pcm: Make more of the low-level PCM API private

* pcm_play_data
 * pcm_play_stop
 * pcm_play_stop_int
 * pcm_is_playing
 * pcm_set_frequency
 * pcm_get_frequency
 * pcm_apply_settings

Now, the only user of these functions are the mixer and recording layers
that provide a higher-level API to plugins and the main [playback]
application.

Outside of the PCM core, pcm_apply_settings() was only used immediately
following a call to mixer_set_frequency(), so the latter function
now always calls the former.

Change-Id: I61c3144dc156b9de9b7963160b525c6d10c6ad4b

apps/lang/bulgarian.lang

@@ -12507,20 +12507,6 @@
     *: "Средна побитова скорост"
   </voice>
 </phrase>
-<phrase>
-  id: LANG_PLAYTIME_ERROR
-  desc: playing time screen
-  user: core
-  <source>
-    *: "Error while gathering info"
-  </source>
-  <dest>
-    *: "Грешка при събирането на информация"
-  </dest>
-  <voice>
-    *: "Грешка при събирането на информация"
-  </voice>
-</phrase>
 <phrase>
   id: LANG_PLAYING_TIME
   desc: onplay menu
@@ -14686,7 +14672,7 @@
   user: core
   <source>
     *: "Press LEFT to cancel."
-    android,hifietma*,zenvision: "Press BACK to cancel."
+    android,hifietma*: "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."
@@ -14701,7 +14687,7 @@
   </source>
   <dest>
     *: "Натиснете LEFT за отказ."
-    android,hifietma*,zenvision: "Натиснете BACK за отказ."
+    android,hifietma*: "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."
@@ -14716,7 +14702,7 @@
   </dest>
   <voice>
     *: "Натиснете LEFT за отказ."
-    android,hifietma*,zenvision: "Натиснете BACK за отказ."
+    android,hifietma*: "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."
@@ -15211,13 +15197,13 @@
 </phrase>
 <phrase>
   id: LANG_VOICED_DATE_FORMAT
-  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"
+  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
   user: core
   <source>
     *: "dAY"
   </source>
   <dest>
-    *: "~dAY"
+    *: "dAy"
   </dest>
   <voice>
     *: ""
@@ -15832,20 +15818,6 @@
     *: "Браузър по подразбиране"
   </voice>
 </phrase>
-<phrase>
-  id: LANG_AMAZE_MENU
-  desc: Amaze game
-  user: core
-  <source>
-    *: "Amaze Main Menu"
-  </source>
-  <dest>
-    *: "Главно меню на Amaze"
-  </dest>
-  <voice>
-    *: "Главно меню на Amaze"
-  </voice>
-</phrase>
 <phrase>
   id: LANG_SET_MAZE_SIZE
   desc: Maze size in Amaze game
@@ -16098,34 +16070,6 @@
     *: "Настройки на Mik mod"
   </voice>
 </phrase>
-<phrase>
-  id: LANG_MIKMOD_MENU
-  desc: mikmod plugin
-  user: core
-  <source>
-    *: "Mikmod Menu"
-  </source>
-  <dest>
-    *: "Меню на Mikmod"
-  </dest>
-  <voice>
-    *: "Меню на Mik mod"
-  </voice>
-</phrase>
-<phrase>
-  id: LANG_CHESSBOX_MENU
-  desc: chessbox plugin
-  user: core
-  <source>
-    *: "Chessbox Menu"
-  </source>
-  <dest>
-    *: "Меню на Chessbox"
-  </dest>
-  <voice>
-    *: "Меню на Chess box"
-  </voice>
-</phrase>
 <phrase>
   id: VOICE_INVALID_VOICE_FILE
   desc: played if the voice file fails to load
@@ -16987,3 +16931,48 @@
     usbdac: "USB ЦАП е активен"
   </voice>
 </phrase>
+<phrase>
+  id: LANG_ANNOUNCE_STATUS
+  desc: announnnce_status plugin
+  user: core
+  <source>
+    *: "Announce Status"
+  </source>
+  <dest>
+    *: "Съобщаване на състоянието"
+  </dest>
+  <voice>
+    *: "Съобщаване на състоянието"
+  </voice>
+</phrase>
+<phrase>
+  id: LANG_USE_LED_INDICATORS
+  desc: LED indicators setting
+  user: core
+  <source>
+    *: none
+    general_purpose_led: "Use LED indicators"
+  </source>
+  <dest>
+    *: none
+    general_purpose_led: "Използване на светодиодни индикатори"
+  </dest>
+  <voice>
+    *: none
+    general_purpose_led: "Използване на светодиодни индикатори"
+  </voice>
+</phrase>
+<phrase>
+  id: LANG_KEEP_DIRECTORY
+  desc: file browser setting
+  user: core
+  <source>
+    *: "Always remember last folder"
+  </source>
+  <dest>
+    *: "Винаги помни последната папка"
+  </dest>
+  <voice>
+    *: "Винаги помни последната папка"
+  </voice>
+</phrase>

apps/lang/deutsch.lang

@@ -14690,7 +14690,7 @@
   user: core
   <source>
     *: "Press LEFT to cancel."
-    android,hifietma*,zenvision: "Press BACK to cancel."
+    android,hifietma*: "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 @@
   </source>
   <dest>
     *: "Zum Abbrechen LINKS drücken."
-    android,hifietma*,vibe500,zenvision: "Zum Abbrechen ZURÜCK drücken."
+    android,hifietma*,vibe500: "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 @@
   </dest>
   <voice>
     *: "Zum Abbrechen LINKS drücken."
-    android,hifietma*,vibe500,zenvision: "Zum Abbrechen ZURÜCK drücken."
+    android,hifietma*,vibe500: "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 @@
 </phrase>
 <phrase>
   id: LANG_VOICED_DATE_FORMAT
-  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"
+  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
   user: core
   <source>
     *: "dAY"
@@ -16978,3 +16978,17 @@
     *: "Status mitteilen"
   </voice>
 </phrase>
+<phrase>
+  id: LANG_KEEP_DIRECTORY
+  desc: file browser setting
+  user: core
+  <source>
+    *: "Always remember last folder"
+  </source>
+  <dest>
+    *: "Zuletzt verwendeten Ordner immer merken"
+  </dest>
+  <voice>
+    *: "Zuletzt verwendeten Ordner immer merken"
+  </voice>
+</phrase>

apps/plugin.c

@@ -644,7 +644,6 @@ 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,

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 279
+#define PLUGIN_API_VERSION 280
 
 /* 239 Marks the removal of ARCHOS HWCODEC and CHARCELL */
 
@@ -743,7 +743,6 @@ 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);

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 = SAMPLECOUNT*2*sizeof(short);
+   *size = MIXBUFFERSIZE*sizeof(short);
 }
 
 

apps/plugins/lua/include_lua/pcm.lua

@@ -24,6 +24,5 @@
 -- [[ 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

apps/plugins/lua/rocklib.c

@@ -549,10 +549,10 @@ RB_WRAP(sound)
 
 RB_WRAP(pcm)
 {
-    enum e_pcm {PCM_APPLYSETTINGS = 0, PCM_PLAYLOCK, PCM_PLAYUNLOCK,
+    enum e_pcm {PCM_PLAYLOCK = 0, PCM_PLAYUNLOCK,
                 PCM_ECOUNT};
 
-    const char *pcm_option[] = {"apply_settings", "play_lock", "play_unlock",
+    const char *pcm_option[] = {"play_lock", "play_unlock",
                                 NULL};
 
     lua_pushnil(L); /*push nil so options w/o return have something to return */
@@ -560,9 +560,6 @@ 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;

apps/plugins/pdbox/PDa/src/s_audio_rockbox.c

@@ -64,7 +64,6 @@ 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++)
@@ -86,7 +85,6 @@ void rockbox_close_audio(void)
 
     /* Restore default sampling rate. */
     rb->mixer_set_frequency(HW_SAMPR_DEFAULT);
-    rb->pcm_apply_settings();
 }
 
 /* Rockbox audio callback. */

apps/plugins/test_sampr.c

@@ -168,7 +168,6 @@ static void set_frequency(int index)
     update_gen_step();
 
     rb->mixer_set_frequency(hw_sampr);
-    rb->pcm_apply_settings();
 }
 
 #ifndef HAVE_VOLUME_IN_LIST

apps/plugins/xrick/system/syssnd_rockbox.c

@@ -253,7 +253,6 @@ 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));
@@ -286,7 +285,6 @@ void syssnd_shutdown(void)
 
     /* Restore default sampling rate. */
     rb->mixer_set_frequency(HW_SAMPR_DEFAULT);
-    rb->pcm_apply_settings();
 
     rb->talk_disable(false);
 

apps/recorder/pcm_record.c

@@ -616,6 +616,10 @@ 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)
 {
@@ -876,7 +880,7 @@ 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;

docs/PLUGIN_API

@@ -1754,10 +1754,6 @@ 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))

firmware/SOURCES

@@ -1942,7 +1942,6 @@ 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
@@ -1959,6 +1958,9 @@ 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

firmware/export/config/echor1.h

@@ -51,9 +51,14 @@
 /* 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

firmware/export/echoplayer_codec.h

@@ -21,9 +21,13 @@
 #ifndef __ECHOPLAYER_CODEC_H__
 #define __ECHOPLAYER_CODEC_H__
 
+#include <stdbool.h>
+
 /* -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__ */

firmware/export/pcm-internal.h

@@ -71,6 +71,20 @@ 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);
 
@@ -125,7 +139,6 @@ 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

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. */
@@ -54,15 +54,6 @@ 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);
@@ -77,11 +68,6 @@ 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
@@ -92,9 +78,6 @@ 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 **/

firmware/pcm.c

@@ -94,8 +94,6 @@ 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];

firmware/pcm_mixer.c

@@ -475,6 +475,9 @@ 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 */

firmware/target/arm/stm32/debug-stm32h7.c

@@ -86,11 +86,68 @@ 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},
 };
 

firmware/target/arm/stm32/echoplayer/audiohw-echoplayer.c

@@ -19,21 +19,151 @@
  *
  ****************************************************************************/
 #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)
 {
-    (void)vol_l;
-    (void)vol_r;
+    pcm_set_master_volume(vol_l, vol_r);
 }

firmware/target/arm/stm32/echoplayer/clock-echoplayer.c

@@ -52,6 +52,7 @@ 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),
@@ -164,7 +165,7 @@ 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, SPI45SEL_V(HSE));
+    reg_writef(RCC_D2CCIP1R, SAI1SEL_V(PLL2P), SPI45SEL_V(HSE));
     reg_writef(RCC_D2CCIP2R, I2C123SEL_V(HSI));
 
     /* Enable AXI SRAM in sleep mode to allow DMA'ing out of it */

firmware/target/arm/stm32/echoplayer/pcm-echoplayer.c

@@ -0,0 +1,381 @@
+/***************************************************************************
+ *             __________               __   ___.
+ *   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 <stdatomic.h>
+
+/* 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++;
+    }
+}

firmware/target/arm/stm32/pcm-stm32h7.c

@@ -1,105 +0,0 @@
-/***************************************************************************
- *             __________               __   ___.
- *   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

firmware/usbstack/usb_audio.c

@@ -493,7 +493,6 @@ 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)
@@ -706,7 +705,6 @@ 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);
@@ -787,7 +785,7 @@ int usb_audio_get_alt_intf(void)
 {
     return usb_as_playback_intf_alt;
 }
-    
+
 int32_t usb_audio_get_samplesperframe(void)
 {
     return samples_fb;
@@ -1348,7 +1346,7 @@ 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)