKSTR_V(TX_FALLING_RX_RISING), /* clock edge for sampling */ - LSBFIRST(0), /* transmit samples MSB first */ - DS(SAI_DSIZE_VAL), /* sample size according to bit depth */ - PRTCFG_V(FREE), /* free protocol */ - MODE_V(MASTER_TX)); /* operate as bus master */ - reg_writelf(sai1a, SAI_SUBBLOCK_CR2, - MUTEVAL_V(ZERO_SAMPLE), /* send zero sample on mute */ - MUTE(1), /* mute output initially */ - TRIS(0), /* don't tri-state outputs */ - FTH(SAI_FIFO_THRESH)); /* fifo threshold (2 or 4 samples) */ - reg_writelf(sai1a, SAI_SUBBLOCK_FRCR, - FSOFF(1), FSPOL(0), FSDEF(1), /* I2S format */ - FSALL(SAI_FRL_VAL/2 - 1), /* FS active for half the frame */ - FRL(SAI_FRL_VAL - 1)); /* set frame length */ - reg_writelf(sai1a, SAI_SUBBLOCK_SLOTR, - SLOTEN(3), NBSLOT(2 - 1), /* enable first two slots */ - SLOTSZ_V(DATASZ), /* slot size = data size */ - FBOFF(0)); /* no bit offset in slot */ - - /* Enable xrun error interrupt */ - reg_writelf(sai1a, SAI_SUBBLOCK_IM, OVRUDR(1)); - - /* Enable interrupts in NVIC */ - nvic_enable_irq(NVIC_IRQN_DMA1_STR0); - nvic_enable_irq(NVIC_IRQN_SAI1); -} - -struct div_settings -{ - uint8_t pllm; - uint8_t plln; - uint8_t pllp; - uint8_t mckdiv; -}; - -_Static_assert(STM32_HSE_FREQ == 24000000, - "Audio clock settings only valid for 24MHz HSE"); - -static const struct div_settings div_settings[] = { - [HW_FREQ_96] = { .pllm = 5, .plln = 128 - 1, .pllp = 25 - 1, .mckdiv = 0 }, /* exact */ - [HW_FREQ_48] = { .pllm = 5, .plln = 128 - 1, .pllp = 25 - 1, .mckdiv = 2 }, /* exact */ - [HW_FREQ_24] = { .pllm = 5, .plln = 128 - 1, .pllp = 25 - 1, .mckdiv = 4 }, /* exact */ - [HW_FREQ_12] = { .pllm = 5, .plln = 128 - 1, .pllp = 25 - 1, .mckdiv = 8 }, /* exact */ - - [HW_FREQ_88] = { .pllm = 4, .plln = 143 - 1, .pllp = 38 - 1, .mckdiv = 0 }, /* -11ppm error */ - [HW_FREQ_44] = { .pllm = 4, .plln = 143 - 1, .pllp = 38 - 1, .mckdiv = 2 }, /* -11ppm error */ - [HW_FREQ_22] = { .pllm = 5, .plln = 147 - 1, .pllp = 125 - 1, .mckdiv = 0 }, /* exact */ - [HW_FREQ_11] = { .pllm = 5, .plln = 147 - 1, .pllp = 125 - 1, .mckdiv = 2 }, /* exact */ - - [HW_FREQ_64] = { .pllm = 15, .plln = 256 - 1, .pllp = 25 - 1, .mckdiv = 0 }, /* exact */ - [HW_FREQ_32] = { .pllm = 15, .plln = 256 - 1, .pllp = 25 - 1, .mckdiv = 2 }, /* exact */ - [HW_FREQ_16] = { .pllm = 15, .plln = 256 - 1, .pllp = 25 - 1, .mckdiv = 4 }, /* exact */ - [HW_FREQ_8 ] = { .pllm = 15, .plln = 256 - 1, .pllp = 25 - 1, .mckdiv = 8 }, /* exact */ -}; - -static void sai_set_pll(const struct div_settings *div) -{ - /* Disable the PLL so it can be reconfigured */ - if (reg_readf(RCC_CR, PLL2RDY)) - { - reg_writef(RCC_CR, PLL2ON(0)); - while (reg_readf(RCC_CR, PLL2RDY)); - while (reg_readf(RCC_CR, PLL2ON)); - } - - /* Set the PLL configuration */ - uint32_t pllcfgr = reg_var(RCC_PLLCFGR); - - reg_writef(RCC_PLLCKSELR, - DIVM2(div->pllm)); - reg_writef(RCC_PLL2DIVR, - DIVN(div->plln), - DIVP(div->pllp), - DIVQ(0), DIVR(0)); - reg_vwritef(pllcfgr, RCC_PLLCFGR, - DIVP2EN(1), - DIVQ2EN(0), - DIVR2EN(0), - PLL2FRACEN(0), - PLL2VCOSEL_V(WIDE)); - - if (div->pllm <= 6) - reg_vwritef(pllcfgr, RCC_PLLCFGR, PLL2RGE_V(4_8MHZ)); - else if (div->pllm <= 12) - reg_vwritef(pllcfgr, RCC_PLLCFGR, PLL2RGE_V(2_4MHZ)); - else - reg_vwritef(pllcfgr, RCC_PLLCFGR, PLL2RGE_V(1_2MHZ)); - - reg_var(RCC_PLLCFGR) = pllcfgr; - - /* Enable the PLL again */ - reg_writef(RCC_CR, PLL2ON(1)); - while (!reg_readf(RCC_CR, PLL2RDY)); -} - -static void sai_set_frequency(int freq) -{ - /* Can't change frequency while the SAI is active */ - if (reg_readlf(sai1a, SAI_SUBBLOCK_CR1, SAIEN)) - panicf("%s while SAI active", __func__); - - const struct div_settings *div = &div_settings[freq]; - const struct div_settings *old_div = NULL; - - if (pcm_last_freq >= 0) - old_div = &div_settings[pcm_last_freq]; - - if (old_div == NULL || - div->pllm != old_div->pllm || - div->plln != old_div->plln || - div->pllp != old_div->pllp) - { - sai_set_pll(div); - } - - /* Configure the MCK divider in SAI */ - reg_writelf(sai1a, SAI_SUBBLOCK_CR1, MCKDIV(div->mckdiv)); -} - -static void sink_dma_init(void) -{ - sai_init(); - audiohw_init(); -} - -static void sink_set_freq(uint16_t freq) -{ - /* - * Muting here doesn't seem to help clicks when switching - * tracks of different frequencies; the audio may need to - * be soft-muted in software before the switch. - */ - audiohw_mute(true); - - sai_set_frequency(freq); - pcm_last_freq = freq; - - audiohw_mute(false); -} - -static void sink_dma_start(const void *addr, size_t size) -{ - play_active = true; - - play_dma_start(addr, size); - reg_writelf(sai1a, SAI_SUBBLOCK_CR1, SAIEN(1), DMAEN(1)); - reg_writelf(sai1a, SAI_SUBBLOCK_CR2, MUTE(0)); -} - -static void sink_dma_stop(void) -{ - play_active = false; - - reg_writelf(sai1a, SAI_SUBBLOCK_CR2, MUTE(1)); - reg_writelf(sai1a, SAI_SUBBLOCK_CR1, SAIEN(0), DMAEN(0)); - reg_writelf(dma1_ch0, DMA_CHN_CR, EN(0)); - - /* Must wait for SAIEN bit to be cleared */ - while (reg_readlf(sai1a, SAI_SUBBLOCK_CR1, SAIEN)); -} - -static void sink_lock(void) -{ - /* Disable IRQ on first lock */ - if (atomic_fetch_add_explicit(&play_lock, 1, memory_order_relaxed) == 0) - nvic_disable_irq_sync(NVIC_IRQN_DMA1_STR0); -} - -static void sink_unlock(void) -{ - /* Enable IRQ on release of last lock */ - if (atomic_fetch_sub_explicit(&play_lock, 1, memory_order_relaxed) == 1) - nvic_enable_irq(NVIC_IRQN_DMA1_STR0); -} - -struct pcm_sink builtin_pcm_sink = { - .caps = { - .samprs = hw_freq_sampr, - .num_samprs = HW_NUM_FREQ, - .default_freq = HW_FREQ_DEFAULT, - }, - .ops = { - .init = sink_dma_init, - .postinit = audiohw_postinit, - .set_freq = sink_set_freq, - .lock = sink_lock, - .unlock = sink_unlock, - .play = sink_dma_start, - .stop = sink_dma_stop, - }, -}; - -void dma1_ch0_irq_handler(void) -{ - uint32_t lisr = reg_varl(dma1, DMA_LISR); - const void *addr; - size_t size; - - if (reg_vreadf(lisr, DMA_LISR, TEIF0)) - pcm_dma_teif_count++; - - if (reg_vreadf(lisr, DMA_LISR, DMEIF0)) - pcm_dma_dmeif_count++; - - if (reg_vreadf(lisr, DMA_LISR, FEIF0)) - pcm_dma_feif_count++; - - if (reg_vreadf(lisr, DMA_LISR, TEIF0) || - reg_vreadf(lisr, DMA_LISR, DMEIF0) || - reg_vreadf(lisr, DMA_LISR, FEIF0)) - { - reg_assignlf(dma1, DMA_LIFCR, TEIF0(1), DMEIF0(1), FEIF0(1)); - reg_writelf(dma1_ch0, DMA_CHN_CR, EN(0)); - - pcm_play_dma_status_callback(PCM_DMAST_ERR_DMA); - } - else if (reg_vreadf(lisr, DMA_LISR, TCIF0)) - { - reg_assignlf(dma1, DMA_LIFCR, TCIF0(1)); - - /* If we call the complete callback while not playing - * it can cause the PCM layer to get stuck... somehow */ - if (!play_active) - return; - - if (pcm_play_dma_complete_callback(PCM_DMAST_OK, &addr, &size)) - play_dma_start(addr, size); - } - else - { - panicf("%s: %08lx", __func__, lisr); - } -} - -void sai1_irq_handler(void) -{ - if (reg_readlf(sai1a, SAI_SUBBLOCK_SR, OVRUDR)) - { - reg_assignlf(sai1a, SAI_SUBBLOCK_CLRFR, OVRUDR(1)); - pcm_sai_xrun_count++; - } -} diff --git a/firmware/target/arm/stm32/pcm-stm32h7.c b/firmware/target/arm/stm32/pcm-stm32h7.c new file mode 100644 index 0000000000..472ab4a3b9 --- /dev/null +++ b/firmware/target/arm/stm32/pcm-stm32h7.c @@ -0,0 +1,105 @@ +/*************************************************************************** + * __________ __ ___. + * Open \______ \ ____ ____ | | _\_ |__ _______ ___ + * Source | _// _ \_/ ___\| |/ /| __ \ / _ \ \/ / + * Jukebox | | ( <_> ) \___| < | \_\ ( <_> > < < + * Firmware |____|_ /\____/ \___ >__|_ \|___ /\____/__/\_ \ + * \/ \/ \/ \/ \/ + * $Id$ + * + * Copyright (C) 2025 Aidan MacDonald + * + * 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 "pcm.h" +#include "pcm-internal.h" +#include "pcm_sampr.h" +#include "pcm_sink.h" + +static void sink_set_freq(uint16_t freq) +{ + (void)freq; +} + +static void sink_dma_init(void) +{ +} + +static void sink_dma_postinit(void) +{ +} + +static void sink_dma_start(const void *addr, size_t size) +{ + (void)addr; + (void)size; +} + +static void sink_dma_stop(void) +{ +} + +static void sink_lock(void) +{ +} + +static void sink_unlock(void) +{ +} + +struct pcm_sink builtin_pcm_sink = { + .caps = { + .samprs = hw_freq_sampr, + .num_samprs = HW_NUM_FREQ, + .default_freq = HW_FREQ_DEFAULT, + }, + .ops = { + .init = sink_dma_init, + .postinit = sink_dma_postinit, + .set_freq = sink_set_freq, + .lock = sink_lock, + .unlock = sink_unlock, + .play = sink_dma_start, + .stop = sink_dma_stop, + }, +}; + +#ifdef HAVE_RECORDING +void pcm_rec_dma_init(void) +{ +} + +void pcm_rec_dma_close(void) +{ +} + +void pcm_rec_dma_start(void *addr, size_t size) +{ + (void)addr; + (void)size; +} + +void pcm_rec_dma_stop(void) +{ +} + +void pcm_rec_lock(void) +{ +} + +void pcm_rec_unlock(void) +{ +} + +const void *pcm_rec_dma_get_peak_buffer(void) +{ + return NULL; +} +#endif diff --git a/firmware/usbstack/usb_audio.c b/firmware/usbstack/usb_audio.c index f8d431705f..3e7bab7f45 100644 --- a/firmware/usbstack/usb_audio.c +++ b/firmware/usbstack/usb_audio.c @@ -493,6 +493,7 @@ static void set_playback_sampling_frequency(unsigned long f) hw_freq_sampr[as_playback_freq_idx], f); mixer_set_frequency(hw_freq_sampr[as_playback_freq_idx]); + pcm_apply_settings(); } unsigned long usb_audio_get_playback_sampling_frequency(void) @@ -705,6 +706,7 @@ static void usb_audio_start_playback(void) #endif logf("usbaudio: start playback at %lu Hz", hw_freq_sampr[as_playback_freq_idx]); mixer_set_frequency(hw_freq_sampr[as_playback_freq_idx]); + pcm_apply_settings(); mixer_channel_set_amplitude(PCM_MIXER_CHAN_USBAUDIO, MIX_AMP_UNITY); usb_drv_recv_nonblocking(EP_ISO_OUT, rx_buffer, BUFFER_SIZE); @@ -785,7 +787,7 @@ int usb_audio_get_alt_intf(void) { return usb_as_playback_intf_alt; } - + int32_t usb_audio_get_samplesperframe(void) { return samples_fb; @@ -1346,7 +1348,7 @@ bool usb_audio_fast_transfer_complete(int ep, int dir, int status, int length) { retval = false; } - + // send feedback value every N frames! // NOTE: important that we need to queue this up _the frame before_ it's needed - on MacOS especially! if ((usb_drv_get_frame_number()+1) % FEEDBACK_UPDATE_RATE_FRAMES == 0 && send_fb)