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authorMichael Sevakis <jethead71@rockbox.org>2013-05-23 13:58:51 -0400
committerMichael Sevakis <jethead71@rockbox.org>2013-07-06 04:22:04 +0200
commitd37bf24d9011addbfbd40942a4e9bbf26de7df00 (patch)
treedafb7eaeb494081668a4841d490fce2bfbb2438d
parent00faabef5e902008172e08d3bcd77683cbafef51 (diff)
downloadrockbox-d37bf24d9011addbfbd40942a4e9bbf26de7df00.tar.gz
rockbox-d37bf24d9011addbfbd40942a4e9bbf26de7df00.tar.bz2
rockbox-d37bf24d9011addbfbd40942a4e9bbf26de7df00.zip
Enable setting of global output samplerate on certain targets.
Replaces the NATIVE_FREQUENCY constant with a configurable frequency. The user may select 48000Hz if the hardware supports it. The default is still 44100Hz and the minimum is 44100Hz. The setting is located in the playback settings, under "Frequency". "Frequency" was duplicated in english.lang for now to avoid having to fix every .lang file for the moment and throwing everything out of sync because of the new play_frequency feature in features.txt. The next cleanup should combine it with the one included for recording and generalize the ID label. If the hardware doesn't support 48000Hz, no setting will be available. On particular hardware where very high rates are practical and desireable, the upper bound can be extended by patching. The PCM mixer can be configured to play at the full hardware frequency range. The DSP core can configure to the hardware minimum up to the maximum playback setting (some buffers must be reserved according to the maximum rate). If only 44100Hz is supported or possible on a given target for playback, using the DSP and mixer at other samperates is possible if the hardware offers them. Change-Id: I6023cf0c0baa8bc6292b6919b4dd3618a6a25622 Reviewed-on: http://gerrit.rockbox.org/479 Reviewed-by: Michael Sevakis <jethead71@rockbox.org> Tested-by: Michael Sevakis <jethead71@rockbox.org>
-rw-r--r--apps/beep.c13
-rw-r--r--apps/codec_thread.c1
-rw-r--r--apps/features.txt4
-rw-r--r--apps/lang/english.lang17
-rw-r--r--apps/menus/playback_menu.c27
-rw-r--r--apps/pcmbuf.c21
-rw-r--r--apps/pcmbuf.h2
-rw-r--r--apps/playback.c11
-rw-r--r--apps/plugin.c2
-rw-r--r--apps/plugin.h4
-rw-r--r--apps/plugins/SOURCES2
-rw-r--r--apps/plugins/mpegplayer/audio_thread.c1
-rw-r--r--apps/plugins/mpegplayer/mpegplayer.h2
-rw-r--r--apps/plugins/mpegplayer/pcm_output.c6
-rw-r--r--apps/plugins/oscilloscope.c19
-rw-r--r--apps/plugins/test_codec.c7
-rw-r--r--apps/rbcodecconfig.h4
-rw-r--r--apps/settings.c38
-rw-r--r--apps/settings.h7
-rw-r--r--apps/settings_list.c5
-rw-r--r--apps/voice_thread.c11
-rw-r--r--firmware/export/config_caps.h34
-rw-r--r--firmware/export/pcm.h3
-rw-r--r--firmware/export/pcm_mixer.h6
-rw-r--r--firmware/export/pcm_sampr.h56
-rw-r--r--firmware/pcm.c6
-rw-r--r--firmware/pcm_mixer.c27
-rw-r--r--lib/rbcodec/dsp/compressor.c18
-rw-r--r--lib/rbcodec/dsp/crossfeed.c117
-rw-r--r--lib/rbcodec/dsp/dsp_arm.S84
-rw-r--r--lib/rbcodec/dsp/dsp_cf.S102
-rw-r--r--lib/rbcodec/dsp/dsp_core.c29
-rw-r--r--lib/rbcodec/dsp/dsp_core.h4
-rw-r--r--lib/rbcodec/dsp/dsp_misc.c15
-rw-r--r--lib/rbcodec/dsp/dsp_misc.h7
-rw-r--r--lib/rbcodec/dsp/dsp_proc_database.h2
-rw-r--r--lib/rbcodec/dsp/dsp_proc_entry.h4
-rw-r--r--lib/rbcodec/dsp/dsp_sample_input.c30
-rw-r--r--lib/rbcodec/dsp/dsp_sample_io.h5
-rw-r--r--lib/rbcodec/dsp/eq.c77
-rw-r--r--lib/rbcodec/dsp/resample.c41
-rw-r--r--lib/rbcodec/dsp/tone_controls.c29
-rw-r--r--lib/rbcodec/rbcodecconfig-example.h4
-rw-r--r--lib/rbcodec/test/warble.c5
44 files changed, 678 insertions, 231 deletions
diff --git a/apps/beep.c b/apps/beep.c
index 8ac7ccf224..25b5e0e391 100644
--- a/apps/beep.c
+++ b/apps/beep.c
@@ -21,10 +21,10 @@
#include "config.h"
#include "system.h"
#include "settings.h"
-#include "dsp_core.h" /* for NATIVE_FREQUENCY */
#include "pcm.h"
#include "pcm_mixer.h"
#include "misc.h"
+#include "fixedpoint.h"
/** Beep generation, CPU optimized **/
#include "asm/beep.c"
@@ -39,8 +39,10 @@ static uint32_t beep_amplitude; /* Amplitude of square wave generator */
#endif
static int beep_count; /* Number of samples remaining to generate */
-/* Reserve enough static space for keyclick to fit */
-#define BEEP_BUF_COUNT (NATIVE_FREQUENCY / 1000 * KEYCLICK_DURATION)
+#define BEEP_COUNT(fs, duration) ((fs) / 1000 * (duration))
+
+/* Reserve enough static space for keyclick to fit in worst case */
+#define BEEP_BUF_COUNT BEEP_COUNT(PLAY_SAMPR_MAX, KEYCLICK_DURATION)
static int16_t beep_buf[BEEP_BUF_COUNT*2] IBSS_ATTR __attribute__((aligned(4)));
/* Callback to generate the beep frames - also don't want inlining of
@@ -75,9 +77,10 @@ void beep_play(unsigned int frequency, unsigned int duration,
amplitude = INT16_MAX;
/* Setup the parameters for the square wave generator */
+ uint32_t fout = mixer_get_frequency();
beep_phase = 0;
- beep_step = 0xffffffffu / NATIVE_FREQUENCY * frequency;
- beep_count = NATIVE_FREQUENCY / 1000 * duration;
+ beep_step = fp_div(frequency, fout, 32);
+ beep_count = BEEP_COUNT(fout, duration);
#ifdef BEEP_GENERIC
beep_amplitude = amplitude;
diff --git a/apps/codec_thread.c b/apps/codec_thread.c
index d4b1c64573..8f9f5a3c74 100644
--- a/apps/codec_thread.c
+++ b/apps/codec_thread.c
@@ -507,6 +507,7 @@ static void run_codec(void)
codec_queue_ack(Q_CODEC_RUN);
trigger_cpu_boost();
+ dsp_configure(ci.dsp, DSP_SET_OUT_FREQUENCY, pcmbuf_get_frequency());
if (!encoder)
{
diff --git a/apps/features.txt b/apps/features.txt
index a65744fd5a..897657f52c 100644
--- a/apps/features.txt
+++ b/apps/features.txt
@@ -274,3 +274,7 @@ lowmem
#if defined(HAVE_HARDWARE_CLICK)
hardware_click
#endif
+
+#if defined(HAVE_PLAY_FREQ)
+play_frequency
+#endif
diff --git a/apps/lang/english.lang b/apps/lang/english.lang
index dbd0baa18a..0dd3b43f5a 100644
--- a/apps/lang/english.lang
+++ b/apps/lang/english.lang
@@ -13156,3 +13156,20 @@
*: "Slow"
</voice>
</phrase>
+<phrase>
+ id: LANG_PLAYBACK_FREQUENCY
+ desc: in playback settings (merge with LANG_RECORDING_FREQUENCY if cleaning)
+ user: core
+ <source>
+ *: none
+ play_frequency: "Frequency"
+ </source>
+ <dest>
+ *: none
+ play_frequency: "Frequency"
+ </dest>
+ <voice>
+ *: none
+ play_frequency: "Frequency"
+ </voice>
+</phrase>
diff --git a/apps/menus/playback_menu.c b/apps/menus/playback_menu.c
index 6beda93991..89472d45b9 100644
--- a/apps/menus/playback_menu.c
+++ b/apps/menus/playback_menu.c
@@ -37,6 +37,10 @@
#include "misc.h"
#if CONFIG_CODEC == SWCODEC
#include "playback.h"
+#include "pcm_sampr.h"
+#ifdef HAVE_PLAY_FREQ
+#include "talk.h"
+#endif
#endif
@@ -192,6 +196,10 @@ MENUITEM_SETTING(prevent_skip, &global_settings.prevent_skip, NULL);
MENUITEM_SETTING(resume_rewind, &global_settings.resume_rewind, NULL);
#endif
MENUITEM_SETTING(pause_rewind, &global_settings.pause_rewind, NULL);
+#ifdef HAVE_PLAY_FREQ
+MENUITEM_SETTING(play_frequency, &global_settings.play_frequency,
+ playback_callback);
+#endif
MAKE_MENU(playback_settings,ID2P(LANG_PLAYBACK),0,
Icon_Playback_menu,
@@ -217,12 +225,15 @@ MAKE_MENU(playback_settings,ID2P(LANG_PLAYBACK),0,
#ifdef HAVE_HEADPHONE_DETECTION
,&unplug_menu
#endif
- ,&skip_length, &prevent_skip,
+ ,&skip_length, &prevent_skip
#if CONFIG_CODEC == SWCODEC
- &resume_rewind,
+ ,&resume_rewind
+#endif
+ ,&pause_rewind
+#ifdef HAVE_PLAY_FREQ
+ ,&play_frequency
#endif
- &pause_rewind,
);
static int playback_callback(int action,const struct menu_item_ex *this_item)
@@ -243,9 +254,19 @@ static int playback_callback(int action,const struct menu_item_ex *this_item)
break;
case ACTION_EXIT_MENUITEM: /* on exit */
+ /* Playing or not */
+#ifdef HAVE_PLAY_FREQ
+ if (this_item == &play_frequency)
+ {
+ settings_apply_play_freq(global_settings.play_frequency, false);
+ break;
+ }
+#endif /* HAVE_PLAY_FREQ */
+
if (!(audio_status() & AUDIO_STATUS_PLAY))
break;
+ /* Playing only */
if (this_item == &shuffle_item)
{
if (old_shuffle == global_settings.playlist_shuffle)
diff --git a/apps/pcmbuf.c b/apps/pcmbuf.c
index cc454a49ce..ff9b3e16a2 100644
--- a/apps/pcmbuf.c
+++ b/apps/pcmbuf.c
@@ -40,7 +40,6 @@
#include "settings.h"
#include "audio.h"
#include "voice_thread.h"
-#include "dsp_core.h"
/* This is the target fill size of chunks on the pcm buffer
Can be any number of samples but power of two sizes make for faster and
@@ -66,11 +65,11 @@
chunks */
/* Return data level in 1/4-second increments */
-#define DATA_LEVEL(quarter_secs) (NATIVE_FREQUENCY * (quarter_secs))
+#define DATA_LEVEL(quarter_secs) (pcmbuf_sampr * (quarter_secs))
/* Number of bytes played per second:
(sample rate * 2 channels * 2 bytes/sample) */
-#define BYTERATE (NATIVE_FREQUENCY * 4)
+#define BYTERATE (pcmbuf_sampr * 2 * 2)
#if MEMORYSIZE > 2
/* Keep watermark high for large memory target - at least (2s) */
@@ -104,6 +103,7 @@ static size_t pcmbuf_size;
static struct chunkdesc *pcmbuf_descriptors;
static unsigned int pcmbuf_desc_count;
static unsigned int position_key = 1;
+static unsigned int pcmbuf_sampr = 0;
static size_t chunk_ridx;
static size_t chunk_widx;
@@ -111,8 +111,7 @@ static size_t chunk_widx;
static size_t pcmbuf_bytes_waiting;
static struct chunkdesc *current_desc;
-/* Only written if HAVE_CROSSFADE */
-static size_t pcmbuf_watermark = PCMBUF_WATERMARK;
+static size_t pcmbuf_watermark = 0;
static bool low_latency_mode = false;
@@ -545,6 +544,8 @@ size_t pcmbuf_init(void *bufend)
}
pcmbuf_finish_crossfade_enable();
+#else
+ pcmbuf_watermark = PCMBUF_WATERMARK;
#endif /* HAVE_CROSSFADE */
init_buffer_state();
@@ -1331,3 +1332,13 @@ void pcmbuf_set_low_latency(bool state)
{
low_latency_mode = state;
}
+
+void pcmbuf_update_frequency(void)
+{
+ pcmbuf_sampr = mixer_get_frequency();
+}
+
+unsigned int pcmbuf_get_frequency(void)
+{
+ return pcmbuf_sampr;
+}
diff --git a/apps/pcmbuf.h b/apps/pcmbuf.h
index 7fa3563e6a..008872be59 100644
--- a/apps/pcmbuf.h
+++ b/apps/pcmbuf.h
@@ -81,5 +81,7 @@ void pcmbuf_sync_position_update(void);
/* Misc */
bool pcmbuf_is_lowdata(void);
void pcmbuf_set_low_latency(bool state);
+void pcmbuf_update_frequency(void);
+unsigned int pcmbuf_get_frequency(void);
#endif /* PCMBUF_H */
diff --git a/apps/playback.c b/apps/playback.c
index 24c268ffc4..8b498f265e 100644
--- a/apps/playback.c
+++ b/apps/playback.c
@@ -2028,8 +2028,11 @@ static int audio_fill_file_buffer(void)
/* Must reset the buffer before use if trashed or voice only - voice
file size shouldn't have changed so we can go straight from
AUDIOBUF_STATE_VOICED_ONLY to AUDIOBUF_STATE_INITIALIZED */
- if (buffer_state != AUDIOBUF_STATE_INITIALIZED)
+ if (buffer_state != AUDIOBUF_STATE_INITIALIZED ||
+ !pcmbuf_is_same_size())
+ {
audio_reset_buffer(AUDIOBUF_STATE_INITIALIZED);
+ }
logf("Starting buffer fill");
@@ -2510,6 +2513,11 @@ static void audio_start_playback(size_t offset, unsigned int flags)
#ifndef PLATFORM_HAS_VOLUME_CHANGE
sound_set_volume(global_settings.volume);
#endif
+#ifdef HAVE_PLAY_FREQ
+ settings_apply_play_freq(global_settings.play_frequency, true);
+#endif
+ pcmbuf_update_frequency();
+
/* Be sure channel is audible */
pcmbuf_fade(false, true);
@@ -3755,6 +3763,7 @@ void INIT_ATTR playback_init(void)
mutex_init(&id3_mutex);
track_list_init();
buffering_init();
+ pcmbuf_update_frequency();
add_event(PLAYBACK_EVENT_VOICE_PLAYING, false, playback_voice_event);
#ifdef HAVE_CROSSFADE
/* Set crossfade setting for next buffer init which should be about... */
diff --git a/apps/plugin.c b/apps/plugin.c
index 24443b58d9..a5cdfc3d8a 100644
--- a/apps/plugin.c
+++ b/apps/plugin.c
@@ -798,6 +798,8 @@ static const struct plugin_api rockbox_api = {
/* new stuff at the end, sort into place next time
the API gets incompatible */
+ mixer_set_frequency,
+ mixer_get_frequency,
};
int plugin_load(const char* plugin, const void* parameter)
diff --git a/apps/plugin.h b/apps/plugin.h
index 936f977fdc..f926b3428d 100644
--- a/apps/plugin.h
+++ b/apps/plugin.h
@@ -155,7 +155,7 @@ void* plugin_get_buffer(size_t *buffer_size);
#define PLUGIN_MAGIC 0x526F634B /* RocK */
/* increase this every time the api struct changes */
-#define PLUGIN_API_VERSION 223
+#define PLUGIN_API_VERSION 224
/* update this to latest version if a change to the api struct breaks
backwards compatibility (and please take the opportunity to sort in any
@@ -970,6 +970,8 @@ struct plugin_api {
/* new stuff at the end, sort into place next time
the API gets incompatible */
+ void (*mixer_set_frequency)(unsigned int samplerate);
+ unsigned int (*mixer_get_frequency)(void);
};
/* plugin header */
diff --git a/apps/plugins/SOURCES b/apps/plugins/SOURCES
index c512a9e02f..00bf9606d4 100644
--- a/apps/plugins/SOURCES
+++ b/apps/plugins/SOURCES
@@ -37,6 +37,8 @@ resistor.c
remote_control.c
#endif
+test_codec.c
+test_sampr.c
#ifdef HAVE_BACKLIGHT
diff --git a/apps/plugins/mpegplayer/audio_thread.c b/apps/plugins/mpegplayer/audio_thread.c
index 1c167ea2a2..764ad111f2 100644
--- a/apps/plugins/mpegplayer/audio_thread.c
+++ b/apps/plugins/mpegplayer/audio_thread.c
@@ -481,6 +481,7 @@ static void audio_thread(void)
init_mad();
td.dsp = rb->dsp_get_config(CODEC_IDX_AUDIO);
+ rb->dsp_configure(td.dsp, DSP_SET_OUT_FREQUENCY, CLOCK_RATE);
#ifdef HAVE_PITCHCONTROL
rb->sound_set_pitch(PITCH_SPEED_100);
rb->dsp_set_timestretch(PITCH_SPEED_100);
diff --git a/apps/plugins/mpegplayer/mpegplayer.h b/apps/plugins/mpegplayer/mpegplayer.h
index 32cc7b25be..4ddf0ca7b1 100644
--- a/apps/plugins/mpegplayer/mpegplayer.h
+++ b/apps/plugins/mpegplayer/mpegplayer.h
@@ -44,7 +44,7 @@
#define AUDIOBUF_ALLOC_SIZE (AUDIOBUF_SIZE+AUDIOBUF_GUARD_SIZE)
/** PCM buffer **/
-#define CLOCK_RATE NATIVE_FREQUENCY /* Our clock rate in ticks/second (samplerate) */
+#define CLOCK_RATE 44100 /* Our clock rate in ticks/second (samplerate) */
/* Define this as "1" to have a test tone instead of silence clip */
#define SILENCE_TEST_TONE 0
diff --git a/apps/plugins/mpegplayer/pcm_output.c b/apps/plugins/mpegplayer/pcm_output.c
index 3af8e91adc..82e3584277 100644
--- a/apps/plugins/mpegplayer/pcm_output.c
+++ b/apps/plugins/mpegplayer/pcm_output.c
@@ -51,6 +51,8 @@ static uint32_t volatile clock_time IBSS_ATTR; /* Timestamp adjusted */
static int pcm_skipped = 0;
static int pcm_underruns = 0;
+static unsigned int old_sampr = 0;
+
/* Small silence clip. ~5.80ms @ 44.1kHz */
static int16_t silence[256*2] ALIGNED_ATTR(4) = { 0 };
@@ -380,9 +382,13 @@ bool pcm_output_init(void)
}
#endif
+ old_sampr = rb->mixer_get_frequency();
+ rb->mixer_set_frequency(CLOCK_RATE);
return true;
}
void pcm_output_exit(void)
{
+ if (old_sampr != 0)
+ rb->mixer_set_frequency(old_sampr);
}
diff --git a/apps/plugins/oscilloscope.c b/apps/plugins/oscilloscope.c
index a4ec6a8789..4d807493d3 100644
--- a/apps/plugins/oscilloscope.c
+++ b/apps/plugins/oscilloscope.c
@@ -1200,13 +1200,14 @@ static long anim_peaks_vertical(void)
/** Waveform View **/
#ifdef OSCILLOSCOPE_GRAPHMODE
-static int16_t waveform_buffer[2*ALIGN_UP(NATIVE_FREQUENCY, 2048)+2*2048]
+static int16_t waveform_buffer[2*ALIGN_UP(PLAY_SAMPR_MAX, 2048)+2*2048]
MEM_ALIGN_ATTR;
static size_t waveform_buffer_threshold = 0;
static size_t volatile waveform_buffer_have = 0;
static size_t waveform_buffer_break = 0;
+static unsigned long mixer_sampr = PLAY_SAMPR_DEFAULT;
#define PCM_SAMPLESIZE (2*sizeof(int16_t))
-#define PCM_BYTERATE (NATIVE_FREQUENCY*PCM_SAMPLESIZE)
+#define PCM_BYTERATE(sampr) ((sampr)*PCM_SAMPLESIZE)
#define WAVEFORM_SCALE_PCM(full_scale, sample) \
((((full_scale) * (sample)) + (1 << 14)) >> 15)
@@ -1390,7 +1391,7 @@ static long anim_waveform_horizontal(void)
return cur_tick + HZ/5;
}
- int count = (NATIVE_FREQUENCY*osc_delay + 100*HZ - 1) / (100*HZ);
+ int count = (mixer_sampr*osc_delay + 100*HZ - 1) / (100*HZ);
waveform_buffer_set_threshold(count*PCM_SAMPLESIZE);
@@ -1516,7 +1517,8 @@ static long anim_waveform_horizontal(void)
osd_lcd_update();
long delay = get_next_delay();
- return cur_tick + delay - waveform_buffer_have * HZ / PCM_BYTERATE;
+ return cur_tick + delay - waveform_buffer_have * HZ /
+ PCM_BYTERATE(mixer_sampr);
}
static void anim_waveform_plot_filled_v(int y, int y_prev,
@@ -1583,7 +1585,7 @@ static long anim_waveform_vertical(void)
return cur_tick + HZ/5;
}
- int count = (NATIVE_FREQUENCY*osc_delay + 100*HZ - 1) / (100*HZ);
+ int count = (mixer_sampr*osc_delay + 100*HZ - 1) / (100*HZ);
waveform_buffer_set_threshold(count*PCM_SAMPLESIZE);
@@ -1709,7 +1711,8 @@ static long anim_waveform_vertical(void)
osd_lcd_update();
long delay = get_next_delay();
- return cur_tick + delay - waveform_buffer_have * HZ / PCM_BYTERATE;
+ return cur_tick + delay - waveform_buffer_have * HZ
+ / PCM_BYTERATE(mixer_sampr);
}
static void anim_waveform_exit(void)
@@ -1872,6 +1875,10 @@ static void osc_setup(void)
osd_lcd_update();
#endif
+#ifdef OSCILLOSCOPE_GRAPHMODE
+ mixer_sampr = rb->mixer_get_frequency();
+#endif
+
/* Turn off backlight timeout */
backlight_ignore_timeout();
graphmode_setup();
diff --git a/apps/plugins/test_codec.c b/apps/plugins/test_codec.c
index 7523d9e9aa..0b409f8e35 100644
--- a/apps/plugins/test_codec.c
+++ b/apps/plugins/test_codec.c
@@ -502,7 +502,12 @@ static void configure(int setting, intptr_t value)
{
case DSP_SET_FREQUENCY:
DEBUGF("samplerate=%d\n",(int)value);
- wavinfo.samplerate = use_dsp ? NATIVE_FREQUENCY : (int)value;
+ if (use_dsp) {
+ wavinfo.samplerate = rb->dsp_configure(
+ ci.dsp, DSP_GET_OUT_FREQUENCY, 0);
+ } else {
+ wavinfo.samplerate = (int)value;
+ }
break;
case DSP_SET_SAMPLE_DEPTH:
diff --git a/apps/rbcodecconfig.h b/apps/rbcodecconfig.h
index ff9fc41342..cc51595cc4 100644
--- a/apps/rbcodecconfig.h
+++ b/apps/rbcodecconfig.h
@@ -71,4 +71,8 @@ static inline void dsp_process_end(struct dsp_loop_context *ctx)
#endif
+#define DSP_OUT_MIN_HZ PLAY_SAMPR_HW_MIN
+#define DSP_OUT_MAX_HZ PLAY_SAMPR_MAX
+#define DSP_OUT_DEFAULT_HZ PLAY_SAMPR_DEFAULT
+
#endif
diff --git a/apps/settings.c b/apps/settings.c
index adc53cd14b..cf51b0793c 100644
--- a/apps/settings.c
+++ b/apps/settings.c
@@ -85,6 +85,11 @@ struct system_status global_status;
#ifdef HAVE_RECORDING
#include "enc_config.h"
#endif
+#include "pcm_sampr.h"
+#ifdef HAVE_PLAY_FREQ
+#include "pcm_mixer.h"
+#include "dsp_core.h"
+#endif
#endif /* CONFIG_CODEC == SWCODEC */
#define NVRAM_BLOCK_SIZE 44
@@ -720,6 +725,36 @@ void settings_apply_pm_range(void)
}
#endif /* HAVE_LCD_BITMAP */
+#ifdef HAVE_PLAY_FREQ
+void settings_apply_play_freq(int value, bool playback)
+{
+ static const unsigned long play_sampr[] = { SAMPR_44, SAMPR_48 };
+ static int prev_setting = 0;
+
+ if ((unsigned)value >= ARRAYLEN(play_sampr))
+ value = 0;
+
+ bool changed = value != prev_setting;
+ prev_setting = value;
+
+ long offset = 0;
+ bool playing = changed && !playback &&
+ audio_status() == AUDIO_STATUS_PLAY;
+
+ if (playing)
+ offset = audio_current_track()->offset;
+
+ if (changed && !playback)
+ audio_hard_stop();
+
+ /* Other sub-areas of playback pick it up from the mixer */
+ mixer_set_frequency(play_sampr[value]);
+
+ if (playing)
+ audio_play(offset);
+}
+#endif /* HAVE_PLAY_FREQ */
+
void sound_settings_apply(void)
{
#ifdef AUDIOHW_HAVE_BASS
@@ -976,6 +1011,9 @@ void settings_apply(bool read_disk)
set_codepage(global_settings.default_codepage);
CHART("<set_codepage");
+#ifdef HAVE_PLAY_FREQ
+ settings_apply_play_freq(global_settings.play_frequency, false);
+#endif
#if CONFIG_CODEC == SWCODEC
#ifdef HAVE_CROSSFADE
audio_set_crossfade(global_settings.crossfade);
diff --git a/apps/settings.h b/apps/settings.h
index 1aec931798..087ff0cb45 100644
--- a/apps/settings.h
+++ b/apps/settings.h
@@ -223,6 +223,9 @@ void settings_apply_skins(void);
void settings_apply(bool read_disk);
void settings_apply_pm_range(void);
+#ifdef HAVE_PLAY_FREQ
+void settings_apply_play_freq(int value, bool playback);
+#endif
void settings_display(void);
enum optiontype { INT, BOOL };
@@ -821,6 +824,10 @@ struct user_settings
#ifdef HAVE_QUICKSCREEN
bool shortcuts_replaces_qs;
#endif
+
+#ifdef HAVE_PLAY_FREQ
+ int play_frequency; /* core audio output frequency selection */
+#endif
};
/** global variables **/
diff --git a/apps/settings_list.c b/apps/settings_list.c
index c1b40a64b6..980c74f0a7 100644
--- a/apps/settings_list.c
+++ b/apps/settings_list.c
@@ -807,6 +807,11 @@ const struct settings_list settings[] = {
,ID2P(LANG_REPEAT_AB)
#endif
), /* CHOICE_SETTING( repeat_mode ) */
+#ifdef HAVE_PLAY_FREQ
+ STRINGCHOICE_SETTING(0, play_frequency, LANG_PLAYBACK_FREQUENCY, 0,
+ "playback frequency", "44.1 kHz,48 kHz", NULL, 2,
+ TALK_ID_DECIMAL(441, 1, UNIT_KHZ), TALK_ID(48, UNIT_KHZ)),
+#endif /* HAVE_PLAY_FREQ */
/* LCD */
#ifdef HAVE_LCD_CONTRAST
/* its easier to leave this one un-macro()ed for the time being */
diff --git a/apps/voice_thread.c b/apps/voice_thread.c
index 46471c0d9b..7788f659b0 100644
--- a/apps/voice_thread.c
+++ b/apps/voice_thread.c
@@ -18,7 +18,7 @@
* KIND, either express or implied.
*
****************************************************************************/
-#include <sys/types.h>
+#include "config.h"
#include "system.h"
#include "core_alloc.h"
#include "thread.h"
@@ -30,8 +30,7 @@
#include "pcm_mixer.h"
#include "codecs/libspeex/speex/speex.h"
-/* Default number of native-frequency PCM frames to queue - adjust as
- necessary per-target */
+/* Default number of PCM frames to queue - adjust as necessary per-target */
#define VOICE_FRAMES 4
/* Define any of these as "1" and uncomment the LOGF_ENABLE line to log
@@ -84,8 +83,8 @@ static struct queue_sender_list voice_queue_sender_list SHAREDBSS_ATTR;
static int quiet_counter SHAREDDATA_ATTR = 0;
static bool voice_playing = false;
-#define VOICE_PCM_FRAME_COUNT ((NATIVE_FREQUENCY*VOICE_FRAME_COUNT + \
- VOICE_SAMPLE_RATE) / VOICE_SAMPLE_RATE)
+#define VOICE_PCM_FRAME_COUNT ((PLAY_SAMPR_MAX*VOICE_FRAME_COUNT + \
+ VOICE_SAMPLE_RATE) / VOICE_SAMPLE_RATE)
#define VOICE_PCM_FRAME_SIZE (VOICE_PCM_FRAME_COUNT*2*sizeof (int16_t))
/* Voice processing states */
@@ -356,11 +355,13 @@ static enum voice_state voice_message(struct voice_thread_data *td)
{
/* Stop any clip still playing */
voice_stop_playback();
+ dsp_configure(td->dsp, DSP_FLUSH, 0);
}
if (quiet_counter <= 0)
{
voice_playing = true;
+ dsp_configure(td->dsp, DSP_SET_OUT_FREQUENCY, mixer_get_frequency());
send_event(PLAYBACK_EVENT_VOICE_PLAYING, &voice_playing);
}
diff --git a/firmware/export/config_caps.h b/firmware/export/config_caps.h
index fcb13debfc..bc0a42bedf 100644
--- a/firmware/export/config_caps.h
+++ b/firmware/export/config_caps.h
@@ -116,3 +116,37 @@
#endif
#endif /* HAVE_RECORDING */
+
+/* Samplerate config */
+#define PCM_SAMPR_CONFIG_ONLY /* no C code */
+#include "pcm_sampr.h"
+#undef PCM_SAMPR_CONFIG_ONLY
+
+#define PLAY_SAMPR_CAPS (HW_SAMPR_CAPS & (SAMPR_CAP_44 | SAMPR_CAP_48))
+/**
+ * PLAY_SAMPR_MIN: The minimum allowable samplerate for global playback.
+ * Music won't play at a lower rate.
+ * PLAY_SAMPR_MAX: The maximum allowable samplerate for global playback.
+ * Music won't play at a faster rate.
+ * PLAY_SAMPR_DEFAULT: The default samplerate, unless configured otherwise.
+ * PLAY_SAMPR_HW_MIN: The minimum allowable rate for some subsystems such
+ * as the DSP core. DSP never exceeds *MAX to lessen
+ * buffer allocation demands and overhead.
+ */
+#if PLAY_SAMPR_CAPS & (PLAY_SAMPR_CAPS - 1)
+#define HAVE_PLAY_FREQ
+# define PLAY_SAMPR_MIN SAMPR_44
+# define PLAY_SAMPR_MAX SAMPR_48
+# define PLAY_SAMPR_DEFAULT SAMPR_44
+# define PLAY_SAMPR_HW_MIN HW_SAMPR_MIN
+#elif PLAY_SAMPR_CAPS & SAMPR_CAP_44
+# define PLAY_SAMPR_MIN SAMPR_44
+# define PLAY_SAMPR_MAX SAMPR_44
+# define PLAY_SAMPR_DEFAULT SAMPR_44
+# define PLAY_SAMPR_HW_MIN HW_SAMPR_MIN
+#elif PLAY_SAMPR_CAPS & SAMPR_CAP_48
+# define PLAY_SAMPR_MIN SAMPR_48
+# define PLAY_SAMPR_MAX SAMPR_48
+# define PLAY_SAMPR_DEFAULT SAMPR_48
+# define PLAY_SAMPR_HW_MIN HW_SAMPR_MIN
+#endif
diff --git a/firmware/export/pcm.h b/firmware/export/pcm.h
index fdd46237a6..23c0bd4a0b 100644
--- a/firmware/export/pcm.h
+++ b/firmware/export/pcm.h
@@ -53,7 +53,10 @@ 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);
diff --git a/firmware/export/pcm_mixer.h b/firmware/export/pcm_mixer.h
index d424083002..f7f869eaaf 100644
--- a/firmware/export/pcm_mixer.h
+++ b/firmware/export/pcm_mixer.h
@@ -127,4 +127,10 @@ void mixer_channel_set_buffer_hook(enum pcm_mixer_channel channel,
/* Stop ALL channels and PCM and reset state */
void mixer_reset(void);
+/* Set output samplerate */
+void mixer_set_frequency(unsigned int samplerate);
+
+/* Get output samplerate */
+unsigned int mixer_get_frequency(void);
+
#endif /* PCM_MIXER_H */
diff --git a/firmware/export/pcm_sampr.h b/firmware/export/pcm_sampr.h
index 01a8ed428e..dcb1bdd80f 100644
--- a/firmware/export/pcm_sampr.h
+++ b/firmware/export/pcm_sampr.h
@@ -20,7 +20,12 @@
****************************************************************************/
#ifndef PCM_SAMPR_H
+
+/* File might be included for CPP config macros only. Allow it to be included
+ * again for full C declarations. */
+#ifndef PCM_SAMPR_CONFIG_ONLY
#define PCM_SAMPR_H
+#endif
#ifndef HW_SAMPR_CAPS
#define HW_SAMPR_CAPS SAMPR_CAP_44 /* if not defined, default to 44100 */
@@ -75,11 +80,14 @@
SAMPR_CAP_24 | SAMPR_CAP_22 | SAMPR_CAP_16 | \
SAMPR_CAP_12 | SAMPR_CAP_11 | SAMPR_CAP_8)
+#ifndef PCM_SAMPR_CONFIG_ONLY
/* Master list of all "standard" rates supported. */
extern const unsigned long audio_master_sampr_list[SAMPR_NUM_FREQ];
+#endif /* PCM_SAMPR_CONFIG_ONLY */
/** Hardware sample rates **/
+#ifndef PCM_SAMPR_CONFIG_ONLY
/* Enumeration of supported frequencies where 0 is the highest rate
supported and REC_NUM_FREQUENCIES is the number available */
enum hw_freq_indexes
@@ -183,14 +191,49 @@ enum hw_freq_indexes
#define HW_HAVE_8_(...)
#endif
HW_NUM_FREQ,
- HW_FREQ_DEFAULT = HW_FREQ_44,
- HW_SAMPR_DEFAULT = SAMPR_44,
}; /* enum hw_freq_indexes */
/* list of hardware sample rates */
extern const unsigned long hw_freq_sampr[HW_NUM_FREQ];
+#endif /* PCM_SAMPR_CONFIG_ONLY */
+
+#define HW_FREQ_DEFAULT HW_FREQ_44
+#define HW_SAMPR_DEFAULT SAMPR_44
+
+
+#if HW_SAMPR_CAPS & SAMPR_CAP_96
+# define HW_SAMPR_MAX SAMPR_96
+#elif HW_SAMPR_CAPS & SAMPR_CAP_88
+# define HW_SAMPR_MAX SAMPR_88
+#elif HW_SAMPR_CAPS & SAMPR_CAP_64
+# define HW_SAMPR_MAX SAMPR_64
+#elif HW_SAMPR_CAPS & SAMPR_CAP_48
+# define HW_SAMPR_MAX SAMPR_48
+#else
+# define HW_SAMPR_MAX SAMPR_44
+#endif
+
+#if HW_SAMPR_CAPS & SAMPR_CAP_8
+# define HW_SAMPR_MIN SAMPR_8
+#elif HW_SAMPR_CAPS & SAMPR_CAP_11
+# define HW_SAMPR_MIN SAMPR_11
+#elif HW_SAMPR_CAPS & SAMPR_CAP_12
+# define HW_SAMPR_MIN SAMPR_12
+#elif HW_SAMPR_CAPS & SAMPR_CAP_16
+# define HW_SAMPR_MIN SAMPR_16
+#elif HW_SAMPR_CAPS & SAMPR_CAP_22
+# define HW_SAMPR_MIN SAMPR_22
+#elif HW_SAMPR_CAPS & SAMPR_CAP_24
+# define HW_SAMPR_MIN SAMPR_24
+#elif HW_SAMPR_CAPS & SAMPR_CAP_32
+# define HW_SAMPR_MIN SAMPR_32
+#else
+# define HW_SAMPR_MIN SAMPR_44
+#endif
#ifdef HAVE_RECORDING
+
+#ifndef PCM_SAMPR_CONFIG_ONLY
/* Enumeration of supported frequencies where 0 is the highest rate
supported and REC_NUM_FREQUENCIES is the number available */
enum rec_freq_indexes
@@ -296,6 +339,10 @@ enum rec_freq_indexes
REC_NUM_FREQ,
}; /* enum rec_freq_indexes */
+/* List of recording supported sample rates (set or subset of master list) */
+extern const unsigned long rec_freq_sampr[REC_NUM_FREQ];
+#endif /* PCM_SAMPR_CONFIG_ONLY */
+
/* Default to 44.1kHz if not otherwise specified */
#ifndef REC_FREQ_DEFAULT
#define REC_FREQ_DEFAULT REC_FREQ_44
@@ -314,8 +361,7 @@ enum rec_freq_indexes
REC_HAVE_16_(",16") REC_HAVE_12_(",12") \
REC_HAVE_11_(",11") REC_HAVE_8_(",8")[1]
-/* List of recording supported sample rates (set or subset of master list) */
-extern const unsigned long rec_freq_sampr[REC_NUM_FREQ];
+
#endif /* HAVE_RECORDING */
#ifdef CONFIG_SAMPR_TYPES
@@ -326,8 +372,10 @@ extern const unsigned long rec_freq_sampr[REC_NUM_FREQ];
#define SAMPR_TYPE_REC (0x01 << 24)
#endif
+#ifndef PCM_SAMPR_CONFIG_ONLY
unsigned int pcm_sampr_to_hw_sampr(unsigned int samplerate,
unsigned int type);
+#endif
#else /* ndef CONFIG_SAMPR_TYPES */
diff --git a/firmware/pcm.c b/firmware/pcm.c
index e095ab2cea..60ccdbd2fc 100644
--- a/firmware/pcm.c
+++ b/firmware/pcm.c
@@ -415,6 +415,12 @@ void pcm_set_frequency(unsigned int samplerate)
pcm_fsel = index;
}
+/* return last-set frequency */
+unsigned int pcm_get_frequency(void)
+{
+ return pcm_sampr;
+}
+
/* apply pcm settings to the hardware */
void pcm_apply_settings(void)
{
diff --git a/firmware/pcm_mixer.c b/firmware/pcm_mixer.c
index 34852e97e9..ceba31962e 100644
--- a/firmware/pcm_mixer.c
+++ b/firmware/pcm_mixer.c
@@ -25,7 +25,6 @@
#include "pcm.h"
#include "pcm-internal.h"
#include "pcm_mixer.h"
-#include "dsp_core.h" /* For NATIVE_FREQUENCY */
/* Channels use standard-style PCM callback interface but a latency of one
frame by double-buffering is introduced in order to facilitate mixing and
@@ -33,6 +32,8 @@
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;
+
/* Define this to nonzero to add a marker pulse at each frame start */
#define FRAME_BOUNDARY_MARKERS 0
@@ -65,7 +66,7 @@ static struct mixer_channel channels[PCM_MIXER_NUM_CHANNELS] IBSS_ATTR;
static struct mixer_channel * active_channels[PCM_MIXER_NUM_CHANNELS+1] IBSS_ATTR;
/* Number of silence frames to play after all data has played */
-#define MAX_IDLE_FRAMES (NATIVE_FREQUENCY*3 / MIX_FRAME_SAMPLES)
+#define MAX_IDLE_FRAMES (mixer_sampr*3 / MIX_FRAME_SAMPLES)
static unsigned int idle_counter = 0;
/** Mixing routines, CPU optmized **/
@@ -256,7 +257,7 @@ static void mixer_start_pcm(void)
#endif
/* Requires a shared global sample rate for all channels */
- pcm_set_frequency(NATIVE_FREQUENCY);
+ pcm_set_frequency(mixer_sampr);
/* Prepare initial frames and set up the double buffer */
mixer_buffer_callback(PCM_DMAST_STARTED);
@@ -438,3 +439,23 @@ void mixer_reset(void)
idle_counter = 0;
}
+
+/* Set output samplerate */
+void mixer_set_frequency(unsigned int samplerate)
+{
+ pcm_set_frequency(samplerate);
+ samplerate = pcm_get_frequency();
+
+ if (samplerate == mixer_sampr)
+ return;
+
+ /* All data is now invalid */
+ mixer_reset();
+ mixer_sampr = samplerate;
+}
+
+/* Get output samplerate */
+unsigned int mixer_get_frequency(void)
+{
+ return mixer_sampr;
+}
diff --git a/lib/rbcodec/dsp/compressor.c b/lib/rbcodec/dsp/compressor.c
index fc73f761a7..a222caed7f 100644
--- a/lib/rbcodec/dsp/compressor.c
+++ b/lib/rbcodec/dsp/compressor.c
@@ -28,6 +28,7 @@
#include "logf.h"
#include "dsp_proc_entry.h"
#include "compressor.h"
+#include "dsp_misc.h"
static struct compressor_settings curr_set; /* Cached settings */
@@ -40,7 +41,8 @@ static int32_t release_gain IBSS_ATTR; /* S7.24 format */
/** COMPRESSOR UPDATE
* Called via the menu system to configure the compressor process */
-static bool compressor_update(const struct compressor_settings *settings)
+static bool compressor_update(struct dsp_config *dsp,
+ const struct compressor_settings *settings)
{
/* make settings values useful */
int threshold = settings->threshold;
@@ -48,9 +50,10 @@ static bool compressor_update(const struct compressor_settings *settings)
static const int comp_ratios[] = { 2, 4, 6, 10, 0 };
int ratio = comp_ratios[settings->ratio];
bool soft_knee = settings->knee == 1;
- int release = settings->release_time * NATIVE_FREQUENCY / 1000;
+ int release = settings->release_time *
+ dsp_get_output_frequency(dsp) / 1000;
- bool changed = false;
+ bool changed = settings == &curr_set; /* If frequency change */
bool active = threshold < 0;
if (memcmp(settings, &curr_set, sizeof (curr_set)))
@@ -300,8 +303,8 @@ static inline int32_t get_compression_gain(struct sample_format *format,
void dsp_set_compressor(const struct compressor_settings *settings)
{
/* enable/disable the compressor depending upon settings */
- bool enable = compressor_update(settings);
struct dsp_config *dsp = dsp_get_config(CODEC_IDX_AUDIO);
+ bool enable = compressor_update(dsp, settings);
dsp_proc_enable(dsp, DSP_PROC_COMPRESSOR, enable);
dsp_proc_activate(dsp, DSP_PROC_COMPRESSOR, true);
}
@@ -386,15 +389,20 @@ static intptr_t compressor_configure(struct dsp_proc_entry *this,
break; /* Already enabled */
this->process = compressor_process;
+ /* Won't have been getting frequency updates */
+ compressor_update(dsp, &curr_set);
/* Fall-through */
case DSP_RESET:
case DSP_FLUSH:
release_gain = UNITY;
break;
+
+ case DSP_SET_OUT_FREQUENCY:
+ compressor_update(dsp, &curr_set);
+ break;
}
return 0;
- (void)dsp;
}
/* Database entry */
diff --git a/lib/rbcodec/dsp/crossfeed.c b/lib/rbcodec/dsp/crossfeed.c
index 36a98f1f33..fc40c6b4d5 100644
--- a/lib/rbcodec/dsp/crossfeed.c
+++ b/lib/rbcodec/dsp/crossfeed.c
@@ -24,6 +24,7 @@
#include "fixedpoint.h"
#include "fracmul.h"
#include "replaygain.h"
+#include "dsp_misc.h"
#include "dsp_proc_entry.h"
#include "dsp_filter.h"
#include "crossfeed.h"
@@ -44,32 +45,40 @@ void crossfeed_meier_process(struct dsp_proc_entry *this,
* to listen to on headphones with no crossfeed.
*/
+#define DELAY_LEN(fs) ((300*(fs) / 1000000)*2) /* ~300 uS */
+
/* Crossfeed */
static struct crossfeed_state
{
- int32_t gain; /* 00h: Direct path gain */
- int32_t coefs[3]; /* 04h: Coefficients for the shelving filter */
union
{
- struct /* 10h: Data for meier crossfeed */
+ struct /* Data for meier crossfeed */
{
- int32_t vcl;
- int32_t vcr;
- int32_t vdiff;
- int32_t coef1;
- int32_t coef2;
+ int32_t reserved; /* 00h: Reserved: overlaps gain */
+ int32_t vcl; /* 04h: Left filter output */
+ int32_t vcr; /* 08h: Right filter output */
+ int32_t vdiff; /* 0ch: L-R difference signal */
+ int32_t coef1; /* 10h: Left/right filter coef */
+ int32_t coef2; /* 14h: Crossfeed filter coef */
};
- struct /* 10h: Data for custom crossfeed */
+ struct /* Data for custom crossfeed */
{
+ int32_t gain; /* 00h: Direct path gain */
+ int32_t coefs[3]; /* 04h: Filter coefficients: b0, b1, a1 */
int32_t history[4]; /* 10h: Format is x[n - 1], y[n - 1] (L + R) */
- int32_t delay[13*2];/* 20h: Delay line buffer (L + R interleaved) */
+ int32_t *index; /* 20h: Current pointer into the delay line */
+ int32_t *index_max; /* 24h: Current max pointer of delay line */
+ /* 28h: Delay line buffer (L + R interleaved) */
+ int32_t delay[DELAY_LEN(DSP_OUT_MAX_HZ)]; /* Target-dependent size */
};
};
- int32_t *index; /* 88h: Current pointer into the delay line */
- /* 8ch */
} crossfeed_state IBSS_ATTR;
static int crossfeed_type = CROSSFEED_TYPE_NONE;
+/* Cached custom settings */
+static long crossfeed_lf_gain;
+static long crossfeed_hf_gain;
+static long crossfeed_cutoff;
/* Discard the sample histories */
static void crossfeed_flush(struct dsp_proc_entry *this)
@@ -82,12 +91,49 @@ static void crossfeed_flush(struct dsp_proc_entry *this)
}
else
{
- memset(state->history, 0,
- sizeof (state->history) + sizeof (state->delay));
+ memset(state->history, 0, sizeof (state->history));
+ memset(state->delay, 0, sizeof (state->delay));
state->index = state->delay;
}
}
+static void crossfeed_meier_update_filter(struct crossfeed_state *state,
+ unsigned int fout)
+{
+ /* 1 / (F.Rforward.C) */
+ state->coef1 = fp_div(2128, fout, 31);
+ /* 1 / (F.Rcross.C) */
+ state->coef2 = fp_div(1000, fout, 31);
+}
+
+static void crossfeed_custom_update_filter(struct crossfeed_state *state,
+ unsigned int fout)
+{
+ long lf_gain = crossfeed_lf_gain;
+ long hf_gain = crossfeed_hf_gain;
+ long cutoff = crossfeed_cutoff;
+ int32_t *c = state->coefs;
+
+ long scaler = get_replaygain_int(lf_gain * 10) << 7;
+
+ cutoff = fp_div(cutoff, fout, 32);
+ hf_gain -= lf_gain;
+ /* Divide cutoff by sqrt(10^(hf_gain/20)) to place cutoff at the -3 dB
+ * point instead of shelf midpoint. This is for compatibility with the old
+ * crossfeed shelf filter and should be removed if crossfeed settings are
+ * ever made incompatible for any other good reason.
+ */
+ cutoff = fp_div(cutoff, get_replaygain_int(hf_gain*5), 24);
+
+ filter_shelf_coefs(cutoff, hf_gain, false, c);
+ /* Scale coefs by LF gain and shift them to s0.31 format. We have no gains
+ * over 1 and can do this safely
+ */
+ c[0] = FRACMUL_SHL(c[0], scaler, 4);
+ c[1] = FRACMUL_SHL(c[1], scaler, 4);
+ c[2] <<= 4;
+}
+
/** DSP interface **/
@@ -114,24 +160,13 @@ void dsp_set_crossfeed_direct_gain(int gain)
/* Both gains should be below 0 dB */
void dsp_set_crossfeed_cross_params(long lf_gain, long hf_gain, long cutoff)
{
- int32_t *c = crossfeed_state.coefs;
- long scaler = get_replaygain_int(lf_gain * 10) << 7;
+ crossfeed_lf_gain = lf_gain;
+ crossfeed_hf_gain = hf_gain;
+ crossfeed_cutoff = cutoff;
- cutoff = 0xffffffff / NATIVE_FREQUENCY * cutoff;
- hf_gain -= lf_gain;
- /* Divide cutoff by sqrt(10^(hf_gain/20)) to place cutoff at the -3 dB
- * point instead of shelf midpoint. This is for compatibility with the old
- * crossfeed shelf filter and should be removed if crossfeed settings are
- * ever made incompatible for any other good reason.
- */
- cutoff = fp_div(cutoff, get_replaygain_int(hf_gain*5), 24);
- filter_shelf_coefs(cutoff, hf_gain, false, c);
- /* Scale coefs by LF gain and shift them to s0.31 format. We have no gains
- * over 1 and can do this safely
- */
- c[0] = FRACMUL_SHL(c[0], scaler, 4);
- c[1] = FRACMUL_SHL(c[1], scaler, 4);
- c[2] <<= 4;
+ struct dsp_config *dsp = dsp_get_config(CODEC_IDX_AUDIO);
+ crossfeed_custom_update_filter(&crossfeed_state,
+ dsp_get_output_frequency(dsp));
}
#if !defined(CPU_COLDFIRE) && !defined(CPU_ARM)
@@ -147,6 +182,7 @@ void crossfeed_process(struct dsp_proc_entry *this, struct dsp_buffer **buf_p)
int32_t *coefs = &state->coefs[0];
int32_t gain = state->gain;
int32_t *di = state->index;
+ int32_t *di_max = state->index_max;
int count = buf->remcount;
@@ -176,7 +212,7 @@ void crossfeed_process(struct dsp_proc_entry *this, struct dsp_buffer **buf_p)
buf->p32[1][i] = FRACMUL(right, gain) + hist_l[1];
/* Wrap delay line index if bigger than delay line size */
- if (di >= delay + 13*2)
+ if (di >= di_max)
di = delay;
}
@@ -234,17 +270,21 @@ static void update_process_fn(struct dsp_proc_entry *this,
struct dsp_config *dsp)
{
struct crossfeed_state *state = (struct crossfeed_state *)this->data;
- dsp_proc_fn_type fn = crossfeed_process;
+ dsp_proc_fn_type fn;
+
+ unsigned int fout = dsp_get_output_frequency(dsp);
if (crossfeed_type != CROSSFEED_TYPE_CUSTOM)
{
- /* Set up for Meier */
- /* 1 / (F.Rforward.C) */
- state->coef1 = (0x7fffffff / NATIVE_FREQUENCY) * 2128;
- /* 1 / (F.Rcross.C) */
- state->coef2 = (0x7fffffff / NATIVE_FREQUENCY) * 1000;
+ crossfeed_meier_update_filter(state, fout);
fn = crossfeed_meier_process;
}
+ else
+ {
+ state->index_max = state->delay + DELAY_LEN(fout);
+ crossfeed_custom_update_filter(state, fout);
+ fn = crossfeed_process;
+ }
if (this->process != fn)
{
@@ -292,6 +332,7 @@ static intptr_t crossfeed_configure(struct dsp_proc_entry *this,
if (value == 0)
this->data = (intptr_t)&crossfeed_state;
+ case DSP_SET_OUT_FREQUENCY:
update_process_fn(this, dsp);
break;
diff --git a/lib/rbcodec/dsp/dsp_arm.S b/lib/rbcodec/dsp/dsp_arm.S
index ed58bed340..16394b8690 100644
--- a/lib/rbcodec/dsp/dsp_arm.S
+++ b/lib/rbcodec/dsp/dsp_arm.S
@@ -196,55 +196,56 @@ crossfeed_process:
@ to keep the count on the stack :/
ldr r1, [r1] @ r1 = buf = *buf_p;
stmfd sp!, { r4-r11, lr } @ stack modified regs
- ldr r12, [r1] @ r12 = buf->remcount
- ldr r14, [r0] @ r14 = this->data = &crossfeed_state
- ldmib r1, { r2-r3 } @ r2 = buf->p32[0], r3 = buf->p32[1]
- ldmia r14!, { r4-r11 } @ load direct gain and filter data
- add r0, r14, #13*2*4 @ calculate end of delay
- stmfd sp!, { r0, r12 } @ stack end of delay adr, count and state
- ldr r0, [r0] @ fetch current delay line address
-
- /* Register usage in loop:
- * r0 = &delay[index][0], r1 = accumulator high, r2 = buf->p32[0],
+ ldr r0, [r0] @ r0 = this->data = &crossfeed_state
+ ldmia r1, { r1-r3 } @ r1 = buf->remcount, r2 = buf->p32[0],
+ @ r3 = buf->p32[1]
+ ldmia r0, { r4-r12, r14 } @ r4 = gain, r5-r7 = coeffs,
+ @ r8-r11 = history, r12 = index,
+ @ r14 = index_max
+ add r0, r0, #0x28 @ r0 = state->delay
+ stmfd sp!, { r0-r1, r14 } @ stack state->delay, count, index_max
+
+ /* Register usage in loop:
+ * r0 = acc low/count, r1 = acc high, r2 = buf->p32[0],
* r3 = buf->p32[1], r4 = direct gain, r5-r7 = b0, b1, a1 (filter coefs),
- * r8-r11 = filter history, r12 = temp, r14 = accumulator low
+ * r8 = dr[n-1], r9 = y_r[n-1], r10 = dl[n-1], r11 = y_l[n-1],
+ * r12 = index, r14 = scratch/index_max
*/
.cfloop:
- smull r14, r1, r6, r8 @ acc = b1*dr[n - 1]
- smlal r14, r1, r7, r9 @ acc += a1*y_l[n - 1]
- ldr r8, [r0, #4] @ r8 = dr[n]
- smlal r14, r1, r5, r8 @ acc += b0*dr[n]
- mov r9, r1, lsl #1 @ fix format for filter history
- ldr r12, [r2] @ load left input
- smlal r14, r1, r4, r12 @ acc += gain*x_l[n]
- mov r1, r1, lsl #1 @ fix format
+ smull r0, r1, r6, r8 @ acc = b1*dr[n - 1]
+ ldr r8, [r12, #4] @ r8 = dr[n]
+ smlal r0, r1, r7, r9 @ acc += a1*y_r[n - 1]
+ smlal r0, r1, r5, r8 @ acc += b0*dr[n]
+ ldr r14, [r2] @ load left input: x_l[n]
+ mov r9, r1, asl #1 @ fix format for filter history
+ smlal r0, r1, r4, r14 @ acc += gain*x_l[n]
+ mov r1, r1, asl #1 @ fix format
str r1, [r2], #4 @ save result
-
- smull r14, r1, r6, r10 @ acc = b1*dl[n - 1]
- smlal r14, r1, r7, r11 @ acc += a1*y_r[n - 1]
- ldr r10, [r0] @ r10 = dl[n]
- str r12, [r0], #4 @ save left input to delay line
- smlal r14, r1, r5, r10 @ acc += b0*dl[n]
- mov r11, r1, lsl #1 @ fix format for filter history
- ldr r12, [r3] @ load right input
- smlal r14, r1, r4, r12 @ acc += gain*x_r[n]
- str r12, [r0], #4 @ save right input to delay line
- mov r1, r1, lsl #1 @ fix format
- ldmia sp, { r12, r14 } @ fetch delay line end addr and count from stack
+ smull r0, r1, r6, r10 @ acc = b1*dl[n - 1]
+ ldr r10, [r12] @ r10 = dl[n]
+ smlal r0, r1, r7, r11 @ acc += a1*y_l[n - 1]
+ smlal r0, r1, r5, r10 @ acc += b0*dl[n]
+ str r14, [r12], #4 @ save left input to delay line
+ ldr r14, [r3] @ load right input: x_r[n]
+ mov r11, r1, asl #1 @ fix format for filter history
+ smlal r0, r1, r4, r14 @ acc += gain*x_r[n]
+ str r14, [r12], #4 @ save right input to delay line
+ ldmib sp, { r0, r14 } @ fetch count and delay end
+ mov r1, r1, asl #1 @ fix format
str r1, [r3], #4 @ save result
- cmp r0, r12 @ need to wrap to start of delay?
- subhs r0, r12, #13*2*4 @ wrap back delay line ptr to start
+ cmp r12, r14 @ need to wrap to start of delay?
+ ldrhs r12, [sp] @ wrap delay index
- subs r14, r14, #1 @ are we finished?
- strgt r14, [sp, #4] @ nope, save count back to stack
+ subs r0, r0, #1 @ are we finished?
+ strgt r0, [sp, #4] @ save count to stack
bgt .cfloop
@ save data back to struct
- str r0, [r12] @ save delay line index
- sub r12, r12, #13*2*4 + 4*4 @ r12 = data->history
- stmia r12, { r8-r11 } @ save filter history
- add sp, sp, #8 @ remove temp variables from stack
+ ldr r0, [sp] @ fetch state->delay
+ sub r0, r0, #0x18 @ save filter history and delay index
+ stmia r0, { r8-r12 } @
+ add sp, sp, #12 @ remove temp variables from stack
ldmpc regs=r4-r11
.size crossfeed_process, .-crossfeed_process
@@ -260,8 +261,7 @@ crossfeed_meier_process:
ldr r0, [r0] @ r0 = this->data = &crossfeed_state
stmfd sp!, { r4-r10, lr } @ stack non-volatile context
ldmia r1, { r1-r3 } @ r1 = buf->remcout, r2=p32[0], r3=p32[1]
- add r0, r0, #16 @ r0 = &state->vcl
- ldmia r0, { r4-r8 } @ r4 = vcl, r5 = vcr, r6 = vdiff
+ ldmib r0, { r4-r8 } @ r4 = vcl, r5 = vcr, r6 = vdiff
@ r7 = coef1, r8 = coef2
.cfm_loop:
ldr r12, [r2] @ r12 = lout
@@ -285,7 +285,7 @@ crossfeed_meier_process:
sub r5, r5, r12 @ r5 = vcr -= res2
bgt .cfm_loop @ more samples?
- stmia r0, { r4-r6 } @ save vcl, vcr, vdiff
+ stmib r0, { r4-r6 } @ save vcl, vcr, vdiff
ldmpc regs=r4-r10 @ restore non-volatile context, return
.size crossfeed_meier_process, .-crossfeed_meier_process
diff --git a/lib/rbcodec/dsp/dsp_cf.S b/lib/rbcodec/dsp/dsp_cf.S
index 02db8f61b6..e34075ef9a 100644
--- a/lib/rbcodec/dsp/dsp_cf.S
+++ b/lib/rbcodec/dsp/dsp_cf.S
@@ -81,58 +81,60 @@ crossfeed_process:
movem.l %d2-%d7/%a2-%a6, (%sp) | save all regs
movem.l 48(%sp), %a1/%a4 | %a1 = this, %a4 = buf_p
move.l (%a4), %a4 | %a4 = buf = *buf_p
- movem.l (%a4), %d7/%a4-%a5 | %d7 = buf->remcount, %a4 = buf->p32[0],
+ movem.l (%a4), %d0/%a4-%a5 | %d0 = buf->remcount, %a4 = buf->p32[0],
| %a5 = buf->p32[1]
- move.l (%a1), %a1 | %a1 = &crossfeed_state
- move.l (%a1)+, %d6 | %d6 = direct gain
- movem.l 12(%a1), %d0-%d3 | fetch filter history samples
- lea.l 132(%a1), %a6 | %a6 = delay line wrap limit
- move.l (%a6), %a0 | fetch delay line address
- movem.l (%a1), %a1-%a3 | load filter coefs
- bra.b 20f | loop start | go to loop start point
+ move.l (%a1), %a6 | %d7 = state = &crossfeed_state
+ movem.l (%a6), %d1-%d6/%a0-%a3 | %d1 = gain, %d2-%d4 = coefs,
+ | %d5..%d6 = history[0..1],
+ | %a0..%a1 = history[2..3],
+ | %a2 = index, %a3 = index_max
+ lea.l 0x28(%a6), %a6 | %a6 = state->delay
+ move.l %a6, -(%sp) | push state->delay
+ bra.b .cfp_loop_start
/* Register usage in loop:
- * %a0 = delay_p, %a1..%a3 = b0, b1, a1 (filter coefs),
- * %a4 = buf[0], %a5 = buf[1],
- * %a6 = delay line pointer wrap limit,
- * %d0..%d3 = history
- * %d4..%d5 = temp.
- * %d6 = direct gain,
- * %d7 = count
+ * %d0 = count, %d1 = direct gain, %d2..%d4 = b0, b1, a1 (filter coefs),
+ * %d5..%d6 = history[0..1], %d7 = scratch
+ * %a0..%a1 = history[2..3], %a2 = index, %a3 = index_max,
+ * %a4 = buf[0], %a5 = buf[1], %a6 = scratch
*/
-10: | loop |
- movclr.l %acc0, %d4 | write outputs
- move.l %d4, (%a4)+ | .
- movclr.l %acc1, %d5 | .
- move.l %d5, (%a5)+ | .
-20: | loop start |
- mac.l %a2, %d0, (%a0)+, %d0, %acc0 | %acc0 = b1*dl[n - 1], %d0 = dl[n]
- mac.l %a1, %d0 , %acc0 | %acc0 += b0*dl[n]
- mac.l %a3, %d1, (%a5), %d5, %acc0 | %acc0 += a1*y_r[n - 1], load R
- mac.l %a2, %d2, (%a0)+, %d2, %acc1 | %acc1 = b1*dr[n - 1], %d2 = dr[n]
- mac.l %a1, %d2 , %acc1 | %acc1 += b0*dr[n]
- mac.l %a3, %d3, (%a4), %d4, %acc1 | %acc1 += a1*y_l[n - 1], load L
- movem.l %d4-%d5, -8(%a0) | save left & right inputs to delay line
- move.l %acc0, %d3 | get filtered delayed left sample (y_l[n])
- move.l %acc1, %d1 | get filtered delayed right sample (y_r[n])
- mac.l %d6, %d4, %acc0 | %acc0 += gain*x_l[n]
- mac.l %d6, %d5, %acc1 | %acc1 += gain*x_r[n]
- cmp.l %a6, %a0 | wrap %a0 if passed end
- bhs.b 30f | wrap buffer |
- tpf.l | trap the buffer wrap
-30: | wrap buffer | ...fwd taken branches more costly
- lea.l -104(%a6), %a0 | wrap it up
- subq.l #1, %d7 | --count > 0 ?
- bgt.b 10b | loop | yes? do more
- movclr.l %acc0, %d4 | write last outputs
- move.l %d4, (%a4) | .
- movclr.l %acc1, %d5 | .
- move.l %d5, (%a5) | .
- movem.l %d0-%d3, -120(%a6) | ...history
- move.l %a0, (%a6) | ...delay_p
+.cfp_loop:
+ movclr.l %acc0, %d7 | write outputs
+ move.l %d7, (%a4)+ | .
+ movclr.l %acc1, %a6 | .
+ move.l %a6, (%a5)+ | .
+.cfp_loop_start:
+ mac.l %d3, %d5, (%a2)+, %d5, %acc1 | %acc1 = b1*dl[n - 1], %d5 = dl[n]
+ mac.l %d2, %d5 , %acc1 | %acc1 += b0*dl[n]
+ mac.l %d4, %d6, (%a4), %d7, %acc1 | %acc1 += a1*y_l[n - 1], %d7 = x_l[n]
+ mac.l %d3, %a0, (%a2)+, %a0, %acc0 | %acc0 = b1*dr[n - 1], %a0 = dr[n]
+ mac.l %a2, %a0 , %acc0 | %acc0 += b0*dr[n]
+ mac.l %d4, %a1, (%a5), %a6, %acc0 | %acc0 += a1*y_r[n - 1], %a6 = x_r[n]
+ movem.l %d7/%a6, -8(%a2) | save x_l[n] and x_r[n] to delay line
+ move.l %acc1, %d6 | get filtered delayed left sample (y_l[n])
+ move.l %acc0, %a1 | get filtered delayed right sample (y_r[n])
+ mac.l %d1, %d7, %acc0 | %acc0 = gain*x_l[n] + y_r[n]
+ mac.l %d1, %a6, %acc1 | %acc1 = gain*x_r[n] + y_l[n]
+
+ cmp.l %a3, %a2 | wrap index if past end
+ bhs.b 1f |
+ tpf.w | trap the buffer wrap
+1: | ...fwd taken branches more costly
+ move.l (%sp), %a2 | 2b | wrap it up
+
+ subq.l #1, %d0 | --count > 0 ?
+ bgt.b .cfp_loop | yes? do more
+
+ movclr.l %acc0, %d7 | write last outputs
+ move.l %d7, (%a4) | .
+ movclr.l %acc1, %a6 | .
+ move.l %a6, (%a5) | .
+
+ move.l (%sp)+, %a6 | pop state->delay
+ movem.l %d5-%d6/%a0-%a2, -0x18(%a6) | save history, index
movem.l (%sp), %d2-%d7/%a2-%a6 | restore all regs
lea.l 44(%sp), %sp |
rts |
- .size crossfeed_process,.-crossfeed_process
+ .size crossfeed_process, .-crossfeed_process
/****************************************************************************
* void crossfeed_meier_process(struct dsp_proc_entry *this,
@@ -147,7 +149,7 @@ crossfeed_meier_process:
movem.l %d2-%d6/%a2, (%sp) | .
move.l (%a0), %a0 | %a0 = &this->data = &crossfeed_state
move.l (%a1), %a1 | %a1 = buf = *buf_p
- movem.l 16(%a0), %d1-%d5 | %d1 = vcl, %d2 = vcr, %d3 = vdiff,
+ movem.l 4(%a0), %d1-%d5 | %d1 = vcl, %d2 = vcr, %d3 = vdiff,
| %d4 = coef1, %d5 = coef2
movem.l (%a1), %d0/%a1-%a2 | %d0 = count = buf->remcount
| %a1 = p32[0], %a2 = p32[1]
@@ -155,7 +157,7 @@ crossfeed_meier_process:
| %d0 = count, %d1 = vcl, %d2 = vcr, %d3 = vdiff/lout,
| %d4 = coef1, %d5 = coef2, %d6 = rout/scratch
| %a1 = p32[0], %a2 = p32[1]
-10: | loop
+.cfmp_loop:
mac.l %d5, %d3, %acc0 | %acc0 = common = coef2*vdiff
move.l %acc0, %acc1 | copy common
mac.l %d4, %d1, (%a1), %d3, %acc0 | %acc0 += coef1*vcl, %d3 = lout
@@ -170,9 +172,9 @@ crossfeed_meier_process:
movclr.l %acc1, %d6 | %d5 = fetch -res2 in s0.31
add.l %d6, %d2 | vcr += -res2
subq.l #1, %d0 | count--
- bgt 10b | loop | more samples?
+ bgt .cfmp_loop | more samples?
|
- movem.l %d1-%d3, 16(%a0) | save vcl, vcr, vdiff
+ movem.l %d1-%d3, 4(%a0) | save vcl, vcr, vdiff
movem.l (%sp), %d2-%d6/%a2 | restore non-volatiles
lea.l 24(%sp), %sp | .
rts |
diff --git a/lib/rbcodec/dsp/dsp_core.c b/lib/rbcodec/dsp/dsp_core.c
index 871ccbfd23..b0e9c8a304 100644
--- a/lib/rbcodec/dsp/dsp_core.c
+++ b/lib/rbcodec/dsp/dsp_core.c
@@ -103,8 +103,21 @@ static intptr_t proc_broadcast(struct dsp_config *dsp, unsigned int setting,
intptr_t value)
{
bool multi = setting < DSP_PROC_SETTING;
- struct dsp_proc_slot *s = multi ? dsp->proc_slots :
- find_proc_slot(dsp, setting - DSP_PROC_SETTING);
+ struct dsp_proc_slot *s;
+
+ if (multi)
+ {
+ /* Message to all enabled stages */
+ if (dsp_sample_io_configure(&dsp->io_data, setting, &value))
+ return value; /* To I/O only */
+
+ s = dsp->proc_slots;
+ }
+ else
+ {
+ /* Message to a particular stage */
+ s = find_proc_slot(dsp, setting - DSP_PROC_SETTING);
+ }
while (s != NULL)
{
@@ -117,7 +130,7 @@ static intptr_t proc_broadcast(struct dsp_config *dsp, unsigned int setting,
s = s->next;
}
- return multi ? 1 : 0;
+ return 0;
}
/* Add an item to the enabled list */
@@ -244,6 +257,12 @@ void dsp_proc_enable(struct dsp_config *dsp, enum dsp_proc_ids id,
proc_db_entry(s)->configure(&s->proc_entry, dsp, DSP_PROC_CLOSE, 0);
}
+/* Is the stage specified by the id currently enabled? */
+bool dsp_proc_enabled(struct dsp_config *dsp, enum dsp_proc_ids id)
+{
+ return (dsp->proc_mask_enabled & BIT_N(id)) != 0;
+}
+
/* Activate or deactivate a stage */
void dsp_proc_activate(struct dsp_config *dsp, enum dsp_proc_ids id,
bool activate)
@@ -454,7 +473,6 @@ void dsp_process(struct dsp_config *dsp, struct dsp_buffer *src,
intptr_t dsp_configure(struct dsp_config *dsp, unsigned int setting,
intptr_t value)
{
- dsp_sample_io_configure(&dsp->io_data, setting, value);
return proc_broadcast(dsp, setting, value);
}
@@ -497,7 +515,8 @@ void INIT_ATTR dsp_init(void)
count = slot_count[i];
dsp->slot_free_mask = MASK_N(uint32_t, count, shift);
- dsp_sample_io_configure(&dsp->io_data, DSP_INIT, i);
+ intptr_t value = i;
+ dsp_sample_io_configure(&dsp->io_data, DSP_INIT, &value);
/* Notify each db entry of global init for each DSP */
for (unsigned int j = 0; j < DSP_NUM_PROC_STAGES; j++)
diff --git a/lib/rbcodec/dsp/dsp_core.h b/lib/rbcodec/dsp/dsp_core.h
index d3cfdd133c..0f63b62e00 100644
--- a/lib/rbcodec/dsp/dsp_core.h
+++ b/lib/rbcodec/dsp/dsp_core.h
@@ -39,14 +39,14 @@ enum dsp_settings
DSP_SET_STEREO_MODE,
DSP_FLUSH,
DSP_SET_PITCH,
+ DSP_SET_OUT_FREQUENCY,
+ DSP_GET_OUT_FREQUENCY,
DSP_PROC_INIT,
DSP_PROC_CLOSE,
DSP_PROC_NEW_FORMAT,
DSP_PROC_SETTING, /* stage-specific should be this + id */
};
-#define NATIVE_FREQUENCY 44100 /* internal/output sample rate */
-
enum dsp_stereo_modes
{
STEREO_INTERLEAVED,
diff --git a/lib/rbcodec/dsp/dsp_misc.c b/lib/rbcodec/dsp/dsp_misc.c
index cc74a790ea..ad6f5b5b31 100644
--- a/lib/rbcodec/dsp/dsp_misc.c
+++ b/lib/rbcodec/dsp/dsp_misc.c
@@ -134,6 +134,21 @@ int32_t dsp_get_pitch(void)
}
#endif /* HAVE_PITCHCONTROL */
+/* Set output samplerate for all DSPs */
+void dsp_set_all_output_frequency(unsigned int samplerate)
+{
+
+ struct dsp_config *dsp;
+ for (int i = 0; (dsp = dsp_get_config(i)); i++)
+ dsp_configure(dsp, DSP_SET_OUT_FREQUENCY, samplerate);
+}
+
+/* Return DSP's output samplerate */
+unsigned int dsp_get_output_frequency(struct dsp_config *dsp)
+{
+ return dsp_configure(dsp, DSP_GET_OUT_FREQUENCY, 0);
+}
+
static void INIT_ATTR misc_dsp_init(struct dsp_config *dsp,
enum dsp_ids dsp_id)
{
diff --git a/lib/rbcodec/dsp/dsp_misc.h b/lib/rbcodec/dsp/dsp_misc.h
index 2fed9400f2..af259bfa3e 100644
--- a/lib/rbcodec/dsp/dsp_misc.h
+++ b/lib/rbcodec/dsp/dsp_misc.h
@@ -59,4 +59,11 @@ void dsp_set_pitch(int32_t pitch);
int32_t dsp_get_pitch(void);
#endif /* HAVE_PITCHCONTROL */
+/* Set output samplerate for all DSPs */
+void dsp_set_all_output_frequency(unsigned int samplerate);
+
+/* Return DSP's output samplerate */
+struct dsp_config;
+unsigned int dsp_get_output_frequency(struct dsp_config *dsp);
+
#endif /* DSP_MISC_H */
diff --git a/lib/rbcodec/dsp/dsp_proc_database.h b/lib/rbcodec/dsp/dsp_proc_database.h
index c4c93ef2d9..534c165eef 100644
--- a/lib/rbcodec/dsp/dsp_proc_database.h
+++ b/lib/rbcodec/dsp/dsp_proc_database.h
@@ -40,7 +40,7 @@ DSP_PROC_DB_START
#ifdef HAVE_PITCHCONTROL
DSP_PROC_DB_ITEM(TIMESTRETCH) /* time-stretching */
#endif
- DSP_PROC_DB_ITEM(RESAMPLE) /* resampler providing NATIVE_FREQUENCY */
+ DSP_PROC_DB_ITEM(RESAMPLE) /* resampler providing output frequency */
DSP_PROC_DB_ITEM(CROSSFEED) /* stereo crossfeed */
DSP_PROC_DB_ITEM(EQUALIZER) /* n-band equalizer */
#ifdef HAVE_SW_TONE_CONTROLS
diff --git a/lib/rbcodec/dsp/dsp_proc_entry.h b/lib/rbcodec/dsp/dsp_proc_entry.h
index 902385f08d..1bf59dd73a 100644
--- a/lib/rbcodec/dsp/dsp_proc_entry.h
+++ b/lib/rbcodec/dsp/dsp_proc_entry.h
@@ -132,6 +132,10 @@ typedef intptr_t (*dsp_proc_config_fn_type)(struct dsp_proc_entry *this,
* by processing code! */
void dsp_proc_enable(struct dsp_config *dsp, enum dsp_proc_ids id,
bool enable);
+
+/* Is the specified stage enabled on the DSP? */
+bool dsp_proc_enabled(struct dsp_config *dsp, enum dsp_proc_ids id);
+
/* Activate/deactivate processing stage, doesn't affect enabled status
* thus will not enable anything -
* may be called during processing to activate/deactivate for format
diff --git a/lib/rbcodec/dsp/dsp_sample_input.c b/lib/rbcodec/dsp/dsp_sample_input.c
index 561cb36d9e..537a659b73 100644
--- a/lib/rbcodec/dsp/dsp_sample_input.c
+++ b/lib/rbcodec/dsp/dsp_sample_input.c
@@ -286,14 +286,17 @@ static void INIT_ATTR dsp_sample_input_init(struct sample_io_data *this,
static void INIT_ATTR dsp_sample_io_init(struct sample_io_data *this,
enum dsp_ids dsp_id)
{
+ this->output_sampr = DSP_OUT_DEFAULT_HZ;
dsp_sample_input_init(this, dsp_id);
dsp_sample_output_init(this);
}
-void dsp_sample_io_configure(struct sample_io_data *this,
+bool dsp_sample_io_configure(struct sample_io_data *this,
unsigned int setting,
- intptr_t value)
+ intptr_t *value_p)
{
+ intptr_t value = *value_p;
+
switch (setting)
{
case DSP_INIT:
@@ -306,15 +309,15 @@ void dsp_sample_io_configure(struct sample_io_data *this,
this->format.num_channels = 2;
this->format.frac_bits = WORD_FRACBITS;
this->format.output_scale = WORD_FRACBITS + 1 - NATIVE_DEPTH;
- this->format.frequency = NATIVE_FREQUENCY;
- this->format.codec_frequency = NATIVE_FREQUENCY;
+ this->format.frequency = this->output_sampr;
+ this->format.codec_frequency = this->output_sampr;
this->sample_depth = NATIVE_DEPTH;
this->stereo_mode = STEREO_NONINTERLEAVED;
break;
case DSP_SET_FREQUENCY:
format_change_set(this);
- value = value > 0 ? value : NATIVE_FREQUENCY;
+ value = value > 0 ? (unsigned int)value : this->output_sampr;
this->format.frequency = value;
this->format.codec_frequency = value;
break;
@@ -345,5 +348,22 @@ void dsp_sample_io_configure(struct sample_io_data *this,
this->format.frequency =
fp_mul(value, this->format.codec_frequency, 16);
break;
+
+ case DSP_SET_OUT_FREQUENCY:
+ value = value > 0 ? value : DSP_OUT_DEFAULT_HZ;
+ value = MIN(DSP_OUT_MAX_HZ, MAX(DSP_OUT_MIN_HZ, value));
+ *value_p = value;
+
+ if ((unsigned int)value == this->output_sampr)
+ return true; /* No change; don't broadcast */
+
+ this->output_sampr = value;
+ break;
+
+ case DSP_GET_OUT_FREQUENCY:
+ *value_p = this->output_sampr;
+ return true; /* Only I/O handles it */
}
+
+ return false;
}
diff --git a/lib/rbcodec/dsp/dsp_sample_io.h b/lib/rbcodec/dsp/dsp_sample_io.h
index 22b7a4a3f4..5117e04a3e 100644
--- a/lib/rbcodec/dsp/dsp_sample_io.h
+++ b/lib/rbcodec/dsp/dsp_sample_io.h
@@ -50,6 +50,7 @@ struct sample_io_data
struct dsp_buffer sample_buf; /* Buffer descriptor for converted samples */
int32_t *sample_buf_p[2]; /* Internal format buffer pointers */
sample_output_fn_type output_samples; /* Final output function */
+ unsigned int output_sampr; /* Master output samplerate */
uint8_t format_dirty; /* Format change set, avoids superfluous
increments before carrying it out */
uint8_t output_version; /* Format version of src buffer at output */
@@ -62,8 +63,8 @@ void dsp_sample_output_format_change(struct sample_io_data *this,
struct sample_format *format);
/* Sample IO watches the format setting from the codec */
-void dsp_sample_io_configure(struct sample_io_data *this,
+bool dsp_sample_io_configure(struct sample_io_data *this,
unsigned int setting,
- intptr_t value);
+ intptr_t *value_p);
#endif /* DSP_SAMPLE_IO_H */
diff --git a/lib/rbcodec/dsp/eq.c b/lib/rbcodec/dsp/eq.c
index 94cb61deec..e4d7bf5e02 100644
--- a/lib/rbcodec/dsp/eq.c
+++ b/lib/rbcodec/dsp/eq.c
@@ -25,6 +25,7 @@
#include "dsp_filter.h"
#include "dsp_proc_entry.h"
#include "dsp_core.h"
+#include "dsp_misc.h"
#include "eq.h"
#include "pga.h"
#include "replaygain.h"
@@ -42,6 +43,9 @@
#error Band count must be greater than or equal to 3
#endif
+/* Cached band settings */
+static struct eq_band_setting settings[EQ_NUM_BANDS];
+
static struct eq_state
{
uint32_t enabled; /* Mask of enabled bands */
@@ -49,16 +53,48 @@ static struct eq_state
struct dsp_filter filters[EQ_NUM_BANDS]; /* Data for each filter */
} eq_data IBSS_ATTR;
+#define FOR_EACH_ENB_BAND(b) \
+ for (uint8_t *b = eq_data.bands; *b < EQ_NUM_BANDS; b++)
+
/* Clear histories of all enabled bands */
static void eq_flush(void)
{
if (eq_data.enabled == 0)
return; /* Not initialized yet/no bands on */
- for (uint8_t *b = eq_data.bands; *b < EQ_NUM_BANDS; b++)
+ FOR_EACH_ENB_BAND(b)
filter_flush(&eq_data.filters[*b]);
}
+static void update_band_filter(int band, unsigned int fout)
+{
+ /* Convert user settings to format required by coef generator
+ functions */
+ typeof (filter_pk_coefs) *coef_gen = filter_pk_coefs;
+
+ /* Only first and last bands are not peaking filters */
+ if (band == 0)
+ coef_gen = filter_ls_coefs;
+ else if (band == EQ_NUM_BANDS-1)
+ coef_gen = filter_hs_coefs;
+
+ const struct eq_band_setting *setting = &settings[band];
+ struct dsp_filter *filter = &eq_data.filters[band];
+
+ coef_gen(fp_div(setting->cutoff, fout, 32), setting->q ?: 1,
+ setting->gain, filter);
+}
+
+/* Resync all bands to a new DSP output frequency */
+static void update_samplerate(unsigned int fout)
+{
+ if (eq_data.enabled == 0)
+ return; /* Not initialized yet/no bands on */
+
+ FOR_EACH_ENB_BAND(b)
+ update_band_filter(*b, fout);
+}
+
/** DSP interface **/
/* Set the precut gain value */
@@ -73,11 +109,14 @@ void dsp_set_eq_coefs(int band, const struct eq_band_setting *setting)
if (band < 0 || band >= EQ_NUM_BANDS)
return;
+ settings[band] = *setting; /* cache setting */
+
+ struct dsp_config *dsp = dsp_get_config(CODEC_IDX_AUDIO);
+
/* NOTE: The coef functions assume the EMAC unit is in fractional mode,
which it should be, since we're executed from the main thread. */
uint32_t mask = eq_data.enabled;
- struct dsp_filter *filter = &eq_data.filters[band];
/* Assume a band is disabled if the gain is zero */
mask &= ~BIT_N(band);
@@ -85,33 +124,19 @@ void dsp_set_eq_coefs(int band, const struct eq_band_setting *setting)
if (setting->gain != 0)
{
mask |= BIT_N(band);
-
- /* Convert user settings to format required by coef generator
- functions */
- void (* coef_gen)(unsigned long cutoff, unsigned long Q, long db,
- struct dsp_filter *f) = filter_pk_coefs;
-
- /* Only first and last bands are not peaking filters */
- if (band == 0)
- coef_gen = filter_ls_coefs;
- else if (band == EQ_NUM_BANDS-1)
- coef_gen = filter_hs_coefs;
-
- coef_gen(0xffffffff / NATIVE_FREQUENCY * setting->cutoff,
- setting->q ?: 1, setting->gain, filter);
+ update_band_filter(band, dsp_get_output_frequency(dsp));
}
if (mask == eq_data.enabled)
return; /* No change in band-enable state */
if (mask & BIT_N(band))
- filter_flush(filter); /* Coming online */
+ filter_flush(&eq_data.filters[band]); /* Coming online */
eq_data.enabled = mask;
/* Only be active if there are bands to process - if EQ is off, then
this call has no effect */
- struct dsp_config *dsp = dsp_get_config(CODEC_IDX_AUDIO);
dsp_proc_activate(dsp, DSP_PROC_EQUALIZER, mask != 0);
/* Prepare list of enabled bands for efficient iteration */
@@ -125,6 +150,11 @@ void dsp_set_eq_coefs(int band, const struct eq_band_setting *setting)
void dsp_eq_enable(bool enable)
{
struct dsp_config *dsp = dsp_get_config(CODEC_IDX_AUDIO);
+ bool enabled = dsp_proc_enabled(dsp, DSP_PROC_EQUALIZER);
+
+ if (enable == enabled)
+ return;
+
dsp_proc_enable(dsp, DSP_PROC_EQUALIZER, enable);
if (enable && eq_data.enabled != 0)
@@ -139,7 +169,7 @@ static void eq_process(struct dsp_proc_entry *this,
int count = buf->remcount;
unsigned int channels = buf->format.num_channels;
- for (uint8_t *b = eq_data.bands; *b < EQ_NUM_BANDS; b++)
+ FOR_EACH_ENB_BAND(b)
filter_process(&eq_data.filters[*b], buf->p32, count, channels);
(void)this;
@@ -154,10 +184,9 @@ static intptr_t eq_configure(struct dsp_proc_entry *this,
switch (setting)
{
case DSP_PROC_INIT:
- if (value != 0)
- break; /* Already enabled */
-
this->process = eq_process;
+ /* Wouldn't have been getting frequency updates */
+ update_samplerate(dsp_get_output_frequency(dsp));
/* Fall-through */
case DSP_PROC_CLOSE:
pga_enable_gain(PGA_EQ_PRECUT, setting == DSP_PROC_INIT);
@@ -166,6 +195,10 @@ static intptr_t eq_configure(struct dsp_proc_entry *this,
case DSP_FLUSH:
eq_flush();
break;
+
+ case DSP_SET_OUT_FREQUENCY:
+ update_samplerate(value);
+ break;
}
return 0;
diff --git a/lib/rbcodec/dsp/resample.c b/lib/rbcodec/dsp/resample.c
index 6e7e5b7b45..0a97bdf70c 100644
--- a/lib/rbcodec/dsp/resample.c
+++ b/lib/rbcodec/dsp/resample.c
@@ -25,6 +25,7 @@
#include "fracmul.h"
#include "fixedpoint.h"
#include "dsp_proc_entry.h"
+#include "dsp_misc.h"
#include <string.h>
/**
@@ -50,9 +51,10 @@ static struct resample_data
int32_t history[2][3]; /* 08h: Last samples for interpolation (L+R)
0 = oldest, 2 = newest */
/* 20h */
- int32_t frequency; /* Virtual samplerate */
+ unsigned int frequency; /* Virtual input samplerate */
+ unsigned int frequency_out; /* Resampler output samplerate */
struct dsp_buffer resample_buf; /* Buffer descriptor for resampled data */
- int32_t *resample_out_p[2]; /* Actual output buffer pointers */
+ int32_t *resample_out_p[2]; /* Actual output buffer pointers */
} resample_data[DSP_COUNT] IBSS_ATTR;
/* Actual worker function. Implemented here or in target assembly code. */
@@ -73,14 +75,16 @@ static void resample_flush(struct dsp_proc_entry *this)
}
static bool resample_new_delta(struct resample_data *data,
- struct sample_format *format)
+ struct sample_format *format,
+ unsigned int fout)
{
- int32_t frequency = format->frequency; /* virtual samplerate */
+ unsigned int frequency = format->frequency; /* virtual samplerate */
data->frequency = frequency;
- data->delta = fp_div(frequency, NATIVE_FREQUENCY, 16);
+ data->frequency_out = fout;
+ data->delta = fp_div(frequency, fout, 16);
- if (frequency == NATIVE_FREQUENCY)
+ if (frequency == data->frequency_out)
{
/* NOTE: If fully glitch-free transistions from no resampling to
resampling are desired, history should be maintained even when
@@ -232,20 +236,23 @@ static intptr_t resample_new_format(struct dsp_proc_entry *this,
DSP_PRINT_FORMAT(DSP_PROC_RESAMPLE, *format);
- int32_t frequency = data->frequency;
+ unsigned int frequency = data->frequency;
+ unsigned int fout = dsp_get_output_frequency(dsp);
bool active = dsp_proc_active(dsp, DSP_PROC_RESAMPLE);
- if (format->frequency != frequency)
+ if ((unsigned int)format->frequency != frequency ||
+ data->frequency_out != fout)
{
- DEBUGF(" DSP_PROC_RESAMPLE- new delta\n");
- active = resample_new_delta(data, format);
+ DEBUGF(" DSP_PROC_RESAMPLE- new settings: %u %u\n",
+ format->frequency, fout);
+ active = resample_new_delta(data, format, fout);
dsp_proc_activate(dsp, DSP_PROC_RESAMPLE, active);
}
- /* Everything after us is NATIVE_FREQUENCY */
+ /* Everything after us is fout */
dst->format = *format;
- dst->format.frequency = NATIVE_FREQUENCY;
- dst->format.codec_frequency = NATIVE_FREQUENCY;
+ dst->format.frequency = fout;
+ dst->format.codec_frequency = fout;
if (active)
return PROC_NEW_FORMAT_OK;
@@ -287,8 +294,10 @@ static void INIT_ATTR resample_dsp_init(struct dsp_config *dsp,
static void INIT_ATTR resample_proc_init(struct dsp_proc_entry *this,
struct dsp_config *dsp)
{
+ struct resample_data *data = &resample_data[dsp_get_id(dsp)];
+ this->data = (intptr_t)data;
dsp_proc_set_in_place(dsp, DSP_PROC_RESAMPLE, false);
- this->data = (intptr_t)&resample_data[dsp_get_id(dsp)];
+ data->frequency_out = DSP_OUT_DEFAULT_HZ;
this->process = resample_process;
}
@@ -322,6 +331,10 @@ static intptr_t resample_configure(struct dsp_proc_entry *this,
case DSP_PROC_NEW_FORMAT:
retval = resample_new_format(this, dsp, (struct sample_format *)value);
break;
+
+ case DSP_SET_OUT_FREQUENCY:
+ dsp_proc_want_format_update(dsp, DSP_PROC_RESAMPLE);
+ break;
}
return retval;
diff --git a/lib/rbcodec/dsp/tone_controls.c b/lib/rbcodec/dsp/tone_controls.c
index e636b04b9a..4266af4d97 100644
--- a/lib/rbcodec/dsp/tone_controls.c
+++ b/lib/rbcodec/dsp/tone_controls.c
@@ -21,9 +21,11 @@
****************************************************************************/
#include "rbcodecconfig.h"
#include "platform.h"
+#include "fixedpoint.h"
#include "dsp_proc_entry.h"
#include "dsp_filter.h"
#include "tone_controls.h"
+#include "dsp_misc.h"
/* These apply to all DSP streams to remain as consistant as possible with
* the behavior of hardware tone controls */
@@ -36,32 +38,39 @@ static const unsigned int tone_treble_cutoff = 3500;
static int tone_bass = 0;
static int tone_treble = 0;
+/* Current prescaler setting */
+static int tone_prescale = 0;
+
/* Data for each DSP */
static struct dsp_filter tone_filters[DSP_COUNT] IBSS_ATTR;
+static void update_filter(int id, unsigned int fout)
+{
+ filter_bishelf_coefs(fp_div(tone_bass_cutoff, fout, 32),
+ fp_div(tone_treble_cutoff, fout, 32),
+ tone_bass, tone_treble, -tone_prescale,
+ &tone_filters[id]);
+}
+
/* Update the filters' coefficients based upon the bass/treble settings */
void tone_set_prescale(int prescale)
{
int bass = tone_bass;
int treble = tone_treble;
- struct dsp_filter tone_filter; /* Temp to hold new version */
- filter_bishelf_coefs(0xffffffff / NATIVE_FREQUENCY * tone_bass_cutoff,
- 0xffffffff / NATIVE_FREQUENCY * tone_treble_cutoff,
- bass, treble, -prescale, &tone_filter);
+ tone_prescale = prescale;
struct dsp_config *dsp;
for (int i = 0; (dsp = dsp_get_config(i)); i++)
{
- struct dsp_filter *filter = &tone_filters[i];
- filter_copy(filter, &tone_filter);
+ update_filter(i, dsp_get_output_frequency(dsp));
bool enable = bass != 0 || treble != 0;
dsp_proc_enable(dsp, DSP_PROC_TONE_CONTROLS, enable);
- if (!dsp_proc_active(dsp, DSP_PROC_TONE_CONTROLS))
+ if (enable && !dsp_proc_active(dsp, DSP_PROC_TONE_CONTROLS))
{
- filter_flush(filter); /* Going online */
+ filter_flush(&tone_filters[i]); /* Going online */
dsp_proc_activate(dsp, DSP_PROC_TONE_CONTROLS, true);
}
}
@@ -109,6 +118,10 @@ static intptr_t tone_configure(struct dsp_proc_entry *this,
case DSP_FLUSH:
filter_flush((struct dsp_filter *)this->data);
break;
+
+ case DSP_SET_OUT_FREQUENCY:
+ update_filter(dsp_get_id(dsp), value);
+ break;
}
return 0;
diff --git a/lib/rbcodec/rbcodecconfig-example.h b/lib/rbcodec/rbcodecconfig-example.h
index 7ecbc1e96f..e5da42feba 100644
--- a/lib/rbcodec/rbcodecconfig-example.h
+++ b/lib/rbcodec/rbcodecconfig-example.h
@@ -5,6 +5,10 @@
#define HAVE_SW_TONE_CONTROLS
#define HAVE_ALBUMART
#define NUM_CORES 1
+/* All the same unless a configuration option is added to warble */
+#define DSP_OUT_MIN_HZ 44100
+#define DSP_OUT_DEFAULT_HZ 44100
+#define DSP_OUT_MAX_HZ 44100
#ifndef __ASSEMBLER__
diff --git a/lib/rbcodec/test/warble.c b/lib/rbcodec/test/warble.c
index 735fa2511f..336438374c 100644
--- a/lib/rbcodec/test/warble.c
+++ b/lib/rbcodec/test/warble.c
@@ -145,7 +145,7 @@ static void write_wav_header(void)
if (use_dsp) {
channels = 2;
sample_size = 16;
- freq = NATIVE_FREQUENCY;
+ freq = dsp_get_output_frequency(ci.dsp);
type = WAVE_FORMAT_PCM;
} else {
channels = format.channels;
@@ -312,7 +312,7 @@ static void playback_start(void)
{
playback_running = true;
SDL_AudioSpec spec = {0};
- spec.freq = NATIVE_FREQUENCY;
+ spec.freq = dsp_get_output_frequency(ci.dsp);
spec.format = AUDIO_S16SYS;
spec.channels = 2;
spec.samples = 0x400;
@@ -776,6 +776,7 @@ static void decode_file(const char *input_fn)
ci.id3 = &id3;
if (use_dsp) {
ci.dsp = dsp_get_config(CODEC_IDX_AUDIO);
+ dsp_configure(ci.dsp, DSP_SET_OUT_FREQUENCY, DSP_OUT_DEFAULT_HZ);
dsp_configure(ci.dsp, DSP_RESET, 0);
dsp_dither_enable(false);
}