Commit FS#12150 - Fully-functional audio mixer - and finally whip old limitations about playback of voice and other sounds when paused. Channels are independent in state and amplitude. Fade on stop/pause is handled by the channel's volume control rather than global volume which means it now works from anywhere. Opens up the possibility of plugin sounds during music playback by merely adding an additional channel enum. If any PCM drivers were not properly modified, see one of the last comments in the task for a description of the simple change that is expected. Some params are tunable in firmware/export/pcm-mixer.h as well.

git-svn-id: svn://svn.rockbox.org/rockbox/trunk@30097 a1c6a512-1295-4272-9138-f99709370657

21 files changed, 754 insertions, 25 deletions

diff --git a/firmware/target/arm/as3525/pcm-as3525.c b/firmware/target/arm/as3525/pcm-as3525.c
index 469833b05c..1b22d48f7f 100644
--- a/firmware/target/arm/as3525/pcm-as3525.c
+++ b/firmware/target/arm/as3525/pcm-as3525.c
@@ -29,6 +29,7 @@
 #include "as3514.h"
 #include "audiohw.h"
 #include "mmu-arm.h"
+#include "pcm-internal.h"
 
 #define MAX_TRANSFER (4*((1<<11)-1)) /* maximum data we can transfer via DMA
                                       * i.e. 32 bits at once (size of I2SO_DATA)
@@ -104,9 +105,13 @@ static void dma_callback(void)
 
         /* force writeback */
         clean_dcache_range(dma_start_addr, dma_start_size);
+        play_start_pcm();
+        pcm_play_dma_started_callback();
+    }
+    else
+    {
+        play_start_pcm();
     }
-
-    play_start_pcm();
 }
 
 void pcm_play_dma_start(const void *addr, size_t size)
diff --git a/firmware/target/arm/imx31/gigabeat-s/pcm-gigabeat-s.c b/firmware/target/arm/imx31/gigabeat-s/pcm-gigabeat-s.c
index c8c1283d12..1f6eef435a 100644
--- a/firmware/target/arm/imx31/gigabeat-s/pcm-gigabeat-s.c
+++ b/firmware/target/arm/imx31/gigabeat-s/pcm-gigabeat-s.c
@@ -26,6 +26,7 @@
 #include "ccm-imx31.h"
 #include "sdma-imx31.h"
 #include "mmu-imx31.h"
+#include "pcm-internal.h"
 
 #define DMA_PLAY_CH_NUM 2
 #define DMA_REC_CH_NUM 1
@@ -105,6 +106,8 @@ static void play_dma_callback(void)
     dma_play_bd.mode.command = TRANSFER_16BIT;
     dma_play_bd.mode.status = BD_DONE | BD_WRAP | BD_INTR;
     sdma_channel_run(DMA_PLAY_CH_NUM);
+
+    pcm_play_dma_started_callback();
 }
 
 void pcm_play_lock(void)
diff --git a/firmware/target/arm/pcm-mixer-armv4.c b/firmware/target/arm/pcm-mixer-armv4.c
new file mode 100644
index 0000000000..4818544d7b
--- /dev/null
+++ b/firmware/target/arm/pcm-mixer-armv4.c
@@ -0,0 +1,182 @@
+/***************************************************************************
+ *             __________               __   ___.
+ *   Open      \______   \ ____   ____ |  | _\_ |__   _______  ___
+ *   Source     |       _//  _ \_/ ___\|  |/ /| __ \ /  _ \  \/  /
+ *   Jukebox    |    |   (  <_> )  \___|    < | \_\ (  <_> > <  <
+ *   Firmware   |____|_  /\____/ \___  >__|_ \|___  /\____/__/\_ \
+ *                     \/            \/     \/    \/            \/
+ * $Id$
+ *
+ * Copyright (C) 2011 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.
+ *
+ ****************************************************************************/
+
+#define MIXER_OPTIMIZED_WRITE_SAMPLES
+#define MIXER_OPTIMIZED_MIX_SAMPLES
+
+/* Mix channels' samples and apply gain factors */
+static FORCE_INLINE void mix_samples(void *out,
+                                     void *src0,
+                                     int32_t src0_amp,
+                                     void *src1,
+                                     int32_t src1_amp,
+                                     size_t size)
+{
+    if (src0_amp == MIX_AMP_UNITY && src1_amp == MIX_AMP_UNITY)
+    {
+        /* Both are unity amplitude */
+        int32_t l0, l1, h0, h1;
+        asm volatile (
+        "1:                             \n"
+            "ldrsh  %4, [%1], #2        \n"
+            "ldrsh  %5, [%2], #2        \n"
+            "ldrsh  %6, [%1], #2        \n"
+            "ldrsh  %7, [%2], #2        \n"
+            "add    %4, %4, %5          \n"
+            "add    %6, %6, %7          \n"
+            "mov    %5, %4, asr #15     \n"
+            "teq    %5, %5, asr #31     \n"
+            "eorne  %4, %8, %4, asr #31 \n"
+            "mov    %7, %6, asr #15     \n"
+            "teq    %7, %7, asr #31     \n"
+            "eorne  %6, %8, %6, asr #31 \n"
+            "subs   %3, %3, #4          \n"
+            "and    %4, %4, %8, lsr #16 \n"
+            "orr    %6, %4, %6, lsl #16 \n"
+            "str    %6, [%0], #4        \n"
+            "bhi    1b                  \n"
+            : "+r"(out), "+r"(src0), "+r"(src1), "+r"(size),
+              "=&r"(l0), "=&r"(l1), "=&r"(h0), "=&r"(h1)
+            : "r"(0xffff7fff));
+    }
+    else if (src0_amp != MIX_AMP_UNITY && src1_amp != MIX_AMP_UNITY)
+    {
+        /* Neither are unity amplitude */
+        int32_t l0, l1, h0, h1;
+        asm volatile (
+        "1:                              \n"
+            "ldrsh  %4, [%1], #2         \n"
+            "ldrsh  %5, [%2], #2         \n"
+            "ldrsh  %6, [%1], #2         \n"
+            "ldrsh  %7, [%2], #2         \n"
+            "mul    %4, %8, %4           \n"
+            "mul    %5, %9, %5           \n"
+            "mul    %6, %8, %6           \n"
+            "mul    %7, %9, %7           \n"
+            "mov    %4, %4, asr #16      \n"
+            "add    %4, %4, %5, asr #16  \n"
+            "mov    %6, %6, asr #16      \n"
+            "add    %6, %6, %7, asr #16  \n"
+            "mov    %5, %4, asr #15      \n"
+            "teq    %5, %5, asr #31      \n"
+            "eorne  %4, %10, %4, asr #31 \n"
+            "mov    %7, %6, asr #15      \n"
+            "teq    %7, %7, asr #31      \n"
+            "eorne  %6, %10, %6, asr #31 \n"
+            "subs   %3, %3, #4           \n"
+            "and    %4, %4, %10, lsr #16 \n"
+            "orr    %6, %4, %6, lsl #16  \n"
+            "str    %6, [%0], #4         \n"
+            "bhi    1b                   \n"
+            : "+r"(out), "+r"(src0), "+r"(src1), "+r"(size),
+              "=&r"(l0), "=&r"(l1), "=&r"(h0), "=&r"(h1)
+            : "r"(src0_amp), "r"(src1_amp), "r"(0xffff7fff));
+    }
+    else
+    {
+        /* One is unity amplitude */
+        if (src0_amp != MIX_AMP_UNITY)
+        {
+            /* Keep unity in src0, amp0 */
+            int16_t *src_tmp = src0;
+            src0 = src1;
+            src1 = src_tmp;
+            src1_amp = src0_amp;
+            src0_amp = MIX_AMP_UNITY;
+        }
+
+        int32_t l0, l1, h0, h1;
+        asm volatile (
+        "1:                             \n"
+            "ldrsh  %4, [%1], #2        \n"
+            "ldrsh  %5, [%2], #2        \n"
+            "ldrsh  %6, [%1], #2        \n"
+            "ldrsh  %7, [%2], #2        \n"
+            "mul    %5, %8, %5          \n"
+            "mul    %7, %8, %7          \n"
+            "add    %4, %4, %5, asr #16 \n"
+            "add    %6, %6, %7, asr #16 \n"
+            "mov    %5, %4, asr #15     \n"
+            "teq    %5, %5, asr #31     \n"
+            "eorne  %4, %9, %4, asr #31 \n"
+            "mov    %7, %6, asr #15     \n"
+            "teq    %7, %7, asr #31     \n"
+            "eorne  %6, %9, %6, asr #31 \n"
+            "subs   %3, %3, #4          \n"
+            "and    %4, %4, %9, lsr #16 \n"
+            "orr    %6, %4, %6, lsl #16 \n"
+            "str    %6, [%0], #4        \n"
+            "bhi    1b                  \n"
+            : "+r"(out), "+r"(src0), "+r"(src1), "+r"(size),
+              "=&r"(l0), "=&r"(l1), "=&r"(h0), "=&r"(h1)
+            : "r"(src1_amp), "r"(0xffff7fff));
+    }
+}
+
+/* Write channel's samples and apply gain factor */
+static FORCE_INLINE void write_samples(void *out,
+                                       void *src,
+                                       int32_t amp,
+                                       size_t size)
+{
+    if (LIKELY(amp == MIX_AMP_UNITY))
+    {
+        /* Channel is unity amplitude */
+        asm volatile (
+            "ands       r1, %2, #0x1f  \n"
+            "beq        2f             \n"
+        "1:                            \n"
+            "ldr        r0, [%1], #4   \n"
+            "subs       r1, r1, #4     \n"
+            "str        r0, [%0], #4   \n"
+            "bne        1b             \n"
+            "bics       %2, %2, #0x1f  \n"
+            "beq        3f             \n"
+        "2:                            \n"
+            "ldmia      %1!, { r0-r7 } \n"
+            "subs       %2, %2, #32    \n"
+            "stmia      %0!, { r0-r7 } \n"
+            "bhi        2b             \n"
+        "3:                            \n"
+            : "+r"(out), "+r"(src), "+r"(size)
+            :
+            : "r0", "r1", "r2", "r3", "r4", "r5", "r6", "r7");
+    }
+    else
+    {
+        /* Channel needs amplitude cut */
+        uint32_t l, h;
+        asm volatile (
+        "1:                              \n"
+            "ldrsh   %3, [%1], #2        \n"
+            "ldrsh   %4, [%1], #2        \n"
+            "subs    %2, %2, #4          \n"
+            "mul     %3, %5, %3          \n"
+            "mul     %4, %5, %4          \n"
+            "and     %4, %4, %6, lsl #16 \n"
+            "orr     %4, %4, %3, lsr #16 \n"
+            "str     %4, [%0], #4        \n"
+            "bhi     1b                  \n"
+            : "+r"(out), "+r"(src), "+r"(size),
+              "=&r"(l), "=&r"(h)
+            : "r"(amp), "r"(0xffffffffu));
+    }     
+}
diff --git a/firmware/target/arm/pcm-mixer-armv5.c b/firmware/target/arm/pcm-mixer-armv5.c
new file mode 100644
index 0000000000..64f2c86f52
--- /dev/null
+++ b/firmware/target/arm/pcm-mixer-armv5.c
@@ -0,0 +1,106 @@
+/***************************************************************************
+ *             __________               __   ___.
+ *   Open      \______   \ ____   ____ |  | _\_ |__   _______  ___
+ *   Source     |       _//  _ \_/ ___\|  |/ /| __ \ /  _ \  \/  /
+ *   Jukebox    |    |   (  <_> )  \___|    < | \_\ (  <_> > <  <
+ *   Firmware   |____|_  /\____/ \___  >__|_ \|___  /\____/__/\_ \
+ *                     \/            \/     \/    \/            \/
+ * $Id$
+ *
+ * Copyright (C) 2011 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.
+ *
+ ****************************************************************************/
+
+#define MIXER_OPTIMIZED_WRITE_SAMPLES
+#define MIXER_OPTIMIZED_MIX_SAMPLES
+
+/* Mix channels' samples and apply gain factors */
+static FORCE_INLINE void mix_samples(void *out,
+                                     void *src0,
+                                     int32_t src0_amp,
+                                     void *src1,
+                                     int32_t src1_amp,
+                                     size_t size)
+{
+    int32_t s0, s1, tmp;
+    asm volatile (
+    "1:                             \n"
+        "ldr    %4, [%1], #4        \n"
+        "ldr    %5, [%2], #4        \n"
+        "smulwb %6, %7, %4          \n"
+        "smulwt %4, %7, %4          \n"
+        "smlawb %6, %8, %5, %6      \n"
+        "smlawt %4, %8, %5, %4      \n"
+        "mov    %5, %6, asr #15     \n"
+        "teq    %5, %5, asr #31     \n"
+        "eorne  %6, %9, %6, asr #31 \n"
+        "mov    %5, %4, asr #15     \n"
+        "teq    %5, %5, asr #31     \n"
+        "eorne  %4, %9, %4, asr #31 \n"
+        "subs   %3, %3, #4          \n"
+        "and    %6, %6, %9, lsr #16 \n"
+        "orr    %6, %6, %4, lsl #16 \n"
+        "str    %6, [%0], #4        \n"
+        "bhi    1b                  \n"
+        : "+r"(out), "+r"(src0), "+r"(src1), "+r"(size),
+          "=&r"(s0), "=&r"(s1), "=&r"(tmp)
+        : "r"(src0_amp), "r"(src1_amp), "r"(0xffff7fff));
+}
+
+/* Write channel's samples and apply gain factor */
+static FORCE_INLINE void write_samples(void *out,
+                                       void *src,
+                                       int32_t amp,
+                                       size_t size)
+{
+    if (LIKELY(amp == MIX_AMP_UNITY))
+    {
+        /* Channel is unity amplitude */
+        asm volatile (
+            "ands   r1, %2, #0x1f  \n"
+            "beq    2f             \n"
+        "1:                        \n"
+            "ldr    r0, [%1], #4   \n"
+            "subs   r1, r1, #4     \n"
+            "str    r0, [%0], #4   \n"
+            "bne    1b             \n"
+            "bics   %2, %2, #0x1f  \n"
+            "beq    3f             \n"
+        "2:                        \n"
+            "ldmia  %1!, { r0-r7 } \n"
+            "subs   %2, %2, #32    \n"
+            "stmia  %0!, { r0-r7 } \n"
+            "bhi    2b             \n"
+        "3:                        \n"
+            : "+r"(out), "+r"(src), "+r"(size)
+            :
+            : "r0", "r1", "r2", "r3", "r4", "r5", "r6", "r7");
+    }
+    else
+    {
+        /* Channel needs amplitude cut */
+        uint32_t l, h;
+        asm volatile (
+        "1:                             \n"
+            "ldr    %3, [%1], #4        \n"
+            "subs   %2, %2, #4          \n"
+            "smulwt %4, %5, %3          \n"
+            "smulwb %3, %5, %3          \n"
+            "mov    %4, %4, lsl #16     \n"
+            "mov    %3, %3, lsl #16     \n"
+            "orr    %4, %4, %3, lsr #16 \n"
+            "str    %4, [%0], #4        \n"
+            "bhi    1b                  \n"
+            : "+r"(out), "+r"(src), "+r"(size),
+              "=&r"(l), "=&r"(h)
+            : "r"(amp));
+    }     
+}
diff --git a/firmware/target/arm/pcm-mixer-armv6.c b/firmware/target/arm/pcm-mixer-armv6.c
new file mode 100644
index 0000000000..94eecd0f90
--- /dev/null
+++ b/firmware/target/arm/pcm-mixer-armv6.c
@@ -0,0 +1,118 @@
+/***************************************************************************
+ *             __________               __   ___.
+ *   Open      \______   \ ____   ____ |  | _\_ |__   _______  ___
+ *   Source     |       _//  _ \_/ ___\|  |/ /| __ \ /  _ \  \/  /
+ *   Jukebox    |    |   (  <_> )  \___|    < | \_\ (  <_> > <  <
+ *   Firmware   |____|_  /\____/ \___  >__|_ \|___  /\____/__/\_ \
+ *                     \/            \/     \/    \/            \/
+ * $Id$
+ *
+ * Copyright (C) 2011 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.
+ *
+ ****************************************************************************/
+#define MIXER_OPTIMIZED_MIX_SAMPLES
+#define MIXER_OPTIMIZED_WRITE_SAMPLES
+
+/* Mix channels' samples and apply gain factors */
+static FORCE_INLINE void mix_samples(void *out,
+                                     void *src0,
+                                     int32_t src0_amp,
+                                     void *src1,
+                                     int32_t src1_amp,
+                                     size_t size)
+{
+    uint32_t s0, s1;
+
+    if (src0_amp == MIX_AMP_UNITY && src1_amp == MIX_AMP_UNITY)
+    {
+        /* Both are unity amplitude */
+        asm volatile (
+        "1:                      \n"
+            "ldr    %4, [%1], #4 \n"
+            "ldr    %5, [%2], #4 \n"
+            "subs   %3, %3, #4   \n"
+            "qadd16 %5, %5, %4   \n"
+            "str    %5, [%0], #4 \n"
+            "bhi    1b           \n"
+            : "+r"(out), "+r"(src0), "+r"(src1), "+r"(size),
+              "=&r"(s0), "=&r"(s1));
+    }
+    else
+    {
+        /* One or neither are unity amplitude */
+        uint32_t tmp;
+        asm volatile (
+        "1:                             \n"
+            "ldr    %4, [%1], #4        \n"
+            "ldr    %5, [%2], #4        \n"
+            "subs   %3, %3, #4          \n"
+            "smulwb %6, %7, %4          \n"
+            "smulwt %4, %7, %4          \n"
+            "smlawb %6, %8, %5, %6      \n"
+            "smlawt %4, %8, %5, %4      \n"
+            "ssat   %6, #16, %6         \n"
+            "ssat   %4, #16, %4         \n"
+            "pkhbt  %6, %6, %4, asl #16 \n"
+            "str    %6, [%0], #4        \n"
+            "bhi    1b                  \n"
+            : "+r"(out), "+r"(src0), "+r"(src1), "+r"(size),
+              "=&r"(s0), "=&r"(s1), "=&r"(tmp)
+            : "r"(src0_amp), "r"(src1_amp));
+    }
+}
+
+/* Write channel's samples and apply gain factor */
+static FORCE_INLINE void write_samples(void *out,
+                                       void *src,
+                                       int32_t amp,
+                                       size_t size)
+{
+    if (LIKELY(amp == MIX_AMP_UNITY))
+    {
+        /* Channel is unity amplitude */
+        asm volatile (
+            "ands       r1, %2, #0x1f  \n"
+            "beq        2f             \n"
+        "1:                            \n"
+            "ldr        r0, [%1], #4   \n"
+            "subs       r1, r1, #4     \n"
+            "str        r0, [%0], #4   \n"
+            "bne        1b             \n"
+            "bics       %2, %2, #0x1f  \n"
+            "beq        3f             \n"
+        "2:                            \n"
+            "ldmia      %1!, { r0-r7 } \n"
+            "subs       %2, %2, #32    \n"
+            "stmia      %0!, { r0-r7 } \n"
+            "bhi        2b             \n"
+        "3:                            \n"
+            : "+r"(out), "+r"(src), "+r"(size)
+            :
+            : "r0", "r1", "r2", "r3", "r4", "r5", "r6", "r7");
+    }
+    else
+    {
+        /* Channel needs amplitude cut */
+        uint32_t s, tmp;
+        asm volatile(
+        "1:                             \n"
+            "ldr    %3, [%1], #4        \n"
+            "subs   %2, %2, #4          \n"
+            "smulwt %4, %5, %3          \n"
+            "smulwb %3, %5, %3          \n"
+            "pkhbt  %4, %3, %4, asl #16 \n"
+            "str    %4, [%0], #4        \n"
+            "bhi    1b                  \n"
+            : "+r"(out), "+r"(src), "+r"(size),
+              "=&r"(s), "=&r"(tmp)
+            : "r"(amp));
+    }     
+}
diff --git a/firmware/target/arm/pcm-pp.c b/firmware/target/arm/pcm-pp.c
index c446f98fcf..704296d407 100644
--- a/firmware/target/arm/pcm-pp.c
+++ b/firmware/target/arm/pcm-pp.c
@@ -26,6 +26,7 @@
 #include "sound.h"
 #include "pcm.h"
 #include "pcm_sampr.h"
+#include "pcm-internal.h"
 
 /** DMA **/
 
@@ -115,6 +116,7 @@ void pcm_dma_apply_settings(void)
 /* NOTE: direct stack use forbidden by GCC stack handling bug for FIQ */
 void ICODE_ATTR __attribute__((interrupt("FIQ"))) fiq_playback(void)
 {
+    bool new_buffer = false;
     register size_t size;
 
     DMA0_STATUS; /* Clear any pending interrupt */
@@ -136,9 +138,14 @@ void ICODE_ATTR __attribute__((interrupt("FIQ"))) fiq_playback(void)
             /* Set the new DMA values and activate channel */
             DMA0_RAM_ADDR = dma_play_data.addr;
             DMA0_CMD = DMA_PLAY_CONFIG | (size - 4) | DMA_CMD_START;
+
+            if (new_buffer)
+                pcm_play_dma_started_callback();
             return;
         }
 
+        new_buffer = true;
+
         /* Buffer empty.  Try to get more. */
         pcm_play_get_more_callback((void **)&dma_play_data.addr,
                                    &dma_play_data.size);
@@ -181,8 +188,9 @@ void fiq_playback(void)
      * r0-r3 and r12 is a working register.
      */
     asm volatile (
-        "stmfd   sp!, { r0-r3, lr }  \n" /* stack scratch regs and lr */
+        "stmfd   sp!, { r0-r4, lr }  \n" /* stack scratch regs and lr */
 
+        "mov     r4, #0              \n" /* Was the callback called? */
 #if CONFIG_CPU == PP5002
         "ldr     r12, =0xcf001040    \n" /* Some magic from iPodLinux */
         "ldr     r12, [r12]          \n"
@@ -212,16 +220,13 @@ void fiq_playback(void)
         "tst     r1, #1              \n" /* two samples (one word) left? */
         "ldrne   r12, [r8], #4       \n" /* load two samples */
         "strne   r12, [r10, %[wr]]   \n" /* write sample 0-1 to IISFIFO_WR */
-
-        "cmp     r9, #0              \n" /* either FIFO is full or source buffer is empty */
-        "bgt     .exit               \n" /* if source buffer is not empty, FIFO must be full */
 #elif SAMPLE_SIZE == 32
     ".check_fifo:                    \n"
         "ldr     r0, [r10, %[cfg]]   \n" /* read IISFIFO_CFG to check FIFO status */
         "and     r0, r0, %[mask]     \n" /* r0 = IIS_TX_FREE_COUNT << 23 (PP5002) */
 
         "movs    r1, r0, lsr #24     \n" /* number of free pairs of FIFO slots */
-        "beq     .exit               \n" /* no complete pair? -> exit */
+        "beq     .fifo_fill_complete \n" /* no complete pair? -> exit */
         "cmp     r1, r9, lsr #2      \n" /* number of words from source */
         "movgt   r1, r9, lsr #2      \n" /* r1 = amount of allowed loops */
         "sub     r9, r9, r1, lsl #2  \n" /* r1 words will be written in following loop */
@@ -234,11 +239,23 @@ void fiq_playback(void)
         "subs    r1, r1, #1          \n" /* one more loop? */
         "bgt     .fifo_loop          \n" /* yes, continue */
 
+    ".fifo_fill_complete:            \n"
+#endif
+        "cmp     r4, #0              \n" /* If fill came after get_more... */
+        "beq     .still_old_buffer   \n"
+        "mov     r4, #0              \n"
+        "ldr     r2, =pcm_play_dma_started \n"
+        "ldrne   r2, [r2]            \n"
+        "cmp     r2, #0              \n"
+        "movne   lr, pc              \n"
+        "bxne    r2                  \n"
+
+    ".still_old_buffer:              \n"
         "cmp     r9, #0              \n" /* either FIFO is full or source buffer is empty */
         "bgt     .exit               \n" /* if source buffer is not empty, FIFO must be full */
-#endif
 
     ".more_data:                     \n"
+        "mov     r4, #1              \n" /* Remember we did this */
         "ldr     r2, =pcm_play_get_more_callback \n"
         "mov     r0, r11             \n" /* r0 = &p */
         "add     r1, r11, #4         \n" /* r1 = &size */
@@ -250,7 +267,7 @@ void fiq_playback(void)
 
     ".exit:                          \n" /* (r9=0 if stopping, look above) */
         "stmia   r11, { r8-r9 }      \n" /* save p and size */
-        "ldmfd   sp!, { r0-r3, lr }  \n"
+        "ldmfd   sp!, { r0-r4, lr }  \n"
         "subs    pc, lr, #4          \n" /* FIQ specific return sequence */
         ".ltorg                      \n"
         : /* These must only be integers! No regs */
@@ -264,6 +281,8 @@ void fiq_playback(void) __attribute__((interrupt ("FIQ"))) ICODE_ATTR;
 /* NOTE: direct stack use forbidden by GCC stack handling bug for FIQ */
 void fiq_playback(void)
 {
+    bool new_buffer = false;
+
 #if CONFIG_CPU == PP5002
     inl(0xcf001040);
 #endif
@@ -271,6 +290,10 @@ void fiq_playback(void)
     do {
         while (dma_play_data.size > 0) {
             if (IIS_TX_FREE_COUNT < 2) {
+                if (new_buffer) {
+                    new_buffer = false;
+                    pcm_play_dma_started_callback();
+                }
                 return;
             }
 #if SAMPLE_SIZE == 16
@@ -282,9 +305,15 @@ void fiq_playback(void)
             dma_play_data.size -= 4;
         }
 
+        if (new_buffer) {
+            new_buffer = false;
+            pcm_play_dma_started_callback();
+        }
+
         /* p is empty, get some more data */
         pcm_play_get_more_callback((void **)&dma_play_data.addr,
                                    &dma_play_data.size);
+        new_buffer = true;
     } while (dma_play_data.size);
 
     /* No more data  */
diff --git a/firmware/target/arm/pcm-telechips.c b/firmware/target/arm/pcm-telechips.c
index 851ebee7de..aff43171f6 100644
--- a/firmware/target/arm/pcm-telechips.c
+++ b/firmware/target/arm/pcm-telechips.c
@@ -27,6 +27,7 @@
 #include "sound.h"
 #include "i2s.h"
 #include "pcm.h"
+#include "pcm-internal.h"
 
 struct dma_data
 {
@@ -247,6 +248,8 @@ void fiq_handler(void)
      * r0-r3 and r12 is a working register.
      */
     asm volatile (
+        "stmfd   sp!, { r0-r4, lr }  \n" /* stack scratch regs and lr */
+        "mov     r4, #0              \n" /* Was the callback called? */
 #if defined(CPU_TCC780X)
         "mov     r8, #0xc000         \n" /* DAI_TX_IRQ_MASK | DAI_RX_IRQ_MASK */
         "ldr     r9, =0xf3001004     \n" /* CREQ */
@@ -279,33 +282,41 @@ void fiq_handler(void)
         "sub     r9, r9, #0x10       \n" /* 4 words written */
         "stmia   r11, { r8-r9 }      \n" /* save p and size */
 
+        "cmp     r4, #0              \n" /* Callback called? */
+        "beq     .exit               \n"
+        /* "mov     r4, #0              \n" If get_more could be called multiple times! */
+        "ldr     r2, =pcm_play_dma_started\n"
+        "ldr     r2, [r2]            \n"
+        "cmp     r2, #0              \n"
+        "blxne   r2                  \n"
+
     ".exit:                          \n"
+        "ldmfd   sp!, { r0-r4, lr }  \n"
         "subs    pc, lr, #4          \n" /* FIQ specific return sequence */
 
     ".more_data:                     \n"
-        "stmfd   sp!, { r0-r3, lr }  \n" /* stack scratch regs and lr */
+        "mov     r4, #1              \n" /* Remember we got more data in this FIQ */
         "ldr     r2, =pcm_play_get_more_callback \n"
         "mov     r0, r11             \n" /* r0 = &p */
         "add     r1, r11, #4         \n" /* r1 = &size */
         "blx     r2                  \n" /* call pcm_play_get_more_callback */
         "ldmia   r11, { r8-r9 }      \n" /* load new p and size */
         "cmp     r9, #0x10           \n" /* did we actually get enough data? */
-        "ldmfd   sp!, { r0-r3, lr }  \n"
         "bpl     .fill_fifo          \n" /* not stop and enough? refill */
         "b       .exit               \n"
         ".ltorg                      \n"
     );
 }
 #else /* C version for reference */
-void fiq_handler(void) ICODE_ATTR __attribute__((naked));
+void fiq_handler(void) ICODE_ATTR;
 void fiq_handler(void)
 {
-    asm volatile(   "stmfd sp!, {r0-r7, ip, lr} \n"   /* Store context */
-                    "sub   sp, sp, #8           \n"); /* Reserve stack */
+    register bool new_buffer = false;
 
     if (dma_play_data.size < 16)
     {
         /* p is empty, get some more data */
+        new_buffer = true;
         pcm_play_get_more_callback((void**)&dma_play_data.p,
                                    &dma_play_data.size);
     }
@@ -327,9 +338,8 @@ void fiq_handler(void)
     /* Clear FIQ status */
     CREQ = DAI_TX_IRQ_MASK | DAI_RX_IRQ_MASK;
 
-    asm volatile(   "add   sp, sp, #8           \n"   /* Cleanup stack   */
-                    "ldmfd sp!, {r0-r7, ip, lr} \n"   /* Restore context */
-                    "subs  pc, lr, #4           \n"); /* Return from FIQ */
+    if (new_buffer)
+        pcm_play_dma_started_callback();
 }
 #endif
 
diff --git a/firmware/target/arm/pnx0101/pcm-pnx0101.c b/firmware/target/arm/pnx0101/pcm-pnx0101.c
index 9d4ffbd773..d4c17454ed 100644
--- a/firmware/target/arm/pnx0101/pcm-pnx0101.c
+++ b/firmware/target/arm/pnx0101/pcm-pnx0101.c
@@ -21,6 +21,7 @@
 #include "system.h"
 #include "audio.h"
 #include "string.h"
+#include "pcm-internal.h"
 
 #define DMA_BUF_SAMPLES 0x100
 
@@ -63,6 +64,8 @@ static inline void fill_dma_buf(int offset)
 
     if (pcm_playing && !pcm_paused)
     {
+        bool new_buffer =false;
+
         do
         {
             int count;
@@ -102,10 +105,20 @@ static inline void fill_dma_buf(int offset)
                 count--;
             }
             p = tmp_p;
+
+            if (new_buffer)
+            {
+                new_buffer = false;
+                pcm_play_dma_started_callback();
+            }
+
             if (l >= lend)
                 return;
 
             pcm_play_get_more_callback((void**)&p, &p_size);
+
+            if (p_size)
+                new_buffer = true;
         }
         while (p_size);
     }
diff --git a/firmware/target/arm/s3c2440/gigabeat-fx/pcm-meg-fx.c b/firmware/target/arm/s3c2440/gigabeat-fx/pcm-meg-fx.c
index c1c9017fbb..33194ae5d9 100644
--- a/firmware/target/arm/s3c2440/gigabeat-fx/pcm-meg-fx.c
+++ b/firmware/target/arm/s3c2440/gigabeat-fx/pcm-meg-fx.c
@@ -25,6 +25,7 @@
 #include "audio.h"
 #include "sound.h"
 #include "file.h"
+#include "pcm-internal.h"
 
 /* PCM interrupt routine lockout */
 static struct
@@ -235,6 +236,8 @@ void fiq_handler(void)
 
     /* Re-Activate the channel */
     DMASKTRIG2 = 0x2;
+
+    pcm_play_dma_started_callback();
 }
 
 size_t pcm_get_bytes_waiting(void)
diff --git a/firmware/target/arm/s3c2440/gigabeat-fx/wmcodec-meg-fx.c b/firmware/target/arm/s3c2440/gigabeat-fx/wmcodec-meg-fx.c
index 01b177da6c..eea4c58e4b 100644
--- a/firmware/target/arm/s3c2440/gigabeat-fx/wmcodec-meg-fx.c
+++ b/firmware/target/arm/s3c2440/gigabeat-fx/wmcodec-meg-fx.c
@@ -99,14 +99,15 @@ void TIMER3(void)
     INTPND = TIMER3_MASK;
 }
 
-void pcmbuf_beep(unsigned int frequency, size_t duration, int amplitude)
+void beep_play(unsigned int frequency, unsigned int duration,
+               unsigned int amplitude)
 {
     #define TIMER3_TICK_SEC (TIMER_FREQ / TIMER234_PRESCALE)
 
     unsigned long tcnt, tcmp;
     int oldstatus;
 
-    if (amplitude <= 0)
+    if (frequency == 0 || duration == 0 || amplitude == 0)
     {
         beep_stop(); /* won't hear it anyway */
         return;
diff --git a/firmware/target/arm/s3c2440/mini2440/pcm-mini2440.c b/firmware/target/arm/s3c2440/mini2440/pcm-mini2440.c
index 8a6b62f31f..0c69c1e6d3 100644
--- a/firmware/target/arm/s3c2440/mini2440/pcm-mini2440.c
+++ b/firmware/target/arm/s3c2440/mini2440/pcm-mini2440.c
@@ -26,6 +26,7 @@
 #include "audio.h"
 #include "sound.h"
 #include "file.h"
+#include "pcm-internal.h"
 
 /* PCM interrupt routine lockout */
 static struct
@@ -275,6 +276,8 @@ void fiq_handler(void)
 
     /* Re-Activate the channel */
     DMASKTRIG2 = 0x2;
+
+    pcm_play_dma_started_callback();
 }
 
 size_t pcm_get_bytes_waiting(void)
diff --git a/firmware/target/arm/s5l8700/pcm-s5l8700.c b/firmware/target/arm/s5l8700/pcm-s5l8700.c
index 08086c37d8..14c515ec47 100644
--- a/firmware/target/arm/s5l8700/pcm-s5l8700.c
+++ b/firmware/target/arm/s5l8700/pcm-s5l8700.c
@@ -27,6 +27,7 @@
 #include "panic.h"
 #include "audiohw.h"
 #include "pcm.h"
+#include "pcm-internal.h"
 #include "pcm_sampr.h"
 #include "dma-target.h"
 #include "mmu-arm.h"
@@ -100,6 +101,7 @@ void pcm_play_unlock(void)
 void INT_DMA(void) ICODE_ATTR;
 void INT_DMA(void)
 {
+    bool new_buffer = false;
     DMACOM0 = 7;
     while (!(DMACON0 & (1 << 18)))
     {
@@ -112,8 +114,12 @@ void INT_DMA(void)
         }
         else
         {
-            if (!nextsize) pcm_play_get_more_callback((void**)&nextbuf, &nextsize);
-            if (!nextsize) break;
+            if (!nextsize)
+            {
+                pcm_play_get_more_callback((void**)&nextbuf, &nextsize);
+                if (!nextsize) break;
+                new_buffer = true;
+            }
             queuedsize = MIN(sizeof(dblbuf), nextsize / 2);
             nextsize -= queuedsize;
             queuedbuf = nextbuf + nextsize;
@@ -124,7 +130,14 @@ void INT_DMA(void)
         clean_dcache();
         DMACOM0 = 4;
         DMACOM0 = 7;
+
+        if (new_buffer)
+        {
+            pcm_play_dma_started_callback();
+            new_buffer = false;
+        }
     }
+
 }
 
 void pcm_play_dma_start(const void* addr, size_t size)
diff --git a/firmware/target/arm/s5l8702/pcm-s5l8702.c b/firmware/target/arm/s5l8702/pcm-s5l8702.c
index c0498a9ce2..dbadf3bac0 100644
--- a/firmware/target/arm/s5l8702/pcm-s5l8702.c
+++ b/firmware/target/arm/s5l8702/pcm-s5l8702.c
@@ -27,6 +27,7 @@
 #include "panic.h"

 #include "audiohw.h"

 #include "pcm.h"

+#include "pcm-internal.h"

 #include "pcm_sampr.h"

 #include "mmu-arm.h"

 #include "pcm-target.h"

@@ -113,6 +114,8 @@ void INT_DMAC0C0(void)
         DMAC0C0CONFIG = 0x8a81;

     }

     else DMAC0C0NEXTLLI = pcm_lli;

+

+    pcm_play_dma_started_callback();

 }

 

 void pcm_play_dma_start(const void* addr, size_t size)

diff --git a/firmware/target/arm/tms320dm320/creative-zvm/pcm-creativezvm.c b/firmware/target/arm/tms320dm320/creative-zvm/pcm-creativezvm.c
index 3c54ce81fb..5ec62cf876 100644
--- a/firmware/target/arm/tms320dm320/creative-zvm/pcm-creativezvm.c
+++ b/firmware/target/arm/tms320dm320/creative-zvm/pcm-creativezvm.c
@@ -27,6 +27,7 @@
 #include "dm320.h"
 #include "audiohw.h"
 #include "dsp-target.h"
+#include "pcm-internal.h"
 
 void pcm_play_dma_init(void)
 {
diff --git a/firmware/target/arm/tms320dm320/mrobe-500/pcm-mr500.c b/firmware/target/arm/tms320dm320/mrobe-500/pcm-mr500.c
index fb94adae71..90c342e868 100644
--- a/firmware/target/arm/tms320dm320/mrobe-500/pcm-mr500.c
+++ b/firmware/target/arm/tms320dm320/mrobe-500/pcm-mr500.c
@@ -28,6 +28,7 @@
 #include "dsp-target.h"
 #include "dsp/ipc.h"
 #include "mmu-arm.h"
+#include "pcm-internal.h"
 
 /* This is global to save some latency when pcm_play_dma_get_peak_buffer is 
  *  called.
@@ -178,6 +179,8 @@ void DSPHINT(void)
             
             DEBUGF("pcm_sdram at 0x%08lx, sdem_addr 0x%08lx",
                 (unsigned long)start, (unsigned long)sdem_addr);
+
+            pcm_play_dma_started_callback();
         }
         
         break;
diff --git a/firmware/target/coldfire/pcm-coldfire.c b/firmware/target/coldfire/pcm-coldfire.c
index a06542c31f..85eeaec815 100644
--- a/firmware/target/coldfire/pcm-coldfire.c
+++ b/firmware/target/coldfire/pcm-coldfire.c
@@ -28,6 +28,7 @@
 #if defined(HAVE_SPDIF_REC) || defined(HAVE_SPDIF_OUT)
 #include "spdif.h"
 #endif
+#include "pcm-internal.h"
 
 #define IIS_PLAY_DEFPARM ( (freq_ent[FPARM_CLOCKSEL] << 12) | \
                            (IIS_PLAY & (7 << 8)) | \
@@ -318,6 +319,9 @@ void DMA0(void)
         SAR0 = (unsigned long)start;     /* Source address */
         BCR0 = size;                     /* Bytes to transfer */
         or_l(DMA_EEXT | DMA_INT, &DCR0); /* per request and int ON */
+
+        /* Call buffer callback */
+        pcm_play_dma_started_callback();
     }
     /* else inished playing */
 } /* DMA0 */
diff --git a/firmware/target/coldfire/pcm-mixer-coldfire.c b/firmware/target/coldfire/pcm-mixer-coldfire.c
new file mode 100644
index 0000000000..d8318fffaf
--- /dev/null
+++ b/firmware/target/coldfire/pcm-mixer-coldfire.c
@@ -0,0 +1,134 @@
+/***************************************************************************
+ *             __________               __   ___.
+ *   Open      \______   \ ____   ____ |  | _\_ |__   _______  ___
+ *   Source     |       _//  _ \_/ ___\|  |/ /| __ \ /  _ \  \/  /
+ *   Jukebox    |    |   (  <_> )  \___|    < | \_\ (  <_> > <  <
+ *   Firmware   |____|_  /\____/ \___  >__|_ \|___  /\____/__/\_ \
+ *                     \/            \/     \/    \/            \/
+ * $Id$
+ *
+ * Copyright (C) 2011 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.
+ *
+ ****************************************************************************/
+
+#define MIXER_OPTIMIZED_MIX_SAMPLES
+#define MIXER_OPTIMIZED_WRITE_SAMPLES
+static struct emac_context
+{
+    unsigned long r[4];
+} emac_context IBSS_ATTR;
+
+/* Save emac context affected in ISR */
+static FORCE_INLINE void save_emac_context(void)
+{
+    asm volatile (
+        "move.l   %%macsr, %%d0             \n"
+        "move.l   %%accext01, %%d1          \n"
+        "movclr.l %%acc0, %%a0              \n"
+        "movclr.l %%acc1, %%a1              \n"
+        "movem.l  %%d0-%%d1/%%a0-%%a1, (%0) \n"
+        :
+        : "a"(&emac_context)
+        : "d0", "d1", "a0", "a1");
+}
+
+/* Restore emac context affected in ISR */
+static FORCE_INLINE void restore_emac_context(void)
+{
+    asm volatile (
+        "movem.l (%0), %%d0-%%d1/%%a0-%%a1  \n"
+        "move.l  %%a1, %%acc1               \n"
+        "move.l  %%a0, %%acc0               \n"
+        "move.l  %%d1, %%accext01           \n"
+        "move.l  %%d0, %%macsr              \n"
+        :
+        : "a"(&emac_context)
+        : "d0", "d1", "a0", "a1");
+}
+
+/* Mix channels' samples and apply gain factors */
+static FORCE_INLINE void mix_samples(void *out,
+                                     void *src0,
+                                     int32_t src0_amp,
+                                     void *src1,
+                                     int32_t src1_amp,
+                                     size_t size)
+{
+    uint32_t s0, s1, s2, s3;
+    save_emac_context();
+    coldfire_set_macsr(EMAC_ROUND | EMAC_SATURATE);
+
+    asm volatile (
+        "move.l     (%1)+, %5                 \n"
+    "1:                                       \n"
+        "movea.w    %5, %4                    \n"
+        "asr.l      %10, %5                   \n"
+        "mac.l      %4, %8,            %%acc0 \n"
+        "mac.l      %5, %8, (%2)+, %5, %%acc1 \n"
+        "movea.w    %5, %4                    \n"
+        "asr.l      %10, %5                   \n"
+        "mac.l      %4, %9,            %%acc0 \n"
+        "mac.l      %5, %9, (%1)+, %5, %%acc1 \n"
+        "movclr.l   %%acc0, %6                \n"
+        "movclr.l   %%acc1, %7                \n"
+        "swap.w     %6                        \n"
+        "move.w     %6, %7                    \n"
+        "move.l     %7, (%0)+                 \n"
+        "subq.l     #4, %3                    \n"
+        "bhi.b      1b                        \n"
+        : "+a"(out), "+a"(src0), "+a"(src1), "+d"(size),
+          "=&a"(s0), "=&d"(s1), "=&d"(s2), "=&d"(s3)
+        : "r"(src0_amp), "r"(src1_amp), "d"(16)
+    );
+
+    restore_emac_context();
+}
+
+/* Write channel's samples and apply gain factor */
+static FORCE_INLINE void write_samples(void *out,
+                                       void *src,
+                                       int32_t amp,
+                                       size_t size)
+{
+    if (LIKELY(amp == MIX_AMP_UNITY))
+    {
+        /* Channel is unity amplitude */
+        memcpy(out, src, size);
+    }
+    else
+    {
+        /* Channel needs amplitude cut */
+        uint32_t s0, s1, s2, s3;
+        save_emac_context();
+        coldfire_set_macsr(EMAC_ROUND | EMAC_SATURATE);
+
+        asm volatile (
+            "move.l     (%1)+, %4                 \n"
+        "1:                                       \n"
+            "movea.w    %4, %3                    \n"
+            "asr.l      %8, %4                    \n"
+            "mac.l      %3, %7,            %%acc0 \n"
+            "mac.l      %4, %7, (%1)+, %4, %%acc1 \n"
+            "movclr.l   %%acc0, %5                \n"
+            "movclr.l   %%acc1, %6                \n"
+            "swap.w     %5                        \n"
+            "move.w     %5, %6                    \n"
+            "move.l     %6, (%0)+                 \n"
+            "subq.l     #4, %2                    \n"
+            "bhi.b      1b                        \n"
+            : "+a"(out), "+a"(src), "+d"(size),
+              "=&a"(s0), "=&d"(s1), "=&d"(s2), "=&d"(s3)
+            : "r"(amp), "d"(16)
+        );
+
+        restore_emac_context();
+    }     
+}
diff --git a/firmware/target/hosted/android/pcm-android.c b/firmware/target/hosted/android/pcm-android.c
index 88792cd76f..cbd6cb3228 100644
--- a/firmware/target/hosted/android/pcm-android.c
+++ b/firmware/target/hosted/android/pcm-android.c
@@ -23,14 +23,18 @@
 #include <stdbool.h>
 #define _SYSTEM_WITH_JNI /* for getJavaEnvironment */
 #include <system.h>
+#include <pthread.h>
 #include "debug.h"
 #include "pcm.h"
+#include "pcm-internal.h"
 
 extern JNIEnv *env_ptr;
 
 /* infos about our pcm chunks */
 static size_t  pcm_data_size;
 static char   *pcm_data_start;
+static int     audio_locked = 0;
+static pthread_mutex_t audio_lock_mutex = PTHREAD_MUTEX_INITIALIZER;
 
 /* cache frequently called methods */
 static jmethodID play_pause_method;
@@ -42,6 +46,20 @@ static jobject   RockboxPCM_instance;
 
 
 /*
+ * mutex lock/unlock wrappers neatness' sake
+ */
+static inline void lock_audio(void)
+{
+    pthread_mutex_lock(&audio_lock_mutex);
+}
+
+static inline void unlock_audio(void)
+{
+    pthread_mutex_unlock(&audio_lock_mutex);
+}
+
+
+/*
  * write pcm samples to the hardware. Calls AudioTrack.write directly (which
  * is usually a blocking call)
  *
@@ -54,10 +72,17 @@ JNIEXPORT jint JNICALL
 Java_org_rockbox_RockboxPCM_nativeWrite(JNIEnv *env, jobject this,
                                         jbyteArray temp_array, jint max_size)
 {
+    bool new_buffer = false;
+
+    lock_audio();
+
     jint left = max_size;
 
     if (!pcm_data_size) /* get some initial data */
+    {
+        new_buffer = true;
         pcm_play_get_more_callback((void**) &pcm_data_start, &pcm_data_size);
+    }
 
     while(left > 0 && pcm_data_size)
     {
@@ -70,23 +95,49 @@ Java_org_rockbox_RockboxPCM_nativeWrite(JNIEnv *env, jobject this,
 
         ret = (*env)->CallIntMethod(env, this, write_method,
                                             temp_array, 0, transfer_size);
+
+        if (new_buffer)
+        {
+            new_buffer = false;
+            pcm_play_dma_started_callback();
+
+            /* NOTE: might need to release the mutex and sleep here if the
+               buffer is shorter than the required buffer (like pcm-sdl.c) to
+               have the mixer clocked at a regular interval */
+        }
+
         if (ret < 0)
+        {
+            unlock_audio();
             return ret;
+        }
 
         if (pcm_data_size == 0) /* need new data */
+        {
+            new_buffer = true;
             pcm_play_get_more_callback((void**)&pcm_data_start, &pcm_data_size);
+        }
         else /* increment data pointer and feed more */
             pcm_data_start += transfer_size;
     }
+
+    if (new_buffer && pcm_data_size)
+        pcm_play_dma_started_callback();
+
+    unlock_audio();
     return max_size - left;
 }
 
 void pcm_play_lock(void)
 {
+    if (++audio_locked == 1)
+        lock_audio();
 }
 
 void pcm_play_unlock(void)
 {
+    if (--audio_locked == 0)
+        unlock_audio();
 }
 
 void pcm_dma_apply_settings(void)
@@ -153,8 +204,6 @@ void pcm_play_dma_init(void)
     set_volume_method = e->GetMethodID(env_ptr, RockboxPCM_class, "set_volume", "(I)V");
     stop_method       = e->GetMethodID(env_ptr, RockboxPCM_class, "stop", "()V");
     write_method      = e->GetMethodID(env_ptr, RockboxPCM_class, "write", "([BII)I");
-    /* get initial pcm data, if any */
-    pcm_play_get_more_callback((void*)&pcm_data_start, &pcm_data_size);
 }
 
 void pcm_postinit(void)
@@ -173,6 +222,7 @@ void pcm_shutdown(void)
     JNIEnv e = *env_ptr;
     jmethodID release = e->GetMethodID(env_ptr, RockboxPCM_class, "release", "()V");
     e->CallVoidMethod(env_ptr, RockboxPCM_instance, release);
+    pthread_mutex_destroy(&audio_lock_mutex);
 }
     
 /* Due to limitations of default_event_handler(), parameters gets swallowed when
diff --git a/firmware/target/hosted/maemo/pcm-gstreamer.c b/firmware/target/hosted/maemo/pcm-gstreamer.c
index e3e40f0619..6069801fba 100644
--- a/firmware/target/hosted/maemo/pcm-gstreamer.c
+++ b/firmware/target/hosted/maemo/pcm-gstreamer.c
@@ -54,6 +54,7 @@
 #endif
 
 #include "pcm.h"
+#include "pcm-internal.h"
 #include "pcm_sampr.h"
 
 /*#define LOGF_ENABLE*/
@@ -182,6 +183,8 @@ static void feed_data(GstElement * appsrc, guint size_hint, void *unused)
 
         if (ret != 0)
             DEBUGF("push-buffer error result: %d\n", ret);
+
+        pcm_play_dma_started_callback();
     } else
     {
         DEBUGF("feed_data: No Data.\n");
diff --git a/firmware/target/hosted/sdl/pcm-sdl.c b/firmware/target/hosted/sdl/pcm-sdl.c
index 7780083b99..dfdd90f29b 100644
--- a/firmware/target/hosted/sdl/pcm-sdl.c
+++ b/firmware/target/hosted/sdl/pcm-sdl.c
@@ -30,6 +30,7 @@
 #include "sound.h"
 #include "audiohw.h"
 #include "system.h"
+#include "panic.h"
 
 #ifdef HAVE_RECORDING
 #include "audiohw.h"
@@ -39,6 +40,7 @@
 #endif
 
 #include "pcm.h"
+#include "pcm-internal.h"
 #include "pcm_sampr.h"
 
 /*#define LOGF_ENABLE*/
@@ -71,15 +73,19 @@ static struct pcm_udata
 
 static SDL_AudioSpec obtained;
 static SDL_AudioCVT cvt;
+static int audio_locked = 0;
+static SDL_mutex *audio_lock;
 
 void pcm_play_lock(void)
 {
-    SDL_LockAudio();
+    if (++audio_locked == 1)
+        SDL_LockMutex(audio_lock);
 }
 
 void pcm_play_unlock(void)
 {
-    SDL_UnlockAudio();
+    if (--audio_locked == 0)
+        SDL_UnlockMutex(audio_lock);
 }
 
 static void pcm_dma_apply_settings_nolock(void)
@@ -227,14 +233,19 @@ static void write_to_soundcard(struct pcm_udata *udata)
 static void sdl_audio_callback(struct pcm_udata *udata, Uint8 *stream, int len)
 {
     logf("sdl_audio_callback: len %d, pcm %d\n", len, pcm_data_size);
+
+    bool new_buffer = false;
     udata->stream = stream;
 
+    SDL_LockMutex(audio_lock);
+
     /* Write what we have in the PCM buffer */
     if (pcm_data_size > 0)
         goto start;
 
     /* Audio card wants more? Get some more then. */
     while (len > 0) {
+        new_buffer = true;
         pcm_play_get_more_callback((void **)&pcm_data, &pcm_data_size);
     start:
         if (pcm_data_size != 0) {
@@ -246,6 +257,28 @@ static void sdl_audio_callback(struct pcm_udata *udata, Uint8 *stream, int len)
             udata->num_in  *= pcm_sample_bytes;
             udata->num_out *= pcm_sample_bytes;
 
+
+            if (new_buffer)
+            {
+                new_buffer = false;
+                pcm_play_dma_started_callback();
+
+                if ((size_t)len > udata->num_out)
+                {
+                    int delay = pcm_data_size*250 / pcm_sampr - 1;
+                
+                    if (delay > 0)
+                    {
+                        SDL_UnlockMutex(audio_lock);
+                        SDL_Delay(delay);
+                        SDL_LockMutex(audio_lock);
+
+                        if (!pcm_is_playing())
+                            break;
+                    }
+                }
+            }
+
             pcm_data      += udata->num_in;
             pcm_data_size -= udata->num_in;
             udata->stream += udata->num_out;
@@ -255,6 +288,8 @@ static void sdl_audio_callback(struct pcm_udata *udata, Uint8 *stream, int len)
             break;
         }
     }
+
+    SDL_UnlockMutex(audio_lock);
 }
 
 const void * pcm_play_dma_get_peak_buffer(int *count)
@@ -320,6 +355,14 @@ void pcm_play_dma_init(void)
         return;
     }
 
+    audio_lock = SDL_CreateMutex();
+
+    if (!audio_lock)
+    {
+        panicf("Could not create audio_lock\n");
+        return;
+    }
+
     SDL_AudioSpec wanted_spec;
 #ifdef DEBUG
     udata.debug = NULL;
diff --git a/firmware/target/mips/ingenic_jz47xx/pcm-jz4740.c b/firmware/target/mips/ingenic_jz47xx/pcm-jz4740.c
index 5cd9c33e18..cfc3c9ef8e 100644
--- a/firmware/target/mips/ingenic_jz47xx/pcm-jz4740.c
+++ b/firmware/target/mips/ingenic_jz47xx/pcm-jz4740.c
@@ -25,6 +25,7 @@
 #include "audio.h"
 #include "sound.h"
 #include "pcm.h"
+#include "pcm-internal.h"
 #include "jz4740.h"
 
 
@@ -109,6 +110,7 @@ static inline void play_dma_callback(void)
     {
         set_dma(start, size);
         REG_DMAC_DCCSR(DMA_AIC_TX_CHANNEL) |= DMAC_DCCSR_EN;
+        pcm_play_dma_started_callback();
     }
 }