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authorMichael Sevakis <jethead71@rockbox.org>2013-04-05 04:36:05 -0400
committerMichael Sevakis <jethead71@rockbox.org>2013-04-11 22:55:16 +0200
commitf5a5b946867677de76c405ee72e2ea47e36e4c83 (patch)
tree8fb97a35059a16681b726973b4a5e13d41f96a35 /firmware
parenta9049a79d706dba61837ad02c7d7e3475cb6c193 (diff)
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Implement universal in-PCM-driver software volume control.
Implements double-buffered volume, balance and prescaling control in the main PCM driver when HAVE_SW_VOLUME_CONTROL is defined ensuring that all PCM is volume controlled and level changes are low in latency. Supports -73 to +6 dB using a 15-bit factor so that no large-integer math is needed. Low-level hardware drivers do not have to implement it themselves but parameters can be changed (currently defined in pcm-internal.h) to work best with a particular SoC or to provide different volume ranges. Volume and prescale calls should be made in the codec driver. It should appear as a normal hardware interface. PCM volume calls expect .1 dB units. Change-Id: Idf6316a64ef4fb8abcede10707e1e6c6d01d57db Reviewed-on: http://gerrit.rockbox.org/423 Reviewed-by: Michael Sevakis <jethead71@rockbox.org> Tested-by: Michael Sevakis <jethead71@rockbox.org>
Diffstat (limited to 'firmware')
-rw-r--r--firmware/SOURCES3
-rw-r--r--firmware/export/config/ondavx747.h5
-rw-r--r--firmware/export/config/ondavx777.h5
-rw-r--r--firmware/export/jz4740-codec.h2
-rw-r--r--firmware/export/pcm-internal.h54
-rw-r--r--firmware/export/pcm_sw_volume.h40
-rw-r--r--firmware/export/sound.h3
-rw-r--r--firmware/pcm.c195
-rw-r--r--firmware/pcm_sw_volume.c264
-rw-r--r--firmware/sound.c15
-rw-r--r--firmware/target/mips/ingenic_jz47xx/codec-jz4740.c27
11 files changed, 478 insertions, 135 deletions
diff --git a/firmware/SOURCES b/firmware/SOURCES
index 92b2f5f87b..964d57ff5d 100644
--- a/firmware/SOURCES
+++ b/firmware/SOURCES
@@ -355,6 +355,9 @@ sound.c
pcm_sampr.c
pcm.c
pcm_mixer.c
+#ifdef HAVE_SW_VOLUME_CONTROL
+pcm_sw_volume.c
+#endif /* HAVE_SW_VOLUME_CONTROL */
#ifdef HAVE_RECORDING
enc_base.c
#endif /* HAVE_RECORDING */
diff --git a/firmware/export/config/ondavx747.h b/firmware/export/config/ondavx747.h
index d303ea5925..8499c15ce9 100644
--- a/firmware/export/config/ondavx747.h
+++ b/firmware/export/config/ondavx747.h
@@ -132,11 +132,6 @@
/* has no volume control, so we use the software ones */
#define HAVE_SW_VOLUME_CONTROL
-/* software controlled volume ranges from -73 -> 0 dB, other than that
- is controlled by hardware */
-#define SW_VOLUME_MIN -73
-#define SW_VOLUME_MAX 0
-
/* define the bitmask of hardware sample rates */
#define HW_SAMPR_CAPS (SAMPR_CAP_48 | SAMPR_CAP_44 | SAMPR_CAP_32 | \
SAMPR_CAP_24 | SAMPR_CAP_22 | SAMPR_CAP_16 | \
diff --git a/firmware/export/config/ondavx777.h b/firmware/export/config/ondavx777.h
index 33bf6442af..a254b0177c 100644
--- a/firmware/export/config/ondavx777.h
+++ b/firmware/export/config/ondavx777.h
@@ -126,11 +126,6 @@
/* has no volume control, so we use the software ones */
#define HAVE_SW_VOLUME_CONTROL
-/* software controlled volume ranges from -73 -> 0 dB, other than that
- is controlled by hardware */
-#define SW_VOLUME_MIN -73
-#define SW_VOLUME_MAX 0
-
/* define the bitmask of hardware sample rates */
#define HW_SAMPR_CAPS (SAMPR_CAP_48 | SAMPR_CAP_44 | SAMPR_CAP_32 | \
SAMPR_CAP_24 | SAMPR_CAP_22 | SAMPR_CAP_16 | \
diff --git a/firmware/export/jz4740-codec.h b/firmware/export/jz4740-codec.h
index 37d2347f5b..3c088f5bf7 100644
--- a/firmware/export/jz4740-codec.h
+++ b/firmware/export/jz4740-codec.h
@@ -24,6 +24,6 @@
#define VOLUME_MIN -730
#define VOLUME_MAX 60
-void audiohw_set_volume(int v);
+void audiohw_set_master_vol(int vol_l, int vol_r);
#endif /* __JZ4740_CODEC_H_ */
diff --git a/firmware/export/pcm-internal.h b/firmware/export/pcm-internal.h
index 397cf6832f..03e5c5e6e7 100644
--- a/firmware/export/pcm-internal.h
+++ b/firmware/export/pcm-internal.h
@@ -24,6 +24,19 @@
#include "config.h"
+#ifdef HAVE_SW_VOLUME_CONTROL
+/* Default settings - architecture may have other optimal values */
+
+#define PCM_FACTOR_BITS 15 /* Allows -73 to +6dB gain, sans 64-bit math */
+#define PCM_PLAY_DBL_BUF_SAMPLES 1024 /* Max 4KByte chunks */
+#define PCM_DBL_BUF_BSS /* In DRAM, uncached may be better */
+#define PCM_FACTOR_MIN 0x00000 /* Minimum final factor */
+#define PCM_FACTOR_MAX 0x10000 /* Maximum final factor */
+
+#define PCM_FACTOR_UNITY (1 << PCM_FACTOR_BITS)
+#endif /* HAVE_SW_VOLUME_CONTROL */
+
+#define PCM_SAMPLE_SIZE (2 * sizeof (int16_t))
/* Cheapo buffer align macro to align to the 16-16 PCM size */
#define ALIGN_AUDIOBUF(start, size) \
({ (start) = (void *)(((uintptr_t)(start) + 3) & ~3); \
@@ -34,6 +47,23 @@ void pcm_do_peak_calculation(struct pcm_peaks *peaks, bool active,
/** The following are for internal use between pcm.c and target-
specific portion **/
+/* Call registered callback to obtain next buffer */
+static inline bool pcm_get_more_int(const void **addr, size_t *size)
+{
+ extern volatile pcm_play_callback_type pcm_callback_for_more;
+ pcm_play_callback_type get_more = pcm_callback_for_more;
+
+ if (UNLIKELY(!get_more))
+ return false;
+
+ *addr = NULL;
+ *size = 0;
+ get_more(addr, size);
+ ALIGN_AUDIOBUF(*addr, *size);
+
+ return *addr && *size;
+}
+
static FORCE_INLINE enum pcm_dma_status pcm_call_status_cb(
pcm_status_callback_type callback, enum pcm_dma_status status)
{
@@ -43,14 +73,34 @@ static FORCE_INLINE enum pcm_dma_status pcm_call_status_cb(
return callback(status);
}
-static FORCE_INLINE enum pcm_dma_status
-pcm_play_dma_status_callback(enum pcm_dma_status status)
+static FORCE_INLINE enum pcm_dma_status pcm_play_call_status_cb(
+ enum pcm_dma_status status)
{
extern enum pcm_dma_status
(* volatile pcm_play_status_callback)(enum pcm_dma_status);
return pcm_call_status_cb(pcm_play_status_callback, status);
}
+static FORCE_INLINE enum pcm_dma_status
+pcm_play_dma_status_callback(enum pcm_dma_status status)
+{
+#ifdef HAVE_SW_VOLUME_CONTROL
+ extern enum pcm_dma_status
+ pcm_play_dma_status_callback_int(enum pcm_dma_status status);
+ return pcm_play_dma_status_callback_int(status);
+#else
+ return pcm_play_call_status_cb(status);
+#endif /* HAVE_SW_VOLUME_CONTROL */
+}
+
+#ifdef HAVE_SW_VOLUME_CONTROL
+void pcm_play_dma_start_int(const void *addr, size_t size);
+void pcm_play_dma_pause_int(bool pause);
+void pcm_play_dma_stop_int(void);
+void pcm_play_stop_int(void);
+const void *pcm_play_dma_get_peak_buffer_int(int *count);
+#endif /* HAVE_SW_VOLUME_CONTROL */
+
/* Called by the bottom layer ISR when more data is needed. Returns true
* if a new buffer is available, false otherwise. */
bool pcm_play_dma_complete_callback(enum pcm_dma_status status,
diff --git a/firmware/export/pcm_sw_volume.h b/firmware/export/pcm_sw_volume.h
new file mode 100644
index 0000000000..b86e78f500
--- /dev/null
+++ b/firmware/export/pcm_sw_volume.h
@@ -0,0 +1,40 @@
+/***************************************************************************
+ * __________ __ ___.
+ * Open \______ \ ____ ____ | | _\_ |__ _______ ___
+ * Source | _// _ \_/ ___\| |/ /| __ \ / _ \ \/ /
+ * Jukebox | | ( <_> ) \___| < | \_\ ( <_> > < <
+ * Firmware |____|_ /\____/ \___ >__|_ \|___ /\____/__/\_ \
+ * \/ \/ \/ \/ \/
+ * $Id$
+ *
+ * Copyright (C) 2013 by Michael Sevakis
+ *
+ * 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.
+ *
+ ****************************************************************************/
+#ifndef PCM_SW_VOLUME_H
+#define PCM_SW_VOLUME_H
+
+#ifdef HAVE_SW_VOLUME_CONTROL
+
+#include <audiohw.h>
+
+#define PCM_MUTE_LEVEL INT_MIN
+
+#ifdef AUDIOHW_HAVE_PRESCALER
+/* Set the prescaler value for all PCM playback */
+void pcm_set_prescaler(int prescale);
+#endif /* AUDIOHW_HAVE_PRESCALER */
+
+/* Set the per-channel volume cut/gain for all PCM playback */
+void pcm_set_master_volume(int vol_l, int vol_r);
+
+#endif /* HAVE_SW_VOLUME_CONTROL */
+
+#endif /* PCM_SW_VOLUME_H */
diff --git a/firmware/export/sound.h b/firmware/export/sound.h
index ba6120ce8f..ebf728c7c7 100644
--- a/firmware/export/sound.h
+++ b/firmware/export/sound.h
@@ -32,9 +32,6 @@ enum {
DSP_CALLBACK_SET_TREBLE,
DSP_CALLBACK_SET_CHANNEL_CONFIG,
DSP_CALLBACK_SET_STEREO_WIDTH,
-#ifdef HAVE_SW_VOLUME_CONTROL
- DSP_CALLBACK_SET_SW_VOLUME,
-#endif
};
#endif
diff --git a/firmware/pcm.c b/firmware/pcm.c
index 94b0d6eefb..6bf0e12c8d 100644
--- a/firmware/pcm.c
+++ b/firmware/pcm.c
@@ -86,7 +86,7 @@
static bool pcm_is_ready = false;
/* The registered callback function to ask for more mp3 data */
-static volatile pcm_play_callback_type
+volatile pcm_play_callback_type
pcm_callback_for_more SHAREDBSS_ATTR = NULL;
/* The registered callback function to inform of DMA status */
volatile pcm_status_callback_type
@@ -102,9 +102,89 @@ unsigned long pcm_sampr SHAREDBSS_ATTR = HW_SAMPR_DEFAULT;
/* samplerate frequency selection index */
int pcm_fsel SHAREDBSS_ATTR = HW_FREQ_DEFAULT;
-/* Called internally by functions to reset the state */
-static void pcm_play_stopped(void)
+static void pcm_play_data_start_int(const void *addr, size_t size);
+static void pcm_play_pause_int(bool play);
+void pcm_play_stop_int(void);
+
+#ifndef HAVE_SW_VOLUME_CONTROL
+/** Standard hw volume control functions - otherwise, see pcm_sw_volume.c **/
+static inline void pcm_play_dma_start_int(const void *addr, size_t size)
+{
+ pcm_play_dma_start(addr, size);
+}
+
+static inline void pcm_play_dma_pause_int(bool pause)
+{
+ if (pause || pcm_get_bytes_waiting() > 0)
+ {
+ pcm_play_dma_pause(pause);
+ }
+ else
+ {
+ logf(" no data");
+ pcm_play_data_start_int(NULL, 0);
+ }
+}
+
+static inline void pcm_play_dma_stop_int(void)
+{
+ pcm_play_dma_stop();
+}
+
+static inline const void * pcm_play_dma_get_peak_buffer_int(int *count)
+{
+ return pcm_play_dma_get_peak_buffer(count);
+}
+
+bool pcm_play_dma_complete_callback(enum pcm_dma_status status,
+ const void **addr, size_t *size)
+{
+ /* Check status callback first if error */
+ if (status < PCM_DMAST_OK)
+ status = pcm_play_dma_status_callback(status);
+
+ if (status >= PCM_DMAST_OK && pcm_get_more_int(addr, size))
+ return true;
+
+ /* Error, callback missing or no more DMA to do */
+ pcm_play_stop_int();
+ return false;
+}
+#endif /* ndef HAVE_SW_VOLUME_CONTROL */
+
+static void pcm_play_data_start_int(const void *addr, size_t size)
{
+ ALIGN_AUDIOBUF(addr, size);
+
+ if ((addr && size) || pcm_get_more_int(&addr, &size))
+ {
+ pcm_apply_settings();
+ logf(" pcm_play_dma_start_int");
+ pcm_play_dma_start_int(addr, size);
+ pcm_playing = true;
+ pcm_paused = false;
+ }
+ else
+ {
+ /* Force a stop */
+ logf(" pcm_play_stop_int");
+ pcm_play_stop_int();
+ }
+}
+
+static void pcm_play_pause_int(bool play)
+{
+ if (play)
+ pcm_apply_settings();
+
+ logf(" pcm_play_dma_pause_int");
+ pcm_play_dma_pause_int(!play);
+ pcm_paused = !play && pcm_playing;
+}
+
+void pcm_play_stop_int(void)
+{
+ pcm_play_dma_stop_int();
pcm_callback_for_more = NULL;
pcm_play_status_callback = NULL;
pcm_paused = false;
@@ -195,7 +275,7 @@ void pcm_calculate_peaks(int *left, int *right)
static struct pcm_peaks peaks;
int count;
- const void *addr = pcm_play_dma_get_peak_buffer(&count);
+ const void *addr = pcm_play_dma_get_peak_buffer_int(&count);
pcm_do_peak_calculation(&peaks, pcm_playing && !pcm_paused,
addr, count);
@@ -207,9 +287,9 @@ void pcm_calculate_peaks(int *left, int *right)
*right = peaks.right;
}
-const void* pcm_get_peak_buffer(int * count)
+const void * pcm_get_peak_buffer(int *count)
{
- return pcm_play_dma_get_peak_buffer(count);
+ return pcm_play_dma_get_peak_buffer_int(count);
}
bool pcm_is_playing(void)
@@ -233,8 +313,6 @@ void pcm_init(void)
{
logf("pcm_init");
- pcm_play_stopped();
-
pcm_set_frequency(HW_SAMPR_DEFAULT);
logf(" pcm_play_dma_init");
@@ -258,41 +336,6 @@ bool pcm_is_initialized(void)
return pcm_is_ready;
}
-/* Common code to pcm_play_data and pcm_play_pause */
-static void pcm_play_data_start(const void *addr, size_t size)
-{
- ALIGN_AUDIOBUF(addr, size);
-
- if (!(addr && size))
- {
- pcm_play_callback_type get_more = pcm_callback_for_more;
- addr = NULL;
- size = 0;
-
- if (get_more)
- {
- logf(" get_more");
- get_more(&addr, &size);
- ALIGN_AUDIOBUF(addr, size);
- }
- }
-
- if (addr && size)
- {
- logf(" pcm_play_dma_start");
- pcm_apply_settings();
- pcm_play_dma_start(addr, size);
- pcm_playing = true;
- pcm_paused = false;
- return;
- }
-
- /* Force a stop */
- logf(" pcm_play_dma_stop");
- pcm_play_dma_stop();
- pcm_play_stopped();
-}
-
void pcm_play_data(pcm_play_callback_type get_more,
pcm_status_callback_type status_cb,
const void *start, size_t size)
@@ -304,41 +347,12 @@ void pcm_play_data(pcm_play_callback_type get_more,
pcm_callback_for_more = get_more;
pcm_play_status_callback = status_cb;
- logf(" pcm_play_data_start");
- pcm_play_data_start(start, size);
+ logf(" pcm_play_data_start_int");
+ pcm_play_data_start_int(start, size);
pcm_play_unlock();
}
-bool pcm_play_dma_complete_callback(enum pcm_dma_status status,
- const void **addr, size_t *size)
-{
- /* Check status callback first if error */
- if (status < PCM_DMAST_OK)
- status = pcm_play_dma_status_callback(status);
-
- pcm_play_callback_type get_more = pcm_callback_for_more;
-
- if (get_more && status >= PCM_DMAST_OK)
- {
- *addr = NULL;
- *size = 0;
-
- /* Call registered callback to obtain next buffer */
- get_more(addr, size);
- ALIGN_AUDIOBUF(*addr, *size);
-
- if (*addr && *size)
- return true;
- }
-
- /* Error, callback missing or no more DMA to do */
- pcm_play_dma_stop();
- pcm_play_stopped();
-
- return false;
-}
-
void pcm_play_pause(bool play)
{
logf("pcm_play_pause: %s", play ? "play" : "pause");
@@ -347,28 +361,8 @@ void pcm_play_pause(bool play)
if (play == pcm_paused && pcm_playing)
{
- if (!play)
- {
- logf(" pcm_play_dma_pause");
- pcm_play_dma_pause(true);
- pcm_paused = true;
- }
- else if (pcm_get_bytes_waiting() > 0)
- {
- logf(" pcm_play_dma_pause");
- pcm_apply_settings();
- pcm_play_dma_pause(false);
- pcm_paused = false;
- }
- else
- {
- logf(" pcm_play_dma_start: no data");
- pcm_play_data_start(NULL, 0);
- }
- }
- else
- {
- logf(" no change");
+ logf(" pcm_play_pause_int");
+ pcm_play_pause_int(play);
}
pcm_play_unlock();
@@ -382,13 +376,8 @@ void pcm_play_stop(void)
if (pcm_playing)
{
- logf(" pcm_play_dma_stop");
- pcm_play_dma_stop();
- pcm_play_stopped();
- }
- else
- {
- logf(" not playing");
+ logf(" pcm_play_stop_int");
+ pcm_play_stop_int();
}
pcm_play_unlock();
diff --git a/firmware/pcm_sw_volume.c b/firmware/pcm_sw_volume.c
new file mode 100644
index 0000000000..bcd498fe46
--- /dev/null
+++ b/firmware/pcm_sw_volume.c
@@ -0,0 +1,264 @@
+/***************************************************************************
+ * __________ __ ___.
+ * Open \______ \ ____ ____ | | _\_ |__ _______ ___
+ * Source | _// _ \_/ ___\| |/ /| __ \ / _ \ \/ /
+ * Jukebox | | ( <_> ) \___| < | \_\ ( <_> > < <
+ * Firmware |____|_ /\____/ \___ >__|_ \|___ /\____/__/\_ \
+ * \/ \/ \/ \/ \/
+ * $Id$
+ *
+ * Copyright (C) 2013 by Michael Sevakis
+ *
+ * This program is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU General Public License
+ * as published by the Free Software Foundation; either version 2
+ * of the License, or (at your option) any later version.
+ *
+ * This software is distributed on an "AS IS" basis, WITHOUT WARRANTY OF ANY
+ * KIND, either express or implied.
+ *
+ ****************************************************************************/
+#include "config.h"
+#include "system.h"
+#include "pcm.h"
+#include "pcm-internal.h"
+#include "dsp-util.h"
+#include "fixedpoint.h"
+#include "pcm_sw_volume.h"
+
+/* source buffer from client */
+static const void * volatile src_buf_addr = NULL;
+static size_t volatile src_buf_rem = 0;
+
+#define PCM_PLAY_DBL_BUF_SIZE (PCM_PLAY_DBL_BUF_SAMPLE*PCM_SAMPLE_SIZE)
+
+/* double buffer and frame length control */
+static int16_t pcm_dbl_buf[2][PCM_PLAY_DBL_BUF_SAMPLES*2]
+ PCM_DBL_BUF_BSS MEM_ALIGN_ATTR;
+static size_t pcm_dbl_buf_size[2];
+static int pcm_dbl_buf_num = 0;
+static size_t frame_size;
+static unsigned int frame_count, frame_err, frame_frac;
+
+#ifdef AUDIOHW_HAVE_PRESCALER
+static int32_t prescale_factor = PCM_FACTOR_UNITY;
+static int32_t vol_factor_l = 0, vol_factor_r = 0;
+#endif /* AUDIOHW_HAVE_PRESCALER */
+
+/* pcm scaling factors */
+static int32_t pcm_factor_l = 0, pcm_factor_r = 0;
+
+#define PCM_FACTOR_CLIP(f) \
+ MAX(MIN((f), PCM_FACTOR_MAX), PCM_FACTOR_MIN)
+#define PCM_SCALE_SAMPLE(f, s) \
+ (((f) * (s) + PCM_FACTOR_UNITY/2) >> PCM_FACTOR_BITS)
+
+
+/* TODO: #include CPU-optimized routines and move this to /firmware/asm */
+static inline void pcm_copy_buffer(int16_t *dst, const int16_t *src,
+ size_t size)
+{
+ int32_t factor_l = pcm_factor_l;
+ int32_t factor_r = pcm_factor_r;
+
+ if (LIKELY(factor_l <= PCM_FACTOR_UNITY && factor_r <= PCM_FACTOR_UNITY))
+ {
+ /* All cut or unity */
+ while (size)
+ {
+ *dst++ = PCM_SCALE_SAMPLE(factor_l, *src++);
+ *dst++ = PCM_SCALE_SAMPLE(factor_r, *src++);
+ size -= PCM_SAMPLE_SIZE;
+ }
+ }
+ else
+ {
+ /* Any positive gain requires clipping */
+ while (size)
+ {
+ *dst++ = clip_sample_16(PCM_SCALE_SAMPLE(factor_l, *src++));
+ *dst++ = clip_sample_16(PCM_SCALE_SAMPLE(factor_r, *src++));
+ size -= PCM_SAMPLE_SIZE;
+ }
+ }
+}
+
+bool pcm_play_dma_complete_callback(enum pcm_dma_status status,
+ const void **addr, size_t *size)
+{
+ /* Check status callback first if error */
+ if (status < PCM_DMAST_OK)
+ status = pcm_play_call_status_cb(status);
+
+ size_t sz = pcm_dbl_buf_size[pcm_dbl_buf_num];
+
+ if (status >= PCM_DMAST_OK && sz)
+ {
+ /* Do next chunk */
+ *addr = pcm_dbl_buf[pcm_dbl_buf_num];
+ *size = sz;
+ return true;
+ }
+ else
+ {
+ /* This is a stop chunk or error */
+ pcm_play_stop_int();
+ return false;
+ }
+}
+
+/* Equitably divide large source buffers amongst double buffer frames;
+ frames smaller than or equal to the double buffer chunk size will play
+ in one chunk */
+static void update_frame_params(size_t size)
+{
+ int count = size / PCM_SAMPLE_SIZE;
+ frame_count = (count + PCM_PLAY_DBL_BUF_SAMPLES - 1) /
+ PCM_PLAY_DBL_BUF_SAMPLES;
+ int perframe = count / frame_count;
+ frame_size = perframe * PCM_SAMPLE_SIZE;
+ frame_frac = count - perframe * frame_count;
+ frame_err = 0;
+}
+
+/* Obtain the next buffer and prepare it for pcm driver playback */
+enum pcm_dma_status
+pcm_play_dma_status_callback_int(enum pcm_dma_status status)
+{
+ if (status != PCM_DMAST_STARTED)
+ return status;
+
+ size_t size = pcm_dbl_buf_size[pcm_dbl_buf_num];
+ const void *addr = src_buf_addr + size;
+
+ size = src_buf_rem - size;
+
+ if (size == 0 && pcm_get_more_int(&addr, &size))
+ {
+ update_frame_params(size);
+ pcm_play_call_status_cb(PCM_DMAST_STARTED);
+ }
+
+ src_buf_addr = addr;
+ src_buf_rem = size;
+
+ if (size != 0)
+ {
+ size = frame_size;
+
+ if ((frame_err += frame_frac) >= frame_count)
+ {
+ frame_err -= frame_count;
+ size += PCM_SAMPLE_SIZE;
+ }
+ }
+
+ pcm_dbl_buf_num ^= 1;
+ pcm_dbl_buf_size[pcm_dbl_buf_num] = size;
+ pcm_copy_buffer(pcm_dbl_buf[pcm_dbl_buf_num], addr, size);
+
+ return PCM_DMAST_OK;
+}
+
+/* Prefill double buffer and start pcm driver */
+static void start_pcm(bool reframe)
+{
+ pcm_dbl_buf_num = 0;
+ pcm_dbl_buf_size[0] = 0;
+
+ if (reframe)
+ update_frame_params(src_buf_rem);
+
+ pcm_play_dma_status_callback(PCM_DMAST_STARTED);
+ pcm_play_dma_status_callback(PCM_DMAST_STARTED);
+
+ pcm_play_dma_start(pcm_dbl_buf[1], pcm_dbl_buf_size[1]);
+}
+
+void pcm_play_dma_start_int(const void *addr, size_t size)
+{
+ src_buf_addr = addr;
+ src_buf_rem = size;
+ start_pcm(true);
+}
+
+void pcm_play_dma_pause_int(bool pause)
+{
+ if (pause)
+ pcm_play_dma_pause(true);
+ else if (src_buf_rem)
+ start_pcm(false); /* Reprocess in case volume level changed */
+ else
+ pcm_play_stop_int(); /* Playing frame was last frame */
+}
+
+void pcm_play_dma_stop_int(void)
+{
+ pcm_play_dma_stop();
+ src_buf_addr = NULL;
+ src_buf_rem = 0;
+}
+
+/* Return playing buffer from the source buffer */
+const void * pcm_play_dma_get_peak_buffer_int(int *count)
+{
+ const void *addr = src_buf_addr;
+ size_t size = src_buf_rem;
+ const void *addr2 = src_buf_addr;
+
+ if (addr == addr2 && size)
+ {
+ *count = size / PCM_SAMPLE_SIZE;
+ return addr;
+ }
+
+ *count = 0;
+ return NULL;
+}
+
+/* Return the scale factor corresponding to the centibel level */
+static int32_t pcm_centibels_to_factor(int volume)
+{
+ if (volume == PCM_MUTE_LEVEL)
+ return 0; /* mute */
+
+ /* Centibels -> fixedpoint */
+ return fp_factor(PCM_FACTOR_UNITY*volume / 10, PCM_FACTOR_BITS);
+}
+
+#ifdef AUDIOHW_HAVE_PRESCALER
+/* Produce final pcm scale factor */
+static void pcm_sync_prescaler(void)
+{
+ int32_t factor_l = fp_mul(prescale_factor, vol_factor_l, PCM_FACTOR_BITS);
+ int32_t factor_r = fp_mul(prescale_factor, vol_factor_r, PCM_FACTOR_BITS);
+ pcm_factor_l = PCM_FACTOR_CLIP(factor_l);
+ pcm_factor_r = PCM_FACTOR_CLIP(factor_r);
+}
+
+/* Set the prescaler value for all PCM playback */
+void pcm_set_prescaler(int prescale)
+{
+ prescale_factor = pcm_centibels_to_factor(-prescale);
+ pcm_sync_prescaler();
+}
+
+/* Set the per-channel volume cut/gain for all PCM playback */
+void pcm_set_master_volume(int vol_l, int vol_r)
+{
+ vol_factor_l = pcm_centibels_to_factor(vol_l);
+ vol_factor_r = pcm_centibels_to_factor(vol_r);
+ pcm_sync_prescaler();
+}
+
+#else /* ndef AUDIOHW_HAVE_PRESCALER */
+
+/* Set the per-channel volume cut/gain for all PCM playback */
+void pcm_set_master_volume(int vol_l, int vol_r)
+{
+ int32_t factor_l = pcm_centibels_to_factor(vol_l);
+ int32_t factor_r = pcm_centibels_to_factor(vol_r);
+ pcm_factor_l = PCM_FACTOR_CLIP(factor_l);
+ pcm_factor_r = PCM_FACTOR_CLIP(factor_r);
+}
+#endif /* AUDIOHW_HAVE_PRESCALER */
diff --git a/firmware/sound.c b/firmware/sound.c
index e442e87f8c..2ffef0e72b 100644
--- a/firmware/sound.c
+++ b/firmware/sound.c
@@ -26,6 +26,9 @@
#include "logf.h"
#include "system.h"
#include "i2c.h"
+#ifdef HAVE_SW_VOLUME_CONTROL
+#include "pcm_sw_volume.h"
+#endif /* HAVE_SW_VOLUME_CONTROL */
/* TODO
* find a nice way to handle 1.5db steps -> see wm8751 ifdef in sound_set_bass/treble
@@ -215,7 +218,7 @@ static void set_prescaled_volume(void)
dsp_callback(DSP_CALLBACK_SET_PRESCALE, prescale);
#endif
- if (current_volume == VOLUME_MIN)
+ if (current_volume <= VOLUME_MIN)
prescale = 0; /* Make sure the chip gets muted at VOLUME_MIN */
l = r = current_volume + prescale;
@@ -231,13 +234,11 @@ static void set_prescaled_volume(void)
r += ((r - (VOLUME_MIN - ONE_DB)) * current_balance) / VOLUME_RANGE;
}
-#ifdef HAVE_SW_VOLUME_CONTROL
- dsp_callback(DSP_CALLBACK_SET_SW_VOLUME, 0);
-#endif
-
/* ypr0 with sdl has separate volume controls */
#if !defined(HAVE_SDL_AUDIO) || defined(SAMSUNG_YPR0)
-#if CONFIG_CODEC == MAS3507D
+#if defined(HAVE_SW_VOLUME_CONTROL) || defined(HAVE_JZ4740_CODEC)
+ audiohw_set_master_vol(l, r);
+#elif CONFIG_CODEC == MAS3507D
dac_volume(tenthdb2reg(l), tenthdb2reg(r), false);
#elif defined(HAVE_UDA1380) || defined(HAVE_WM8975) || defined(HAVE_WM8758) \
|| defined(HAVE_WM8711) || defined(HAVE_WM8721) || defined(HAVE_WM8731) \
@@ -253,7 +254,7 @@ static void set_prescaled_volume(void)
#elif defined(HAVE_TLV320) || defined(HAVE_WM8978) || defined(HAVE_WM8985) || defined(HAVE_IMX233_CODEC) || defined(HAVE_AIC3X)
audiohw_set_headphone_vol(tenthdb2master(l), tenthdb2master(r));
-#elif defined(HAVE_JZ4740_CODEC) || defined(HAVE_SDL_AUDIO) || defined(ANDROID)
+#elif defined(HAVE_SDL_AUDIO) || defined(ANDROID)
audiohw_set_volume(current_volume);
#endif
#else /* HAVE_SDL_AUDIO */
diff --git a/firmware/target/mips/ingenic_jz47xx/codec-jz4740.c b/firmware/target/mips/ingenic_jz47xx/codec-jz4740.c
index ab9efc91b0..065433e12a 100644
--- a/firmware/target/mips/ingenic_jz47xx/codec-jz4740.c
+++ b/firmware/target/mips/ingenic_jz47xx/codec-jz4740.c
@@ -24,11 +24,12 @@
#include "sound.h"
#include "jz4740.h"
#include "system.h"
+#include "pcm_sw_volume.h"
/* TODO */
const struct sound_settings_info audiohw_settings[] = {
#ifdef HAVE_SW_VOLUME_CONTROL
- [SOUND_VOLUME] = {"dB", 0, 1, SW_VOLUME_MIN, 6, 0},
+ [SOUND_VOLUME] = {"dB", 0, 1, -74, 6, -25},
#else
[SOUND_VOLUME] = {"dB", 0, 1, 0, 6, 0},
#endif
@@ -293,16 +294,24 @@ void audiohw_init(void)
i2s_codec_init();
}
-void audiohw_set_volume(int v)
+void audiohw_set_master_vol(int vol_l, int vol_r)
{
- if(v >= 0)
- {
- /* 0 <= v <= 60 */
- unsigned int codec_volume = ICDC_CDCCR2_HPVOL(v / 20);
+#ifdef HAVE_SW_VOLUME_CONTROL
+ /* SW volume for <= 1.0 gain, HW at unity, < VOLUME_MIN == MUTE */
+ int sw_volume_l = vol_l < VOLUME_MIN ? PCM_MUTE_LEVEL : MIN(vol_l, 0);
+ int sw_volume_r = vol_r < VOLUME_MIN ? PCM_MUTE_LEVEL : MIN(vol_r, 0);
+ pcm_set_master_volume(sw_volume_l, sw_volume_r);
+#endif /* HAVE_SW_VOLUME_CONTROL */
- if((REG_ICDC_CDCCR2 & ICDC_CDCCR2_HPVOL(0x3)) != codec_volume)
- REG_ICDC_CDCCR2 = (REG_ICDC_CDCCR2 & ~ICDC_CDCCR2_HPVOL(0x3)) | codec_volume;
- }
+ /* NOTE: the channel being cut if balance is not equal will need
+ adjusting downward so maintain proportion if using volume boost */
+
+ /* HW volume for > 1.0 gain */
+ int v = MAX(vol_l, vol_r);
+ unsigned int hw_volume = v > 0 ? ICDC_CDCCR2_HPVOL(v / 20) : 0;
+
+ if((REG_ICDC_CDCCR2 & ICDC_CDCCR2_HPVOL(0x3)) != hw_volume)
+ REG_ICDC_CDCCR2 = (REG_ICDC_CDCCR2 & ~ICDC_CDCCR2_HPVOL(0x3)) | hw_volume;
}
void audiohw_set_frequency(int freq)