hosted pcm-alsa improvements

* xduoo x3ii/x20:  Better line out support
 * less granular volume settings (too many steps before)
 * Better handling of swiching sample rates
 * Log actual sample rate in debug menu

Most credit goes to Roman Stolyarov
Additional integration [re]work by myself

Change-Id: I63af3740678cf2ed3170f61534e1029c81826bb6

apps/debug_menu.c

@@ -2525,7 +2525,7 @@ static const struct {
         { "Screendump", dbg_screendump },
 #endif
         { "Skin Engine RAM usage", dbg_skin_engine },
-#if (CONFIG_PLATFORM & PLATFORM_NATIVE) || (defined(SONY_NWZ_LINUX) && !defined(SIMULATOR))
+#if (CONFIG_PLATFORM & PLATFORM_NATIVE) || defined(SONY_NWZ_LINUX) || defined(AGPTEK_ROCKER) || defined(XDUOO_X3II) || defined(XDUOO_X20) && !defined(SIMULATOR)
         { "View HW info", dbg_hw_info },
 #endif
 #if (CONFIG_PLATFORM & PLATFORM_NATIVE)

apps/playback.c

@@ -44,6 +44,7 @@
 #include "storage.h"
 #include "misc.h"
 #include "settings.h"
+#include "audiohw.h"
 
 #ifdef HAVE_TAGCACHE
 #include "tagcache.h"
@@ -3850,6 +3851,10 @@ static void audio_change_frequency_callback(unsigned short id, void *data)
     static bool starting_playback = false;
     struct mp3entry *id3;
 
+#ifdef AUDIOHW_HAVE_SET_OUTPUT
+    audiohw_set_output();
+#endif
+
     switch (id)
     {
     case PLAYBACK_EVENT_START_PLAYBACK:

firmware/drivers/audio/rocker_codec.c

@@ -29,6 +29,9 @@
 
 static int fd_hw;
 
+static long int vol_l_hw = 255;
+static long int vol_r_hw = 255;
+
 static void hw_open(void)
 {
     fd_hw = open("/dev/snd/controlC0", O_RDWR);
@@ -41,19 +44,32 @@ static void hw_close(void)
     close(fd_hw);
 }
 
+void audiohw_mute(int mute)
+{
+    if(mute)
+    {
+        long int ps0 = 0;
+        alsa_controls_set_ints("Output Port Switch", 1, &ps0);
+    }
+    else
+    {
+        long int ps2 = 2;
+        alsa_controls_set_ints("Output Port Switch", 1, &ps2);
+    }
+}
+
 void audiohw_preinit(void)
 {
-    long int hp = 2;
-
     alsa_controls_init();
     hw_open();
-
-    /* Output port switch set to Headphones */
-    alsa_controls_set_ints("Output Port Switch", 1, &hp);
 }
 
 void audiohw_postinit(void)
 {
+    long int hp = 2;
+
+    /* Output port switch set to Headphones */
+    alsa_controls_set_ints("Output Port Switch", 1, &hp);
 }
 
 void audiohw_close(void)
@@ -69,8 +85,8 @@ void audiohw_set_frequency(int fsel)
 
 void audiohw_set_volume(int vol_l, int vol_r)
 {
-    long int vol_l_hw = -vol_l/5;
+    vol_l_hw = -vol_l/5;
-    long int vol_r_hw = -vol_r/5;
+    vol_r_hw = -vol_r/5;
 
     alsa_controls_set_ints("Left Playback Volume", 1, &vol_l_hw);
     alsa_controls_set_ints("Right Playback Volume", 1, &vol_r_hw);

firmware/drivers/audio/xduoolinux_codec.c

@@ -29,9 +29,14 @@
 #include "panic.h"
 #include "sysfs.h"
 #include "alsa-controls.h"
+#include "pcm-alsa.h"
 
 static int fd_hw;
 
+static long int vol_l_hw = 255;
+static long int vol_r_hw = 255;
+static long int last_ps = 0;
+
 static void hw_open(void)
 {
     fd_hw = open("/dev/snd/controlC0", O_RDWR);
@@ -44,44 +49,69 @@ static void hw_close(void)
     close(fd_hw);
 }
 
-void audiohw_preinit(void)
+void audiohw_mute(int mute)
 {
-    alsa_controls_init();
+    if(mute)
-    hw_open();
+    {
+#if defined(XDUOO_X3II)
+        alsa_controls_set_bool("AK4490 Soft Mute", true);
+#endif
+#if defined(XDUOO_X20)
+        long int ps0 = (last_ps > 1) ? 1 : 2;
+        alsa_controls_set_ints("Output Port Switch", 1, &ps0);
+#endif
+    }
+    else
+    {
+#if defined(XDUOO_X3II)
+        alsa_controls_set_bool("AK4490 Soft Mute", false);
+#endif
+#if defined(XDUOO_X20)
+        alsa_controls_set_ints("Output Port Switch", 1, &last_ps);
+#endif
+    }
 }
 
-void audiohw_postinit(void)
+void audiohw_set_output(void)
 {
     long int ps = 2; // headset
+
     int status = 0;
 
     const char * const sysfs_lo_switch = "/sys/class/switch/lineout/state";
     const char * const sysfs_hs_switch = "/sys/class/switch/headset/state";
-#ifdef XDUOO_X20
+#if defined(XDUOO_X20)
     const char * const sysfs_bal_switch = "/sys/class/switch/balance/state";
 #endif
 
-#if defined(XDUOO_X3II)
-    alsa_controls_set_bool("AK4490 Soft Mute", true);
-#endif
-
     sysfs_get_int(sysfs_lo_switch, &status);
     if (status) ps = 1; // lineout
 
     sysfs_get_int(sysfs_hs_switch, &status);
     if (status) ps = 2; // headset
 
-#ifdef XDUOO_X20
+#if defined(XDUOO_X20)
     sysfs_get_int(sysfs_bal_switch, &status);
     if (status) ps = 3; // balance
 #endif
 
+    if (last_ps != ps)
+    {
         /* Output port switch */
-    alsa_controls_set_ints("Output Port Switch", 1, &ps);
+        last_ps = ps;
+        alsa_controls_set_ints("Output Port Switch", 1, &last_ps);
+    }
+}
 
-#if defined(XDUOO_X3II)
+void audiohw_preinit(void)
-    alsa_controls_set_bool("AK4490 Soft Mute", false);
+{
-#endif
+    alsa_controls_init();
+    hw_open();
+}
+
+void audiohw_postinit(void)
+{
+    audiohw_set_output();
 }
 
 void audiohw_close(void)
@@ -97,8 +127,8 @@ void audiohw_set_frequency(int fsel)
 
 void audiohw_set_volume(int vol_l, int vol_r)
 {
-    long int vol_l_hw = -vol_l/5;
+    vol_l_hw = -vol_l/5;
-    long int vol_r_hw = -vol_r/5;
+    vol_r_hw = -vol_r/5;
  
     alsa_controls_set_ints("Left Playback Volume", 1, &vol_l_hw);
     alsa_controls_set_ints("Right Playback Volume", 1, &vol_r_hw);
@@ -106,15 +136,15 @@ void audiohw_set_volume(int vol_l, int vol_r)
 
 void audiohw_set_filter_roll_off(int value)
 {
-    /* 0 = fast (sharp);
+    /* 0 = Sharp; 
-       1 = slow;
+       1 = Slow;
-       2 = fast2
+       2 = Short Sharp
-       3 = slow2
+       3 = Short Slow */
-       4 = NOS ? */
-    long int value_hw = value;
 #if defined(XDUOO_X3II)
+    long int value_hw = value;
     alsa_controls_set_ints("AK4490 Digital Filter", 1, &value_hw);
 #elif defined(XDUOO_X20)
+    long int value_hw = value;
     alsa_controls_set_ints("ES9018_K2M Digital Filter", 1, &value_hw);
 #else
     (void)value;

firmware/export/audiohw.h

@@ -223,7 +223,7 @@ struct sound_settings_info
 #elif defined(HAVE_ROCKER_CODEC)
 #include "rocker_codec.h"
 #elif defined(HAVE_XDUOO_LINUX_CODEC)
-#include "rocker_codec.h"
+#include "xduoolinux_codec.h"
 #endif
 
 /* convert caps into defines */
@@ -452,6 +452,10 @@ void audiohw_set_volume(int vol_l, int vol_r);
 void audiohw_set_lineout_volume(int vol_l, int vol_r);
 #endif
 
+#ifdef AUDIOHW_HAVE_SET_OUTPUT
+void audiohw_set_output(void);
+#endif
+
 #ifndef AUDIOHW_HAVE_CLIPPING
 #if defined(AUDIOHW_HAVE_BASS) || defined(AUDIOHW_HAVE_TREBLE) \
     || defined(AUDIOHW_HAVE_EQ)

firmware/export/config/agptekrocker.h

@@ -75,10 +75,6 @@
 /* The number of bytes reserved for loadable plugins */
 #define PLUGIN_BUFFER_SIZE 0x100000
 
-
-
-#define HAVE_ROCKER_CODEC
-
 #define HAVE_HEADPHONE_DETECTION
 
 /* KeyPad configuration for plugins */
@@ -108,10 +104,7 @@
 #define CPU_FREQ            1008000000
 
 /* No special storage */
-#define CONFIG_STORAGE (STORAGE_HOSTFS)//|STORAGE_SD)
+#define CONFIG_STORAGE STORAGE_HOSTFS
-//#define MULTIDRIVE_DIR "/mnt/sd_0"
-//#define NUM_DRIVES 1
-//#define HAVE_HOTSWAP
 #define HAVE_STORAGE_FLUSH
 
 /* Battery */

firmware/export/config/xduoox20.h

@@ -63,18 +63,12 @@
 /* define this if you have a real-time clock */
 #define CONFIG_RTC APPLICATION
 
-/* Define if the device can wake from an RTC alarm */
-//#define HAVE_RTC_ALARM
-
 /* The number of bytes reserved for loadable codecs */
 #define CODEC_SIZE 0x80000
 
 /* The number of bytes reserved for loadable plugins */
 #define PLUGIN_BUFFER_SIZE 0x100000
 
-
-
-
 #define HAVE_HEADPHONE_DETECTION
 
 /* KeyPad configuration for plugins */
@@ -125,6 +119,8 @@
 /* HW codec is flexible */
 #define HW_SAMPR_CAPS SAMPR_CAP_ALL_192
 
+#define AUDIOHW_HAVE_SET_OUTPUT
+
 /* Battery */
 #define BATTERY_CAPACITY_DEFAULT 2400 /* default battery capacity */
 #define BATTERY_CAPACITY_MIN 2400  /* min. capacity selectable */

firmware/export/config/xduoox3ii.h

@@ -60,9 +60,6 @@
 /* define this if you have a real-time clock */
 #define CONFIG_RTC APPLICATION
 
-/* Define if the device can wake from an RTC alarm */
-//#define HAVE_RTC_ALARM
-
 /* The number of bytes reserved for loadable codecs */
 #define CODEC_SIZE 0x80000
 
@@ -119,6 +116,8 @@
 /* HW codec is flexible */
 #define HW_SAMPR_CAPS SAMPR_CAP_ALL_192
 
+#define AUDIOHW_HAVE_SET_OUTPUT
+
 /* Battery */
 #define BATTERY_CAPACITY_DEFAULT 2000 /* default battery capacity */
 #define BATTERY_CAPACITY_MIN 2000  /* min. capacity selectable */

firmware/export/rocker_codec.h

@@ -2,5 +2,7 @@
 #define __ROCKER_CODEC__
 
 #define AUDIOHW_CAPS 0
-AUDIOHW_SETTING(VOLUME,       "dB", 1,  5, -1020,  0, -300, )
+AUDIOHW_SETTING(VOLUME, "dB", 0, 1, -127,  0, -30)
 #endif
+
+void audiohw_mute(int mute);

firmware/export/system.h

@@ -347,7 +347,7 @@ static inline void cpu_boost_unlock(void)
 #ifndef SIMULATOR
 bool dbg_ports(void);
 #endif
-#if (CONFIG_PLATFORM & PLATFORM_NATIVE) || defined(SONY_NWZ_LINUX)
+#if (CONFIG_PLATFORM & PLATFORM_NATIVE) || defined(SONY_NWZ_LINUX) || defined(AGPTEK_ROCKER) || defined(XDUOO_X3II) || defined(XDUOO_X20)
 bool dbg_hw_info(void);
 #endif
 

firmware/export/xduoolinux_codec.h

@@ -5,3 +5,6 @@
 AUDIOHW_SETTING(VOLUME, "dB", 0, 1, -127,  0, -30)
 AUDIOHW_SETTING(FILTER_ROLL_OFF, "", 0, 1, 0, 4, 0)
 #endif
+
+void audiohw_mute(int mute);
+void audiohw_set_output(void);

firmware/target/hosted/agptek/debug-agptek.c

@@ -1,6 +1,52 @@
-#include <stdbool.h>
+/***************************************************************************
+ *             __________               __   ___.
+ *   Open      \______   \ ____   ____ |  | _\_ |__   _______  ___
+ *   Source     |       _//  _ \_/ ___\|  |/ /| __ \ /  _ \  \/  /
+ *   Jukebox    |    |   (  <_> )  \___|    < | \_\ (  <_> > <  <
+ *   Firmware   |____|_  /\____/ \___  >__|_ \|___  /\____/__/\_ \
+ *                     \/            \/     \/    \/            \/
+ *
+ * Copyright (C) 2020 by Solomon Peachy
+ *
+ * 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.
+ *
+ ****************************************************************************/
 
-bool debug_hw_info(void)
+#include "config.h"
+#include "font.h"
+#include "lcd.h"
+#include "kernel.h"
+#include "button.h"
+
+#ifndef BOOTLOADER
+
+#include "pcm-alsa.h"
+
+static int line = 0;
+
+bool dbg_hw_info(void)
 {
-    return false;
+    int btn = 0;
+
+    lcd_setfont(FONT_SYSFIXED);
+
+    while(btn ^ BUTTON_POWER) {
+        lcd_clear_display();
+        line = 0;
+
+        lcd_putsf(0, line++, "pcm srate: %d", pcm_alsa_get_rate());
+        btn = button_read_device();
+
+        lcd_update();
+        sleep(HZ/16);
+    }
+    return true;
 }
+
+#endif /* !BOOTLOADER */

firmware/target/hosted/pcm-alsa.c

@@ -68,7 +68,7 @@
  * with multple applications running */
 static char device[] = "plughw:0,0";                    /* playback device */
 static const snd_pcm_access_t access_ = SND_PCM_ACCESS_RW_INTERLEAVED; /* access mode */
-#ifdef SONY_NWZ_LINUX
+#if defined(SONY_NWZ_LINUX) || defined(HAVE_FIIO_LINUX_CODEC)
 /* Sony NWZ must use 32-bit per sample */
 static const snd_pcm_format_t format = SND_PCM_FORMAT_S32_LE;    /* sample format */
 typedef long sample_t;
@@ -77,6 +77,9 @@ static const snd_pcm_format_t format = SND_PCM_FORMAT_S16;    /* sample format *
 typedef short sample_t;
 #endif
 static const int channels = 2;                                /* count of channels */
+static unsigned int sample_rate = 0;
+static unsigned int real_sample_rate = 0;
+
 static snd_pcm_t *handle = NULL;
 static snd_pcm_sframes_t buffer_size = MIX_FRAME_SAMPLES * 32; /* ~16k */
 static snd_pcm_sframes_t period_size = MIX_FRAME_SAMPLES * 4;  /*  ~4k */
@@ -93,14 +96,13 @@ static char signal_stack[SIGSTKSZ];
 static int recursion;
 #endif
 
-static int set_hwparams(snd_pcm_t *handle, unsigned sample_rate)
+static int set_hwparams(snd_pcm_t *handle)
 {
-    unsigned int rrate;
     int err;
+    unsigned int srate;
     snd_pcm_hw_params_t *params;
     snd_pcm_hw_params_malloc(&params);
 
-
     /* choose all parameters */
     err = snd_pcm_hw_params_any(handle, params);
     if (err < 0)
@@ -130,16 +132,17 @@ static int set_hwparams(snd_pcm_t *handle, unsigned sample_rate)
         goto error;
     }
     /* set the stream rate */
-    rrate = sample_rate;
+    sample_rate = srate = pcm_sampr;
-    err = snd_pcm_hw_params_set_rate_near(handle, params, &rrate, 0);
+    err = snd_pcm_hw_params_set_rate_near(handle, params, &srate, 0);
     if (err < 0)
     {
         printf("Rate %iHz not available for playback: %s\n", sample_rate, snd_strerror(err));
         goto error;
     }
-    if (rrate != sample_rate)
+    real_sample_rate = srate;
+    if (real_sample_rate != sample_rate)
     {
-        printf("Rate doesn't match (requested %iHz, get %iHz)\n", sample_rate, rrate);
+        printf("Rate doesn't match (requested %iHz, get %iHz)\n", sample_rate, real_sample_rate);
         err = -EINVAL;
         goto error;
     }
@@ -159,8 +162,9 @@ static int set_hwparams(snd_pcm_t *handle, unsigned sample_rate)
         printf("Unable to set period size %ld for playback: %s\n", period_size, snd_strerror(err));
         goto error;
     }
-    if (!frames)
+
-        frames = malloc(period_size * channels * sizeof(sample_t));
+    free(frames);
+    frames = calloc(1, period_size * channels * sizeof(sample_t));
 
     /* write the parameters to device */
     err = snd_pcm_hw_params(handle, params);
@@ -229,26 +233,37 @@ error:
  * and add 48dB to the input volume. We cannot go lower -43dB because several
  * values between -48dB and -43dB would require a fractional multiplier, which is
  * stupid to implement for such very low volume. */
-static int dig_vol_mult = 2 ^ 16; /* multiplicative factor to apply to each sample */
+static int dig_vol_mult_l = 2 ^ 16; /* multiplicative factor to apply to each sample */
+static int dig_vol_mult_r = 2 ^ 16; /* multiplicative factor to apply to each sample */
 
-void pcm_alsa_set_digital_volume(int vol_db)
+void pcm_alsa_set_digital_volume(int vol_db_l, int vol_db_r)
 {
-    if(vol_db > 0 || vol_db < -43)
+    if(vol_db_l > 0 || vol_db_r > 0 || vol_db_l < -43 || vol_db_r < -43)
         panicf("invalid pcm alsa volume");
     if(format != SND_PCM_FORMAT_S32_LE)
         panicf("this function assumes 32-bit sample size");
-    vol_db += 48; /* -42dB .. 0dB => 5dB .. 48dB */
+    vol_db_l += 48; /* -42dB .. 0dB => 5dB .. 48dB */
+    vol_db_r += 48; /* -42dB .. 0dB => 5dB .. 48dB */
     /* NOTE if vol_dB = 5 then vol_shift = 1 but r = 1 so we do vol_shift - 1 >= 0
      * otherwise vol_dB >= 0 implies vol_shift >= 2 so vol_shift - 2 >= 0 */
-    int vol_shift = vol_db / 3;
+    int vol_shift_l = vol_db_l / 3;
-    int r = vol_db % 3;
+    int vol_shift_r = vol_db_r / 3;
-    if(r == 0)
+    int r_l = vol_db_l % 3;
-        dig_vol_mult = 1 << vol_shift;
+    int r_r = vol_db_r % 3;
-    else if(r == 1)
+    if(r_l == 0)
-        dig_vol_mult = 1 << vol_shift | 1 << (vol_shift - 2);
+        dig_vol_mult_l = 1 << vol_shift_l;
+    else if(r_l == 1)
+        dig_vol_mult_l = 1 << vol_shift_l | 1 << (vol_shift_l - 2);
     else
-        dig_vol_mult = 1 << vol_shift | 1 << (vol_shift - 1);
+        dig_vol_mult_l = 1 << vol_shift_l | 1 << (vol_shift_l - 1);
-    printf("%d dB -> factor = %d\n", vol_db - 48, dig_vol_mult);
+    printf("l: %d dB -> factor = %d\n", vol_db_l - 48, dig_vol_mult_l);
+    if(r_r == 0)
+        dig_vol_mult_r = 1 << vol_shift_r;
+    else if(r_r == 1)
+        dig_vol_mult_r = 1 << vol_shift_r | 1 << (vol_shift_r - 2);
+    else
+        dig_vol_mult_r = 1 << vol_shift_r | 1 << (vol_shift_r - 1);
+    printf("r: %d dB -> factor = %d\n", vol_db_r - 48, dig_vol_mult_r);
 }
 
 /* copy pcm samples to a spare buffer, suitable for snd_pcm_writei() */
@@ -279,8 +294,11 @@ static bool fill_frames(void)
              * sample by some value so the sound is not too low */
             const short *pcm_ptr = pcm_data;
             sample_t *sample_ptr = &frames[2*(period_size-frames_left)];
-            for (int i = 0; i < copy_n*2; i++)
+            for (int i = 0; i < copy_n; i++)
-                *sample_ptr++ = *pcm_ptr++ * dig_vol_mult;
+            {
+                *sample_ptr++ = *pcm_ptr++ * dig_vol_mult_l;
+                *sample_ptr++ = *pcm_ptr++ * dig_vol_mult_r;
+            }
         }
         else
         {
@@ -378,7 +396,7 @@ static int async_rw(snd_pcm_t *handle)
 
     /* fill buffer with silence to initiate playback without noisy click */
     sample_size = buffer_size;
-    samples = malloc(sample_size * channels * sizeof(sample_t));
+    samples = calloc(1, sample_size * channels * sizeof(sample_t));
 
     snd_pcm_format_set_silence(format, samples, sample_size);
     err = snd_pcm_writei(handle, samples, sample_size);
@@ -428,7 +446,7 @@ void pcm_play_dma_init(void)
     if ((err = snd_pcm_nonblock(handle, 1)))
         panicf("Could not set non-block mode: %s\n", snd_strerror(err));
 
-    if ((err = set_hwparams(handle, pcm_sampr)) < 0)
+    if ((err = set_hwparams(handle)) < 0)
     {
         panicf("Setting of hwparams failed: %s\n", snd_strerror(err));
     }
@@ -473,15 +491,28 @@ void pcm_play_unlock(void)
 #endif
 }
 
+#if defined(HAVE_XDUOO_LINUX_CODEC) || defined(HAVE_FIIO_LINUX_CODEC) || defined(HAVE_ROCKER_CODEC)
+static void pcm_dma_apply_settings_nolock(void)
+{
+    if (sample_rate != pcm_sampr)
+    {
+	audiohw_mute(true);
+        snd_pcm_drop(handle);
+        set_hwparams(handle);
+	audiohw_mute(false);
+    }
+}
+#else
 static void pcm_dma_apply_settings_nolock(void)
 {
     snd_pcm_drop(handle);
-    set_hwparams(handle, pcm_sampr);
+    set_hwparams(handle);
 #if defined(HAVE_NWZ_LINUX_CODEC)
     /* Sony NWZ linux driver uses a nonstandard mecanism to set the sampling rate */
     audiohw_set_frequency(pcm_sampr);
 #endif
 }
+#endif
 
 void pcm_dma_apply_settings(void)
 {
@@ -571,11 +602,16 @@ void pcm_play_dma_postinit(void)
     audiohw_postinit();
 }
 
-
 void pcm_set_mixer_volume(int volume)
 {
     (void)volume;
 }
+
+int pcm_alsa_get_rate(void)
+{
+    return real_sample_rate;
+}
+
 #ifdef HAVE_RECORDING
 void pcm_rec_lock(void)
 {

firmware/target/hosted/pcm-alsa.h

@@ -22,10 +22,12 @@
 
 #include <config.h>
 
-#ifdef SONY_NWZ_LINUX
+#if defined(SONY_NWZ_LINUX) || defined(HAVE_FIIO_LINUX_CODEC)
 /* Set the PCM volume in dB: each sample with have this volume applied digitally
  * before being sent to ALSA. Volume must satisfy -43 <= dB <= 0 */
-void pcm_alsa_set_digital_volume(int vol_db);
+void pcm_alsa_set_digital_volume(int vol_db_l, int vol_db_r);
 #endif
 
+int pcm_alsa_get_rate(void);
+
 #endif /* __PCM_ALSA_RB_H__ */

firmware/target/hosted/xduoo/debug-xduoo.c

@@ -1,6 +1 @@
-#include <stdbool.h>
+#include "../agptek/debug-agptek.c"
-
-bool debug_hw_info(void)
-{
-    return false;
-}