voice: Add Norwegian Nynorsk voicefile synthesis

(Enabled, as it now has >98% coverage)

Change-Id: Ie14c28cf279cf8bb2215deb8f1ec597cb980b3ab

apps/lang/russian.lang

@@ -13142,20 +13142,6 @@
     *: "в сторону ферзя"
   </voice>
 </phrase>
-<phrase>
-  id: LANG_PLAYTIME_ERROR
-  desc: playing time screen
-  user: core
-  <source>
-    *: "Error while gathering info"
-  </source>
-  <dest>
-    *: "Сбой при сборе информации"
-  </dest>
-  <voice>
-    *: "Сбой при сборе информации"
-  </voice>
-</phrase>
 <phrase>
   id: LANG_CLEAR_ALL_RESUMES
   desc: in the mpegplayer settings menu
@@ -14389,7 +14375,7 @@
   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."
@@ -14404,7 +14390,7 @@
   </source>
   <dest>
     *: "Нажмите ВЛЕВО для отмены."
-    android,hifietma*,zenvision: "Нажмите НАЗАД для отмены."
+    android,hifietma*: "Нажмите НАЗАД для отмены."
     cowond2,creativezenxfi2,ibassodx50,ibassodx90,mrobe500,ondavx747: "Нажмите ВКЛ. для отмены."
     ihifi760,ihifi960: "Дважды нажмите ВОЗВРАТ для отмены."
     ihifi770,ihifi770c,ihifi800: "Нажмите ДОМОЙ для отмены."
@@ -14419,7 +14405,7 @@
   </dest>
   <voice>
     *: "Нажмите кнопку Влево для отмены."
-    android,hifietma*,zenvision: "Нажмите кнопку Назад для отмены."
+    android,hifietma*: "Нажмите кнопку Назад для отмены."
     cowond2,creativezenxfi2,ibassodx50,ibassodx90,mrobe500,ondavx747: "Нажмите кнопку включения для отмены."
     ihifi760,ihifi960: "Дважды нажмите кнопку Назад для отмены."
     ihifi770,ihifi770c,ihifi800: "Нажмите кнопку Домой для отмены."
@@ -15790,13 +15776,13 @@
 </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>
     *: ""
@@ -15872,20 +15858,6 @@
     *: "Браузер по умолчанию"
   </voice>
 </phrase>
-<phrase>
-  id: LANG_AMAZE_MENU
-  desc: Amaze game
-  user: core
-  <source>
-    *: "Amaze Main Menu"
-  </source>
-  <dest>
-    *: "Главное меню Amaze"
-  </dest>
-  <voice>
-    *: "Главное меню Amaze"
-  </voice>
-</phrase>
 <phrase>
   id: LANG_SET_MAZE_SIZE
   desc: Maze size in Amaze game
@@ -16138,34 +16110,6 @@
     *: "Настройки Mik mod"
   </voice>
 </phrase>
-<phrase>
-  id: LANG_MIKMOD_MENU
-  desc: mikmod plugin
-  user: core
-  <source>
-    *: "Mikmod Menu"
-  </source>
-  <dest>
-    *: "Меню Mikmod"
-  </dest>
-  <voice>
-    *: "Меню Mik mod"
-  </voice>
-</phrase>
-<phrase>
-  id: LANG_CHESSBOX_MENU
-  desc: chessbox plugin
-  user: core
-  <source>
-    *: "Chessbox Menu"
-  </source>
-  <dest>
-    *: "Меню Chessbox"
-  </dest>
-  <voice>
-    *: "Меню Chessbox"
-  </voice>
-</phrase>
 <phrase>
   id: VOICE_INVALID_VOICE_FILE
   desc: played if the voice file fails to load
@@ -16999,3 +16943,48 @@
     *: "Все трэки сортируются по альбому"
   </voice>
 </phrase>
+<phrase>
+  id: LANG_ANNOUNCE_STATUS
+  desc: announnnce_status plugin
+  user: core
+  <source>
+    *: "Announce Status"
+  </source>
+  <dest>
+    *: "Объявлять статус"
+  </dest>
+  <voice>
+    *: "Объявлять статус"
+  </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: "Использовать LED индикаторы"
+  </dest>
+  <voice>
+    *: none
+    general_purpose_led: "Использовать LED индикаторы"
+  </voice>
+</phrase>
+<phrase>
+  id: LANG_KEEP_DIRECTORY
+  desc: file browser setting
+  user: core
+  <source>
+    *: "Always remember last folder"
+  </source>
+  <dest>
+    *: "Всегда запоминать последнюю папку"
+  </dest>
+  <voice>
+    *: "Всегда запоминать последнюю папку"
+  </voice>
+</phrase>

apps/misc.c

@@ -866,20 +866,10 @@ void check_bootfile(bool do_rolo)
 #endif
 #endif
 
-/* check range, set volume and save settings */
+/* set volume and save settings */
 void setvol(void)
 {
-    const int min_vol = sound_min(SOUND_VOLUME);
+    sound_set_volume(global_status.volume);
-    const int max_vol = sound_max(SOUND_VOLUME);
-    int volume = global_status.volume;
-    if (volume < min_vol)
-        volume = min_vol;
-    if (volume > max_vol)
-        volume = max_vol;
-    if (volume > global_settings.volume_limit)
-        volume = global_settings.volume_limit;
-
-    sound_set_volume(volume);
     global_status.last_volume_change = current_tick;
     status_save(false);
 }

apps/plugins/disktidy.config

@@ -19,9 +19,13 @@
     .Trashes/: no
     .Spotlight-V100/: no
     .DS_Store: no
+    __MACOSX/: no
     ._*: no
 < Linux/BSD >: no
     .Trash-*/: no
     .dolphinview: no
     .d3lphinview: no
+< Android >: no
+    Android/: no
+    LOST.DIR/: no
 < Other >: no

apps/plugins/midi/midiplay.c

@@ -381,6 +381,8 @@
 
 struct MIDIfile * mf IBSS_ATTR;
 
+int sample_rate IBSS_ATTR;
+int max_voices IBSS_ATTR;
 int number_of_samples IBSS_ATTR; /* the number of samples in the current tick */
 int playing_time IBSS_ATTR;  /* How many seconds into the file have we been playing? */
 int samples_this_second IBSS_ATTR;    /* How many samples produced during this second so far? */
@@ -490,11 +492,56 @@ static void get_more(const void** start, size_t* size)
     *size = samples_in_buf*sizeof(int32_t);
 }
 
+UNUSED_ATTR static int find_min_sampr_ge_22(void)
+{
+    const struct pcm_sink_caps* caps = rb->pcm_current_sink_caps();
+    int ret = caps->samprs[0];
+    for (size_t i = 1; i < caps->num_samprs; i += 1)
+    {
+        /* caps->samprs is in descending order */
+        if (caps->samprs[i] >= SAMPR_22)
+            ret = caps->samprs[i];
+        else
+            break;
+    }
+    return ret;
+}
+
 static int midimain(const void * filename)
 {
     int a, notes_used, vol;
     bool is_playing = true;  /* false = paused */
 
+    /* decide sample_rate and max_voices */
+#if defined(SIMULATOR) /* Simulator requires 44100Hz, and we can afford to use more voices */ || \
+    (CONFIG_PLATFORM & PLATFORM_HOSTED) /* All hosted targets have CPU to spare */ || \
+    defined(CPU_MIPS) /* All MIPS targets are pretty fast */
+    sample_rate = SAMPR_44;
+    max_voices = 48;
+#elif defined(CPU_PP)
+    /* Some of the pp based targets can't handle too many voices
+       mainly because they have to use 44100Hz sample rate, this could be
+       improved to increase max_voices for targets that can do 22kHz */
+    sample_rate = find_min_sampr_ge_22();
+    max_voices = sample_rate == SAMPR_22 ? 24 : 16;
+#elif defined(CPU_ARM)
+    /* ARMv4 targets are slow, but treat everything else as fast */
+#if (ARM_ARCH >= 6)
+    sample_rate = SAMPR_44;
+    max_voices = 32;
+#elif (ARM_ARCH >= 5)
+    sample_rate = find_min_sampr_ge_22();
+    max_voices = 32;
+#else /* ie v4 */
+    sample_rate = find_min_sampr_ge_22();
+    max_voices = sample_rate == SAMPR_22 ? 24 : 16;
+#endif
+#else /* !CPU_ARM */
+    /* Treat everything else as slow */
+    sample_rate = find_min_sampr_ge_22();
+    max_voices = sample_rate == SAMPR_22 ? 24 : 16;
+#endif
+
 #if defined(HAVE_ADJUSTABLE_CPU_FREQ)
     rb->cpu_boost(true);
 #endif
@@ -537,7 +584,7 @@ static int midimain(const void * filename)
     rb->dsp_set_timestretch(PITCH_SPEED_100);
 #endif
     rb->dsp_configure(dsp, DSP_SET_SAMPLE_DEPTH, 22);
-    rb->dsp_configure(dsp, DSP_SET_FREQUENCY, SAMPLE_RATE); /* 44100 22050 11025 */
+    rb->dsp_configure(dsp, DSP_SET_FREQUENCY, sample_rate); /* 44100 22050 11025 */
     rb->dsp_configure(dsp, DSP_SET_STEREO_MODE, STEREO_INTERLEAVED);
 
     /*
@@ -553,14 +600,14 @@ static int midimain(const void * filename)
     midi_debug("Okay, starting sequencing");
 
     bpm = mf->div*1000000/tempo;
-    number_of_samples = SAMPLE_RATE/bpm;
+    number_of_samples = sample_rate/bpm;
 
     /* Skip over any junk in the beginning of the file, so start playing */
     /* after the first note event */
     do
     {
         notes_used = 0;
-        for (a = 0; a < MAX_VOICES; a++)
+        for (a = 0; a < max_voices; a++)
             if (voices[a].isUsed)
                 notes_used++;
         tick();

apps/plugins/midi/midiutil.c

@@ -35,7 +35,7 @@ struct GPatch * gusload(char *);
 struct GPatch * patchSet[128];
 struct GPatch * drumSet[128];
 
-struct SynthObject voices[MAX_VOICES] IBSS_ATTR;
+struct SynthObject voices[48] IBSS_ATTR;
 
 static void *alloc(int size)
 {

apps/plugins/midi/midiutil.h

@@ -27,67 +27,6 @@
 #define NBUF   2
 #define MAX_SAMPLES 512
 
-#ifdef SIMULATOR
-
-/* Simulator requires 44100Hz, and we can afford to use more voices */
-#define SAMPLE_RATE SAMPR_44
-#define MAX_VOICES 48
-
-#elif (CONFIG_PLATFORM & PLATFORM_HOSTED)
-
-/* All hosted targets have CPU to spare */
-#define MAX_VOICES 48
-#define SAMPLE_RATE SAMPR_44
-
-#elif defined(CPU_PP)
-
-/* Some of the pp based targets can't handle too many voices
-   mainly because they have to use 44100Hz sample rate, this could be
-   improved to increase MAX_VOICES for targets that can do 22kHz */
-#define SAMPLE_RATE HW_SAMPR_MIN_GE_22
-#if HW_SAMPR_CAPS & SAMPR_CAP_22
-#define MAX_VOICES 24 /* General MIDI minimum */
-#else
-#define MAX_VOICES 16
-#endif
-
-#elif defined(CPU_MIPS)
-
-/* All MIPS targets are pretty fast */
-#define MAX_VOICES 48
-#define SAMPLE_RATE SAMPR_44
-
-#elif defined(CPU_ARM)
-
-/* ARMv4 targets are slow, but treat everything else as fast */
-
-#if (ARM_ARCH >= 6)
-#define MAX_VOICES 32
-#define SAMPLE_RATE SAMPR_44
-#elif (ARM_ARCH >= 5)
-#define MAX_VOICES 32
-#define SAMPLE_RATE HW_SAMPR_MIN_GE_22
-#else /* ie v4 */
-#define SAMPLE_RATE HW_SAMPR_MIN_GE_22
-#if HW_SAMPR_CAPS & SAMPR_CAP_22
-#define MAX_VOICES 24 /* General MIDI minimum */
-#else
-#define MAX_VOICES 16
-#endif
-#endif /* ARM_ARCH < 5*/
-
-#else /* !CPU_ARM */
-
-/* Treat everything else as slow */
-#define SAMPLE_RATE HW_SAMPR_MIN_GE_22
-#if HW_SAMPR_CAPS & SAMPR_CAP_22
-#define MAX_VOICES 24 /* General MIDI minimum */
-#else
-#define MAX_VOICES 16
-#endif
-
-#endif /* Wrap it up. */
-
 #define BYTE unsigned char
 
 /* Data chunk ID types, returned by readID() */
@@ -189,7 +128,7 @@ void * readData(int file, int len);
 #define malloc(n) my_malloc(n)
 void * my_malloc(int size);
 
-extern struct SynthObject voices[MAX_VOICES];
+extern struct SynthObject voices[48]; /* 48: maximum possible max_voices */
 
 extern int chVol[16];       /* Channel volume                */
 extern int chPan[16];       /* Channel panning               */
@@ -207,6 +146,8 @@ extern struct GPatch * drumSet[128];
 
 extern struct MIDIfile * mf;
 
+extern int sample_rate;
+extern int max_voices;
 extern int number_of_samples;
 extern int playing_time IBSS_ATTR;
 extern int samples_this_second IBSS_ATTR;

apps/plugins/midi/sequencer.c

@@ -65,7 +65,7 @@ static inline void setVol(int ch, int vol)
 
     /* If channel volume changes, we need to recalculate the volume scale */
     /* factor for all voices active on this channel                       */
-    for (a = 0; a < MAX_VOICES; a++)
+    for (a = 0; a < max_voices; a++)
         if (voices[a].ch == ch)
             setVolScale(a);
 }
@@ -156,7 +156,7 @@ static void findDelta(struct SynthObject * so, int ch, int note)
     delta = (((freqtable[note+chPBNoteOffset[ch]]))); /* anywhere from 8000 to 8000000 */
     delta = delta * wf->sampRate;           /* approx 20000 - 44000 but can vary with tuning */
     delta = (delta * chPBFractBend[ch]);    /* approx 60000 - 70000 */
-    delta = delta / (SAMPLE_RATE);  /* 44100 or 22050 */
+    delta = delta / (sample_rate);  /* 44100 or 22050 */
     delta = delta / (wf->rootFreq); /* anywhere from 8000 to 8000000 */
 
     /* Pitch bend is encoded as a fractional of 16 bits, hence the 16 */
@@ -167,7 +167,7 @@ static void findDelta(struct SynthObject * so, int ch, int note)
 static inline void computeDeltas(int ch)
 {
     int a;
-    for (a = 0; a < MAX_VOICES; a++)
+    for (a = 0; a < max_voices; a++)
     {
         if (voices[a].isUsed && voices[a].ch == ch)
         {
@@ -210,7 +210,7 @@ static inline void pressNote(int ch, int note, int vol)
     if(ch == 15) return;
 */
     int a;
-    for (a = 0; a < MAX_VOICES; a++)
+    for (a = 0; a < max_voices; a++)
     {
         if (voices[a].ch == ch && voices[a].note == note)
             break;
@@ -218,7 +218,7 @@ static inline void pressNote(int ch, int note, int vol)
         if (!voices[a].isUsed)
             break;
     }
-    if (a == MAX_VOICES)
+    if (a == max_voices)
     {
 //        midi_debug("\nVoice kill");
 //        midi_debug("\nToo many voices playing at once. No more left");
@@ -226,7 +226,7 @@ static inline void pressNote(int ch, int note, int vol)
 //        for(a=0; a<48; a++)
 //            midi_debug("\n#%d  Ch=%d  Note=%d  curRate=%d   curOffset=%d   curPoint=%d   targetOffset=%d", a, voices[a].ch, voices[a].note, voices[a].curRate, voices[a].curOffset, voices[a].curPoint, voices[a].targetOffset);
         lastKill++;
-        if (lastKill == MAX_VOICES)
+        if (lastKill == max_voices)
             lastKill = 0;
         a = lastKill;
 //        return; /* None available */
@@ -263,7 +263,7 @@ static inline void pressNote(int ch, int note, int vol)
 
             struct GWaveform * wf = drumSet[note]->waveforms[0];
             voices[a].wf = wf;
-            voices[a].delta = (((freqtable[note]<<FRACTSIZE) / wf->rootFreq) * wf->sampRate / SAMPLE_RATE);
+            voices[a].delta = (((freqtable[note]<<FRACTSIZE) / wf->rootFreq) * wf->sampRate / sample_rate);
             if (wf->mode & 28)
 //                midi_debug("\nWoah, a drum patch has a loop. Stripping the loop...");
             wf->mode = wf->mode & (255-28);
@@ -285,7 +285,7 @@ static void releaseNote(int ch, int note)
         return;
 
     int a;
-    for (a = 0; a < MAX_VOICES; a++)
+    for (a = 0; a < max_voices; a++)
     {
         if (voices[a].ch == ch && voices[a].note == note)
         {
@@ -432,7 +432,7 @@ int tick(void)
                         tempo = (((short)e->evData[0])<<16)|(((short)e->evData[1])<<8)|(e->evData[2]);
 /*                        midi_debug("\nMeta-Event: Tempo Set = %d", tempo); */
                         bpm=mf->div*1000000/tempo;
-                        number_of_samples=SAMPLE_RATE/bpm;
+                        number_of_samples=sample_rate/bpm;
 
                     }
                 }
@@ -446,9 +446,9 @@ int tick(void)
 
     samples_this_second += number_of_samples;
 
-    while (samples_this_second >= SAMPLE_RATE)
+    while (samples_this_second >= sample_rate)
     {
-        samples_this_second -= SAMPLE_RATE;
+        samples_this_second -= sample_rate;
         playing_time++;
     }
 
@@ -471,7 +471,7 @@ void seekBackward(int nsec)
 
     /* Set the tempo to defalt */
     bpm = mf->div*1000000/tempo;
-    number_of_samples = SAMPLE_RATE/bpm;
+    number_of_samples = sample_rate/bpm;
 
     /* Reset the tracks to start */
     rewindFile();
@@ -484,7 +484,7 @@ void seekBackward(int nsec)
     do
     {
         notes_used = 0;
-        for (a = 0; a < MAX_VOICES; a++)
+        for (a = 0; a < max_voices; a++)
             if (voices[a].isUsed)
                 notes_used++;
         tick();

apps/plugins/midi/synth.c

@@ -46,7 +46,7 @@ static void readTextBlock(int file, char * buf)
 void resetControllers()
 {
     int a=0;
-    for(a=0; a<MAX_VOICES; a++)
+    for(a=0; a<max_voices; a++)
     {
         voices[a].cp=0;
         voices[a].vol=0;
@@ -448,13 +448,13 @@ static inline void synthVoice(struct SynthObject * so, int32_t * out, unsigned i
 size_t synthSamples(int32_t *buf_ptr, size_t num_samples) ICODE_ATTR;
 size_t synthSamples(int32_t *buf_ptr, size_t num_samples)
 {
-    unsigned int i;
+    int i;
     struct SynthObject *voicept;
     size_t nsamples = MIN(num_samples, MAX_SAMPLES);
 
     rb->memset(buf_ptr, 0, nsamples * 2 * sizeof(int32_t));
 
-    for(i=0; i < MAX_VOICES; i++)
+    for(i=0; i < max_voices; i++)
     {
         voicept=&voices[i];
         if(voicept->isUsed)

apps/plugins/pacbox/pacbox.c

@@ -395,19 +395,19 @@ static void start_sound(void)
 #endif
 
     /* Get the closest rate >= to what is preferred */
+    const struct pcm_sink_caps* caps = rb->pcm_current_sink_caps();
     sr_index = rb->round_value_to_list32(PREFERRED_SAMPLING_RATE,
-                        rb->hw_freq_sampr, HW_NUM_FREQ, false);
+                        caps->samprs, caps->num_samprs, false);
 
-    if (rb->hw_freq_sampr[sr_index] < PREFERRED_SAMPLING_RATE
+    if (caps->samprs[sr_index] < PREFERRED_SAMPLING_RATE && sr_index > 0)
-        && sr_index > 0)
     {
         /* Round up */
         sr_index--;
     }
 
-    wsg3_set_sampling_rate(rb->hw_freq_sampr[sr_index]);
+    wsg3_set_sampling_rate(caps->samprs[sr_index]);
 
-    rb->mixer_set_frequency(rb->hw_freq_sampr[sr_index]);
+    rb->mixer_set_frequency(caps->samprs[sr_index]);
     rb->mixer_channel_play_data(PCM_MIXER_CHAN_PLAYBACK, get_more, NULL, 0);
 
     sound_playing = true;

apps/plugins/pdbox/pdbox.c

@@ -188,6 +188,26 @@ enum plugin_status plugin_start(const void* parameter)
     /* Set audio API. */
     sys_set_audio_api(API_ROCKBOX);
 
+    /* Determine sample rate. */
+#if defined(SIMULATOR)
+#define PREFERRED_SAMPR SAMPR_44
+#else
+#define PREFERRED_SAMPR SAMPR_22
+#endif
+    int samplerate;
+    {
+        int i;
+        const struct pcm_sink_caps* caps = rb->pcm_current_sink_caps();
+        for (i = 0; i < caps->num_samprs; i++) {
+            if (caps->samprs[i] == PREFERRED_SAMPR)
+                break;
+        }
+        if (i == caps->num_samprs)
+            samplerate = SAMPR_44;
+        else
+            samplerate = caps->samprs[i];
+    }
+
     /* Initialize audio subsystem. */
     sys_open_audio(0, /* No sound input yet */
                    sys_soundindevlist,
@@ -197,7 +217,7 @@ enum plugin_status plugin_start(const void* parameter)
                    sys_soundoutdevlist,
                    -1, /* Use the default amount (2) of channels */
                    sys_choutlist,
-                   PD_SAMPLERATE, /* Sample rate */
+                   samplerate, /* Sample rate */
                    DEFAULTADVANCE, /* Scheduler advance */
                    1 /* Enable */);
 

apps/plugins/pdbox/pdbox.h

@@ -40,17 +40,9 @@
 #define calloc(elements, elem_size) tlsf_calloc(elements, elem_size)
 
 /* Audio declarations. */
-#ifdef SIMULATOR
+/* Maximum sample rate; used for buffer sizing.
-  #define PD_SAMPLERATE 44100
+ * Actual rate is determined at runtime from pcm sink capabilities. */
-#elif (HW_SAMPR_CAPS & SAMPR_CAP_22)
+#define PD_SAMPLERATE 44100
-  #define PD_SAMPLERATE 22050
-#elif (HW_SAMPR_CAPS & SAMPR_CAP_32)
-  #define PD_SAMPLERATE 32000
-#elif (HW_SAMPR_CAPS & SAMPR_CAP_44)
-  #define PD_SAMPLERATE 44100
-#else
-  #error No sufficient sample rate available!
-#endif
 #define PD_SAMPLES_PER_HZ ((PD_SAMPLERATE / HZ) + \
                            (PD_SAMPLERATE % HZ > 0 ? 1 : 0))
 #define PD_OUT_CHANNELS 2

apps/plugins/xrick/system/syssnd_rockbox.c

@@ -222,8 +222,8 @@ void syssnd_update(void)
 
         if (!isAudioPlaying && fillCount > 0)
         {
-            rb->mixer_channel_play_data(PCM_MIXER_CHAN_PLAYBACK, get_more, NULL, 0);
             isAudioPlaying = true;
+            rb->mixer_channel_play_data(PCM_MIXER_CHAN_PLAYBACK, get_more, NULL, 0);
         }
     }
 }

docs/CREDITS

@@ -758,6 +758,8 @@ Ingmar Steen
 Lourenço Soares
 Yegor Chernyshov
 Javier Gutiérrez Gertrúdix
+Sergey Puskov
+Eivind Ødegård
 
 The libmad team
 The wavpack team

firmware/SOURCES

@@ -527,12 +527,11 @@ drivers/rds.c
 #endif /* HAVE_RDS_CAP */
 #endif /* PLATFORM_NATIVE */
 #endif /* CONFIG_TUNER */
-#endif /* BOOTLOADER */
+#endif /* !BOOTLOADER */
 
 /* Sound */
-sound.c
-
 #ifndef BOOTLOADER
+sound.c
 pcm_sampr.c
 pcm.c
 pcm_mixer.c
@@ -544,7 +543,7 @@ enc_base.c
 #endif /* HAVE_RECORDING */
 
 drivers/audio/audiohw-swcodec.c
-#endif /* BOOTLOADER */
+#endif /* !BOOTLOADER */
 
 /* Audio codec */
 #if !defined(BOOTLOADER)

firmware/export/config.h

@@ -1362,6 +1362,13 @@ Lyre prototype 1 */
 //#define USB_HAS_INTERRUPT -- seems to be broken
 #endif /* CONFIG_USBOTG */
 
+#if CONFIG_USBOTG == USBOTG_ARC
+#define USB_BATCH_SLOTS 16
+#else
+#define USB_BATCH_NON_NATIVE
+#define USB_BATCH_SLOTS 1
+#endif
+
 /* define the class drivers to enable */
 #ifdef BOOTLOADER
 

firmware/export/pcm_sink.h

@@ -47,6 +47,7 @@ struct pcm_sink {
     /* runtime states */
     unsigned long pending_freq;
     unsigned long configured_freq;
+    unsigned long pcm_is_ready;
 };
 
 enum pcm_sink_ids {

firmware/export/usb_drv.h

@@ -109,6 +109,23 @@ int usb_drv_deinit_endpoint(int endpoint);
 int usb_drv_get_frame_number(void);
 #endif
 
+/* batched request api is for workloads that perform very high-frequency,
+ * performance-sensitive isochronous transactions continuously.
+ * this api allows multiple transaction requests to be buffered in udc driver.
+ * so that it can maximize the controller's performance. */
+
+/* called when usb system requires more buffer */
+typedef void(*usb_drv_batch_get_more)(const void** ptr, size_t* len);
+
+/* prepare request queue */
+int usb_drv_batch_init(int ep, usb_drv_batch_get_more get_more);
+/* destroy request queue */
+int usb_drv_batch_deinit(void);
+/* start processing */
+int usb_drv_batch_start(void);
+/* stop processing */
+int usb_drv_batch_stop(void);
+
 /* USB_STRING_INITIALIZER(u"Example String") */
 #define USB_STRING_INITIALIZER(S) { \
     sizeof(struct usb_string_descriptor) + sizeof(S) - sizeof(*S), \

firmware/pcm.c

@@ -76,9 +76,6 @@
  *
  */
 
-/* 'true' when all stages of pcm initialization have completed */
-static bool pcm_is_ready = false;
-
 static struct pcm_sink* sinks[1] = {
     [PCM_SINK_BUILTIN] = &builtin_pcm_sink,
 };
@@ -142,7 +139,7 @@ void pcm_play_stop_int(void)
 
 static void pcm_wait_for_init(void)
 {
-    while (!pcm_is_ready)
+    while (!sinks[cur_sink]->pcm_is_ready)
         sleep(0);
 }
 
@@ -205,7 +202,7 @@ void pcm_do_peak_calculation(struct pcm_peaks *peaks, bool active,
     if (active)
     {
         struct pcm_sink* sink = sinks[cur_sink];
-        if (sink->configured_freq == -1UL)
+        if (sink->configured_freq == -1U)
         {
             logf("not configured yet");
             return;
@@ -251,9 +248,11 @@ void pcm_init(void)
     logf("pcm_init");
 
     for(size_t i = 0; i < ARRAYLEN(sinks); i += 1) {
-        sinks[i]->pending_freq = sinks[i]->caps.default_freq;
+        struct pcm_sink* sink = sinks[i];
-        sinks[i]->configured_freq = -1UL;
+        sink->pending_freq = sink->caps.default_freq;
-        sinks[i]->ops.init();
+        sink->configured_freq = -1U;
+        sink->pcm_is_ready = false;
+        sink->ops.init();
     }
 }
 
@@ -263,15 +262,18 @@ void pcm_postinit(void)
     logf("pcm_postinit");
 
     for(size_t i = 0; i < ARRAYLEN(sinks); i += 1) {
-        sinks[i]->ops.postinit();
+        struct pcm_sink* sink = sinks[i];
+        sink->ops.postinit();
+        sink->pcm_is_ready = true;
     }
 
-    pcm_is_ready = true;
+    /* Ensure mixer is in a sane state */
+    mixer_set_frequency(pcm_get_frequency());
 }
 
 bool pcm_is_initialized(void)
 {
-    return pcm_is_ready;
+    return sinks[cur_sink]->pcm_is_ready;
 }
 
 enum pcm_sink_ids pcm_current_sink(void)

firmware/pcm_mixer.c

@@ -33,7 +33,7 @@
    before the last samples are sent to the codec and so things are done in
    parallel (as much as possible) with sending-out data. */
 
-static unsigned int mixer_sampr = HW_SAMPR_DEFAULT;
+static unsigned int mixer_sampr = -1U;
 static unsigned int mix_frame_size = MIX_FRAME_SAMPLES*4;
 
 /* Define this to nonzero to add a marker pulse at each frame start */
@@ -266,7 +266,10 @@ static void mixer_start_pcm(void)
 #endif
 
     /* Requires a shared global sample rate for all channels */
-    pcm_set_frequency(mixer_sampr);
+    if (mixer_sampr == -1U)
+        mixer_sampr = pcm_get_frequency();
+    else
+        pcm_set_frequency(mixer_sampr);
 
     /* Prepare initial frames and set up the double buffer */
     mixer_buffer_callback(PCM_DMAST_STARTED);

firmware/sound.c

@@ -313,6 +313,16 @@ void sound_set_volume(int value)
         return;
 
 #ifndef BOOTLOADER
+    /* Apply limits */
+    const int min_vol = sound_min(SOUND_VOLUME);
+    const int max_vol = sound_max(SOUND_VOLUME);
+    if (value < min_vol)
+        value = min_vol;
+    if (value > max_vol)
+        value = max_vol;
+    if (value > global_settings.volume_limit)
+        value = global_settings.volume_limit;
+
     global_status.volume = value;
 #endif
 

firmware/target/arm/as3525/pcm-as3525.c

@@ -189,7 +189,7 @@ static void sink_dma_init(void)
     I2SOUT_CONTROL = (1<<6) | (1<<3);  /* enable dma, stereo */
 
     audiohw_preinit();
-    sink_set_freq(HW_SAMPR_DEFAULT);
+    sink_set_freq(builtin_pcm_sink.caps.default_freq);
 }
 
 #ifdef HAVE_PCM_DMA_ADDRESS

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

@@ -89,8 +89,6 @@ static bool swd_menu(void)
 #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
 {
@@ -102,8 +100,6 @@ 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)

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

@@ -36,19 +36,16 @@
 # 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"
@@ -67,8 +64,6 @@ 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)
 {
@@ -105,7 +100,7 @@ static void sai_init(void)
     /* 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 */
+                PBURST_V(SINGLE),       /* (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 */
@@ -117,10 +112,8 @@ static void sai_init(void)
                 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 */
+                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 */
 
@@ -146,7 +139,7 @@ static void sai_init(void)
                 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) */
+                FTH_V(FULL));                   /* 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 */
@@ -336,22 +329,12 @@ void dma1_ch0_irq_handler(void)
     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));
+        pcm_dma_teif_count++;
-        reg_writelf(dma1_ch0, DMA_CHN_CR, EN(0));
+        reg_assignlf(dma1, DMA_LIFCR, TEIF0(1));
 
-        pcm_play_dma_status_callback(PCM_DMAST_ERR_DMA);
+        /* TODO: this case may need additional handling */
+        pcm_play_dma_complete_callback(PCM_DMAST_ERR_DMA, NULL, NULL);
     }
     else if (reg_vreadf(lisr, DMA_LISR, TCIF0))
     {

firmware/target/arm/usb-drv-arc.c

@@ -362,6 +362,7 @@ static const unsigned int pipe2mask[USB_NUM_ENDPOINTS*2] = {
 /*-------------------------------------------------------------------------*/
 static void transfer_completed(void);
 static void control_received(void);
+static void sof_received(void);
 static int prime_transfer(int ep_num, void* ptr, int len, bool send, bool wait);
 static void prepare_td(struct transfer_descriptor* td,
         struct transfer_descriptor* previous_td, void *ptr, int len,int pipe);
@@ -415,6 +416,14 @@ static void usb_drv_reset(void)
 #endif
 }
 
+static void td_set_buf_ptr(struct transfer_descriptor* td, const void* ptr){
+    td->buff_ptr0 = (unsigned int)ptr;
+    td->buff_ptr1 = ((unsigned int)ptr & 0xfffff000) + 0x1000;
+    td->buff_ptr2 = ((unsigned int)ptr & 0xfffff000) + 0x2000;
+    td->buff_ptr3 = ((unsigned int)ptr & 0xfffff000) + 0x3000;
+    td->buff_ptr4 = ((unsigned int)ptr & 0xfffff000) + 0x4000;
+}
+
 /* One-time driver startup init */
 void usb_drv_startup(void)
 {
@@ -544,6 +553,12 @@ void usb_drv_int(void)
     if (status & USBSTS_PORT_CHANGE) {
         REG_USBSTS = USBSTS_PORT_CHANGE;
     }
+
+    /* sof */
+    if (status & USBSTS_SOF) {
+        REG_USBSTS = USBSTS_SOF;
+        sof_received();
+    }
 }
 
 bool usb_drv_stalled(int endpoint,bool in)
@@ -664,6 +679,183 @@ void usb_drv_set_test_mode(int mode)
     REG_USBCMD |= USBCMD_RUN;
 }
 
+/* batched request api  */
+static struct transfer_descriptor batch_td_array[USB_BATCH_SLOTS] USB_DEVBSS_ATTR __attribute__((aligned(32)));
+
+static uint8_t batch_ep = 0;
+static uint8_t batch_write_cursor;
+static bool batch_stopped;
+static usb_drv_batch_get_more batch_get_more;
+
+static int ep_to_pipe_index(uint8_t ep) {
+    const int ep_num = EP_NUM(ep);
+    const int ep_dir = EP_DIR(ep);
+    const int pipe = ep_num * 2 + (ep_dir == DIR_IN ? 1 : 0);
+    return pipe;
+}
+
+int usb_drv_batch_init(int ep, usb_drv_batch_get_more get_more) {
+    logf("batch init");
+    if(batch_ep != 0) {
+        logf("batch function not available");
+        return -1;
+    }
+    batch_ep = ep;
+    batch_get_more = get_more;
+    return 0;
+}
+
+int usb_drv_batch_deinit() {
+    logf("batch deinit");
+    usb_drv_batch_stop();
+    batch_ep = 0;
+    return 0;
+}
+
+/* returns whether priming is needed */
+static bool batch_fill_tds(void) {
+    while(!(batch_td_array[batch_write_cursor].size_ioc_sts & DTD_STATUS_ACTIVE)) {
+        /* batch_get_more may call batch_stop() through:
+        *  - batch_get_more()
+        *   - pcm_play_dma_complete_callback()
+        *    - sink_stop()
+        *     - usb_drv_batch_stop() */
+        const void* ptr;
+        size_t len;
+        batch_get_more(&ptr, &len);
+        if(len == 0 || batch_stopped) {
+            return false;
+        }
+
+        struct transfer_descriptor* td = &batch_td_array[batch_write_cursor];
+        td_set_buf_ptr(td, ptr);
+        td->size_ioc_sts = (len << DTD_LENGTH_BIT_POS) | DTD_STATUS_ACTIVE;
+        batch_write_cursor = (batch_write_cursor + 1)  % USB_BATCH_SLOTS;
+    }
+    return true;
+}
+
+int usb_drv_batch_start(void) {
+    logf("batch start");
+
+    /* reset variables */
+    batch_write_cursor = 0;
+    batch_stopped = false;
+
+    /* chain tds */
+    memset(batch_td_array, 0, sizeof(struct transfer_descriptor) * USB_BATCH_SLOTS);
+    for(int i = 0; i < USB_BATCH_SLOTS; i += 1) {
+        struct transfer_descriptor* td = &batch_td_array[i];
+        td->next_td_ptr = (unsigned int)&batch_td_array[(i + 1) % USB_BATCH_SLOTS];
+    }
+
+    /* configure queue head */
+    const int pipe = ep_to_pipe_index(batch_ep);
+    struct queue_head* const qh = &qh_array[pipe];
+
+    qh->curr_dtd_ptr = (unsigned int)&batch_td_array[0]; /* or error check in sof_received may read random location */
+    qh->dtd.next_td_ptr = (unsigned int)&batch_td_array[0];
+    qh->dtd.size_ioc_sts = 0;
+
+    /* pull initial buffers */
+    batch_fill_tds();
+
+    /* monitor sof */
+    REG_USBINTR |= USBINTR_SOF_EN;
+    return 0;
+}
+
+int usb_drv_batch_stop(void) {
+    batch_stopped = true;
+
+    /* disable sof interrupt */
+    REG_USBINTR &= ~USBINTR_SOF_EN;
+
+    /* break the chain */
+    /* terminating qh->dtd.next_td_ptr is not reliable */
+    for(int i = 0; i < USB_BATCH_SLOTS; i += 1) {
+        struct transfer_descriptor* td = &batch_td_array[i];
+        td->next_td_ptr = DTD_NEXT_TERMINATE;
+    }
+
+    /* flush endpoint */
+    const int pipe = ep_to_pipe_index(batch_ep);
+    const unsigned int mask = pipe2mask[pipe];
+
+    REG_ENDPTFLUSH = mask;
+    while (REG_ENDPTFLUSH & mask);
+
+    return 0;
+}
+
+#if defined(LOGF_ENABLE) && defined(ROCKBOX_HAS_LOGF)
+void usb_drv_dump_regs(void) {
+    logf("==== register dump %ld ====", current_tick);
+    logf("USBSTS         0x%08X", REG_USBSTS);
+    logf("ENDPTSETUPSTAT 0x%08X", REG_ENDPTSETUPSTAT);
+    logf("ENDPTPRIME     0x%08X", REG_ENDPTPRIME);
+    logf("ENDPTFLUSH     0x%08X", REG_ENDPTFLUSH);
+    logf("ENDPTSTATUS    0x%08X", REG_ENDPTSTATUS);
+    logf("ENDPTCOMPLETE  0x%08X", REG_ENDPTCOMPLETE);
+    logf("ENDPTCTRL0     0x%08X", REG_ENDPTCTRL0);
+    logf("ENDPTCTRL1     0x%08X", REG_ENDPTCTRL1);
+    logf("ENDPTCTRL2     0x%08X", REG_ENDPTCTRL2);
+}
+
+static void dump_td_array(struct queue_head* qh, struct transfer_descriptor* tds, size_t size) {
+    void* current = (void*)qh->curr_dtd_ptr;
+    void* next = (void*)qh->dtd.next_td_ptr;
+
+    logf("==== td dump %ld n=%p c=%p ====", current_tick, next, current);
+    for(int i = 0; i < size; i += 1) {
+        int len = (tds[i].size_ioc_sts & DTD_PACKET_SIZE) >> DTD_LENGTH_BIT_POS;
+        char sts[] = {
+            &tds[i] == current ? 'c' : ' ',
+            &tds[i] == next ? 'n' : ' ',
+            i == batch_write_cursor ? 'w' : ' ',
+            '\0',
+        };
+        logf("%s td[%02d] status=0x%08X len=%d", sts, i, tds[i].size_ioc_sts, len);
+    }
+}
+
+void usb_drv_dump_tds(int ep) {
+    const int pipe = ep_to_pipe_index(ep);
+    struct queue_head* const qh = &qh_array[pipe];
+    dump_td_array(qh, &td_array[pipe * NUM_TDS_PER_EP], NUM_TDS_PER_EP);
+}
+
+void usb_drv_batch_dump_tds(void) {
+    const int pipe = ep_to_pipe_index(batch_ep);
+    struct queue_head* const qh = &qh_array[pipe];
+    dump_td_array(qh, batch_td_array, USB_BATCH_SLOTS);
+}
+#endif
+
+static void sof_received(void) {
+    if(batch_ep == 0) {
+        /* should not happen */
+        return;
+    }
+
+    /* error recovery */
+    const int pipe = ep_to_pipe_index(batch_ep);
+    const unsigned int mask = pipe2mask[pipe];
+    struct queue_head* const qh = &qh_array[pipe];
+    struct transfer_descriptor* const td = (void*)qh->curr_dtd_ptr;
+    if(td->size_ioc_sts & DTD_ERROR_MASK) {
+        logf("td error status=0x%08X", td->size_ioc_sts);
+        REG_ENDPTPRIME |= mask;
+    }
+
+    /* do refill */
+    if(batch_fill_tds()) {
+        const int pipe = ep_to_pipe_index(batch_ep);
+        const unsigned int mask = pipe2mask[pipe];
+        REG_ENDPTPRIME |= mask;
+    }
+}
+
 /*-------------------------------------------------------------------------*/
 
 /* manual: 32.14.5.2 */
@@ -854,11 +1046,7 @@ static void prepare_td(struct transfer_descriptor* td,
     td->next_td_ptr = DTD_NEXT_TERMINATE;
     td->size_ioc_sts = (len<< DTD_LENGTH_BIT_POS) |
         DTD_STATUS_ACTIVE | DTD_IOC;
-    td->buff_ptr0 = (unsigned int)ptr;
+    td_set_buf_ptr(td, ptr);
-    td->buff_ptr1 = ((unsigned int)ptr & 0xfffff000) + 0x1000;
-    td->buff_ptr2 = ((unsigned int)ptr & 0xfffff000) + 0x2000;
-    td->buff_ptr3 = ((unsigned int)ptr & 0xfffff000) + 0x3000;
-    td->buff_ptr4 = ((unsigned int)ptr & 0xfffff000) + 0x4000;
     td->reserved |= DTD_RESERVED_LENGTH_MASK & len;
     td->reserved |= DTD_RESERVED_IN_USE;
     td->reserved |= (pipe << DTD_RESERVED_PIPE_OFFSET);

firmware/usbstack/usb_core.c

@@ -554,6 +554,64 @@ void usb_core_hotswap_event(int volume, bool inserted)
 }
 #endif
 
+
+#ifdef USB_BATCH_NON_NATIVE
+static uint8_t batch_ep = 0;
+static bool batch_stopped;
+static usb_drv_batch_get_more batch_get_more;
+
+int usb_drv_batch_init(int ep, usb_drv_batch_get_more get_more)
+{
+    if(batch_ep != 0) {
+        logf("usb_core: batch api in use user=0x%02X", batch_ep);
+        return -1;
+    }
+    batch_ep = ep;
+    batch_get_more = get_more;
+    batch_stopped  = false;
+    return 0;
+}
+
+int usb_drv_batch_deinit(void)
+{
+    logf("usb_core: batch deinit");
+    usb_drv_batch_stop();
+    batch_ep = 0;
+    return 0;
+}
+
+static int batch_get_and_send(void)
+{
+    const void* ptr;
+    size_t len;
+    batch_get_more(&ptr, &len);
+    if(len == 0 || batch_stopped) {
+        return 0;
+    }
+    return usb_drv_send_nonblocking(EP_NUM(batch_ep), (void*)ptr, len);
+}
+
+int usb_drv_batch_start(void)
+{
+    logf("usb_core: batch start");
+    batch_stopped = false;
+    return batch_get_and_send();
+}
+
+int usb_drv_batch_stop(void)
+{
+    batch_stopped = true;
+    return 0;
+}
+
+static void batch_xfer_complete(void) {
+    if(batch_stopped) {
+        return;
+    }
+    batch_get_and_send();
+}
+#endif
+
 static void usb_core_set_serial_function_id(void)
 {
     int i, id = 0;
@@ -1130,8 +1188,14 @@ void usb_core_bus_reset(void)
 /* called by usb_drv_transfer_completed() */
 void usb_core_transfer_complete(int endpoint, int dir, int status, int length)
 {
-    struct usb_transfer_completion_event_data* completion_event =
+#ifdef USB_BATCH_NON_NATIVE
-        &ep_data[endpoint].completion_event[EP_DIR(dir)];
+    /* batch api */
+    if(batch_ep != 0 && (endpoint | dir) == batch_ep) {
+        batch_xfer_complete();
+        return;
+    }
+#endif
+
     /* Fast notification */
     fast_completion_handler_t handler = ep_data[endpoint].fast_completion_handler[EP_DIR(dir)];
     if(handler != NULL && handler(endpoint, dir, status, length))
@@ -1154,6 +1218,9 @@ void usb_core_transfer_complete(int endpoint, int dir, int status, int length)
     }
 #endif
 
+    struct usb_transfer_completion_event_data* completion_event =
+        &ep_data[endpoint].completion_event[EP_DIR(dir)];
+
     completion_event->endpoint = endpoint;
     completion_event->dir = dir;
     completion_event->data[0] = data0;

tools/builds.pm

@@ -782,7 +782,7 @@ sub simbuilds {
     'norsk' => {
 	'lang' => 'norsk',
 	'name' => 'Norsk (Norwegian)',
-        'short' => 'no',
+        'short' => 'nb',
         'defengine' => 'piper',
 	'engines' => {
 	    'espeak' => '-vno',
@@ -791,6 +791,18 @@ sub simbuilds {
 	},
         'enabled' => 0,
     },
+    'norsk-nynorsk' => {
+	'lang' => 'norsk-nynorsk',
+	'name' => 'Nynorsk (Norwegian Nynorsk)',
+        'short' => 'nn',
+        'defengine' => 'piper',
+	'engines' => {
+	    'espeak' => '-vno',
+	    'gtts' => '-l no',
+            'piper' => 'no_NO-talesyntese-medium.onnx',
+	},
+        'enabled' => 1,
+    },
     'moldoveneste' => {
 	'lang' => 'moldoveneste',
 	'name' => 'Moldovenească (Moldavian)',