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>

13 files changed, 478 insertions, 156 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)
diff --git a/lib/rbcodec/dsp/dsp_misc.c b/lib/rbcodec/dsp/dsp_misc.c
index 1083215c17..a8c9423cfb 100644
--- a/lib/rbcodec/dsp/dsp_misc.c
+++ b/lib/rbcodec/dsp/dsp_misc.c
@@ -35,11 +35,6 @@
 #endif
 #include <string.h>
 
-#if defined(HAVE_SW_TONE_CONTROLS) && defined(HAVE_SW_VOLUME_CONTROL)
-/* Still need this for volume control */
-#include "settings.h"
-#endif
-
 /** Firmware callback interface **/
 
 /* Hook back from firmware/ part of audio, which can't/shouldn't call apps/
@@ -58,19 +53,6 @@ int dsp_callback(int msg, intptr_t param)
     case DSP_CALLBACK_SET_TREBLE:
         tone_set_treble(param);
         break;
-    /* FIXME: This must be done by bottom-level PCM driver so it works with
-              all PCM, not here and not in mixer. I won't fully support it
-              here with all streams. -- jethead71 */
-#ifdef HAVE_SW_VOLUME_CONTROL
-    case DSP_CALLBACK_SET_SW_VOLUME:
-        if (global_settings.volume < SW_VOLUME_MAX ||
-            global_settings.volume > SW_VOLUME_MIN)
-        {
-            int vol_gain = get_replaygain_int(global_settings.volume * 100);
-            pga_set_gain(PGA_VOLUME, vol_gain);
-        }
-        break;
-#endif /* HAVE_SW_VOLUME_CONTROL */
 #endif /* HAVE_SW_TONE_CONTROLS */
     case DSP_CALLBACK_SET_CHANNEL_CONFIG:
         channel_mode_set_config(param);
diff --git a/lib/rbcodec/dsp/pga.h b/lib/rbcodec/dsp/pga.h
index f0c4c4717f..c05265873b 100644
--- a/lib/rbcodec/dsp/pga.h
+++ b/lib/rbcodec/dsp/pga.h
@@ -28,9 +28,6 @@ enum pga_gain_ids
 {
     PGA_EQ_PRECUT = 0,
     PGA_REPLAYGAIN,
-#ifdef HAVE_SW_VOLUME_CONTROL
-    PGA_VOLUME,
-#endif
     PGA_NUM_GAINS,
 };