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authorMichael Sevakis <jethead71@rockbox.org>2013-05-18 01:45:03 -0400
committerMichael Sevakis <jethead71@rockbox.org>2013-05-21 00:02:14 -0400
commit87021f7c0ac4620eafd185ff11905ee643f72b6c (patch)
tree03ae48f3d999cd8743af40cc5df933f64f6df2d2 /lib/rbcodec/codecs
parenta17d6de5bc727b0bb55764ecef2605ae689e8dab (diff)
downloadrockbox-87021f7c0ac4620eafd185ff11905ee643f72b6c.tar.gz
rockbox-87021f7c0ac4620eafd185ff11905ee643f72b6c.zip
SPC Codec: Refactor for CPU and clean up some things.
CPU optimization gets its own files in which to fill-in optimizable routines. Some pointless #if 0's for profiling need removal. Those macros are empty if not profiling. Force some functions that are undesirable to be force-inlined by the compiler to be not inlined. Change-Id: Ia7b7e45380d7efb20c9b1a4d52e05db3ef6bbaab
Diffstat (limited to 'lib/rbcodec/codecs')
-rw-r--r--lib/rbcodec/codecs/libspc/cpu/spc_dsp_armv4.c253
-rw-r--r--lib/rbcodec/codecs/libspc/cpu/spc_dsp_armv4.h45
-rw-r--r--lib/rbcodec/codecs/libspc/cpu/spc_dsp_armv6.c244
-rw-r--r--lib/rbcodec/codecs/libspc/cpu/spc_dsp_armv6.h45
-rw-r--r--lib/rbcodec/codecs/libspc/cpu/spc_dsp_coldfire.c198
-rw-r--r--lib/rbcodec/codecs/libspc/cpu/spc_dsp_coldfire.h45
-rw-r--r--lib/rbcodec/codecs/libspc/spc_codec.h147
-rw-r--r--lib/rbcodec/codecs/libspc/spc_cpu.c4
-rw-r--r--lib/rbcodec/codecs/libspc/spc_dsp.c1733
-rw-r--r--lib/rbcodec/codecs/libspc/spc_dsp_generic.c211
-rw-r--r--lib/rbcodec/codecs/libspc/spc_dsp_generic.h45
-rw-r--r--lib/rbcodec/codecs/libspc/spc_emu.c15
12 files changed, 1690 insertions, 1295 deletions
diff --git a/lib/rbcodec/codecs/libspc/cpu/spc_dsp_armv4.c b/lib/rbcodec/codecs/libspc/cpu/spc_dsp_armv4.c
new file mode 100644
index 0000000000..7eacc3baf9
--- /dev/null
+++ b/lib/rbcodec/codecs/libspc/cpu/spc_dsp_armv4.c
@@ -0,0 +1,253 @@
+/***************************************************************************
+ * __________ __ ___.
+ * Open \______ \ ____ ____ | | _\_ |__ _______ ___
+ * Source | _// _ \_/ ___\| |/ /| __ \ / _ \ \/ /
+ * Jukebox | | ( <_> ) \___| < | \_\ ( <_> > < <
+ * Firmware |____|_ /\____/ \___ >__|_ \|___ /\____/__/\_ \
+ * \/ \/ \/ \/ \/
+ * $Id$
+ *
+ * Copyright (C) 2007-2010 Michael Sevakis (jhMikeS)
+ *
+ * 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.
+ *
+ ****************************************************************************/
+#if !SPC_NOINTERP
+
+#define SPC_GAUSSIAN_FAST_INTERP
+static inline int gaussian_fast_interp( int16_t const* samples,
+ int32_t position,
+ int16_t const* fwd,
+ int16_t const* rev )
+{
+ int output;
+ int t0, t1, t2, t3;
+
+ asm volatile (
+ "ldrsh %[t0], [%[samp]] \n"
+ "ldrsh %[t2], [%[fwd]] \n"
+ "ldrsh %[t1], [%[samp], #2] \n"
+ "ldrsh %[t3], [%[fwd], #2] \n"
+ "mul %[out], %[t0], %[t2] \n" /* out= fwd[0]*samp[0] */
+ "ldrsh %[t0], [%[samp], #4] \n"
+ "ldrsh %[t2], [%[rev], #2] \n"
+ "mla %[out], %[t1], %[t3], %[out] \n" /* out+=fwd[1]*samp[1] */
+ "ldrsh %[t1], [%[samp], #6] \n"
+ "ldrsh %[t3], [%[rev]] \n"
+ "mla %[out], %[t0], %[t2], %[out] \n" /* out+=rev[1]*samp[2] */
+ "mla %[out], %[t1], %[t3], %[out] \n" /* out+=rev[0]*samp[3] */
+ : [out]"=&r"(output),
+ [t0]"=&r"(t0), [t1]"=&r"(t1), [t2]"=&r"(t2), [t3]"=&r"(t3)
+ : [fwd]"r"(fwd), [rev]"r"(rev),
+ [samp]"r"(samples + (position >> 12)));
+
+ return output;
+}
+
+#define SPC_GAUSSIAN_FAST_AMP
+static inline int gaussian_fast_amp( struct voice_t* voice, int output,
+ int* amp_0, int* amp_1 )
+{
+ int t0;
+
+ asm volatile (
+ "mov %[t0], %[out], asr #11 \n"
+ "mul %[out], %[t0], %[envx] \n"
+ : [out]"+r"(output), [t0]"=&r"(t0)
+ : [envx]"r"((int) voice->envx));
+
+ asm volatile (
+ "mov %[out], %[out], asr #11 \n"
+ "mul %[a0], %[out], %[v0] \n"
+ "mul %[a1], %[out], %[v1] \n"
+ : [out]"+r"(output),
+ [a0]"=&r"(*amp_0), [a1]"=r"(*amp_1)
+ : [v0]"r"((int) voice->volume [0]),
+ [v1]"r"((int) voice->volume [1]));
+
+ return output;
+}
+
+#define SPC_GAUSSIAN_SLOW_INTERP
+static inline int gaussian_slow_interp( int16_t const* samples,
+ int32_t position,
+ int16_t const* fwd,
+ int16_t const* rev )
+{
+ int output;
+ int t0, t1, t2, t3;
+
+ asm volatile (
+ "ldrsh %[t0], [%[samp]] \n"
+ "ldrsh %[t2], [%[fwd]] \n"
+ "ldrsh %[t1], [%[samp], #2] \n"
+ "ldrsh %[t3], [%[fwd], #2] \n"
+ "mul %[out], %[t2], %[t0] \n" /* fwd[0]*samp[0] */
+ "ldrsh %[t2], [%[rev], #2] \n"
+ "mul %[t0], %[t3], %[t1] \n" /* fwd[1]*samp[1] */
+ "ldrsh %[t1], [%[samp], #4] \n"
+ "mov %[out], %[out], asr #12 \n"
+ "ldrsh %[t3], [%[rev]] \n"
+ "mul %[t2], %[t1], %[t2] \n" /* rev[1]*samp[2] */
+ "ldrsh %[t1], [%[samp], #6] \n"
+ "add %[t0], %[out], %[t0], asr #12 \n"
+ "mul %[t3], %[t1], %[t3] \n" /* rev[0]*samp[3] */
+ "add %[t2], %[t0], %[t2], asr #12 \n"
+ "mov %[t2], %[t2], lsl #17 \n"
+ "mov %[t3], %[t3], asr #12 \n"
+ "mov %[t3], %[t3], asl #1 \n"
+ "add %[out], %[t3], %[t2], asr #16 \n"
+ : [out]"=&r"(output),
+ [t0]"=&r"(t0), [t1]"=&r"(t1), [t2]"=&r"(t2), [t3]"=&r"(t3)
+ : [fwd]"r"(fwd), [rev]"r"(rev),
+ [samp]"r"(samples + (position >> 12)));
+
+ return CLAMP16( output );
+}
+
+#define SPC_GAUSSIAN_SLOW_AMP
+static inline int gaussian_slow_amp( struct voice_t* voice, int output,
+ int* amp_0, int* amp_1 )
+{
+ int t0;
+
+ asm volatile (
+ "mul %[t0], %[out], %[envx]"
+ : [t0]"=r"(t0)
+ : [out]"r"(output), [envx]"r"((int) voice->envx));
+ asm volatile (
+ "mov %[t0], %[t0], asr #11 \n"
+ "bic %[t0], %[t0], #0x1 \n"
+ "mul %[a0], %[t0], %[v0] \n"
+ "mul %[a1], %[t0], %[v1] \n"
+ : [t0]"+r"(t0),
+ [a0]"=&r"(*amp_0), [a1]"=r"(*amp_1)
+ : [v0]"r"((int) voice->volume [0]),
+ [v1]"r"((int) voice->volume [1]));
+
+ return t0;
+}
+
+#else /* SPC_NOINTERP */
+
+#define SPC_LINEAR_INTERP
+static inline int linear_interp( int16_t const* samples, int32_t position )
+{
+ int output = (int) samples;
+ int y1;
+
+ asm volatile(
+ "mov %[y1], %[f], lsr #12 \n"
+ "eor %[f], %[f], %[y1], lsl #12 \n"
+ "add %[y1], %[y0], %[y1], lsl #1 \n"
+ "ldrsh %[y0], [%[y1], #2] \n"
+ "ldrsh %[y1], [%[y1], #4] \n"
+ "sub %[y1], %[y1], %[y0] \n"
+ "mul %[f], %[y1], %[f] \n"
+ "add %[y0], %[y0], %[f], asr #12 \n"
+ : [f]"+r"(position), [y0]"+r"(output), [y1]"=&r"(y1));
+
+ return output;
+}
+
+#define SPC_LINEAR_AMP
+static inline int linear_amp( struct voice_t* voice, int output,
+ int* amp_0, int* amp_1 )
+{
+ int t0;
+
+ asm volatile(
+ "mul %[t0], %[out], %[envx]"
+ : [t0]"=&r"(t0)
+ : [out]"r"(output), [envx]"r"(voice->envx));
+ asm volatile(
+ "mov %[t0], %[t0], asr #11 \n"
+ "mul %[a1], %[t0], %[v1] \n"
+ "mul %[a0], %[t0], %[v0] \n"
+ : [t0]"+r"(t0),
+ [a0]"=&r"(*amp_0), [a1]"=&r"(*amp_1)
+ : [v0]"r"((int) voice->volume [0]),
+ [v1]"r"((int) voice->volume [1]));
+
+ return t0;
+}
+
+#endif /* !SPC_NOINTERP */
+
+
+#if !SPC_NOECHO
+
+#define SPC_DSP_ECHO_APPLY
+
+/* Echo filter history */
+static int32_t fir_buf[FIR_BUF_CNT] IBSS_ATTR_SPC
+ __attribute__(( aligned(FIR_BUF_ALIGN*1) ));
+
+static inline void echo_init( struct Spc_Dsp* this )
+{
+ this->fir.ptr = fir_buf;
+ ci->memset( fir_buf, 0, sizeof fir_buf );
+}
+
+static inline void echo_apply( struct Spc_Dsp* this, uint8_t *echo_ptr,
+ int* out_0, int* out_1 )
+{
+ int t0 = GET_LE16SA( echo_ptr );
+ int t1 = GET_LE16SA( echo_ptr + 2 );
+
+ /* Keep last 8 samples */
+ int32_t *fir_ptr;
+ asm volatile (
+ "add %[p], %[t_p], #8 \n"
+ "bic %[t_p], %[p], %[mask] \n"
+ "str %[t0], [%[p], #-8] \n"
+ "str %[t1], [%[p], #-4] \n"
+ /* duplicate at +8 eliminates wrap checking below */
+ "str %[t0], [%[p], #56] \n"
+ "str %[t1], [%[p], #60] \n"
+ : [p]"=&r"(fir_ptr), [t_p]"+r"(this->fir.ptr)
+ : [t0]"r"(t0), [t1]"r"(t1), [mask]"i"(~FIR_BUF_MASK));
+
+ int32_t *fir_coeff = this->fir.coeff;
+
+ asm volatile (
+ "ldmia %[c]!, { r0-r1 } \n"
+ "ldmia %[p]!, { r4-r5 } \n"
+ "mul %[acc0], r0, %[acc0] \n"
+ "mul %[acc1], r0, %[acc1] \n"
+ "mla %[acc0], r4, r1, %[acc0] \n"
+ "mla %[acc1], r5, r1, %[acc1] \n"
+ "ldmia %[c]!, { r0-r1 } \n"
+ "ldmia %[p]!, { r2-r5 } \n"
+ "mla %[acc0], r2, r0, %[acc0] \n"
+ "mla %[acc1], r3, r0, %[acc1] \n"
+ "mla %[acc0], r4, r1, %[acc0] \n"
+ "mla %[acc1], r5, r1, %[acc1] \n"
+ "ldmia %[c]!, { r0-r1 } \n"
+ "ldmia %[p]!, { r2-r5 } \n"
+ "mla %[acc0], r2, r0, %[acc0] \n"
+ "mla %[acc1], r3, r0, %[acc1] \n"
+ "mla %[acc0], r4, r1, %[acc0] \n"
+ "mla %[acc1], r5, r1, %[acc1] \n"
+ "ldmia %[c]!, { r0-r1 } \n"
+ "ldmia %[p]!, { r2-r5 } \n"
+ "mla %[acc0], r2, r0, %[acc0] \n"
+ "mla %[acc1], r3, r0, %[acc1] \n"
+ "mla %[acc0], r4, r1, %[acc0] \n"
+ "mla %[acc1], r5, r1, %[acc1] \n"
+ : [acc0]"+r"(t0), [acc1]"+r"(t1),
+ [p]"+r"(fir_ptr), [c]"+r"(fir_coeff)
+ :
+ : "r0", "r1", "r2", "r3", "r4", "r5");
+
+ *out_0 = t0;
+ *out_1 = t1;
+}
+
+#endif /* SPC_NOECHO */
diff --git a/lib/rbcodec/codecs/libspc/cpu/spc_dsp_armv4.h b/lib/rbcodec/codecs/libspc/cpu/spc_dsp_armv4.h
new file mode 100644
index 0000000000..c9985e124a
--- /dev/null
+++ b/lib/rbcodec/codecs/libspc/cpu/spc_dsp_armv4.h
@@ -0,0 +1,45 @@
+/***************************************************************************
+ * __________ __ ___.
+ * Open \______ \ ____ ____ | | _\_ |__ _______ ___
+ * Source | _// _ \_/ ___\| |/ /| __ \ / _ \ \/ /
+ * Jukebox | | ( <_> ) \___| < | \_\ ( <_> > < <
+ * Firmware |____|_ /\____/ \___ >__|_ \|___ /\____/__/\_ \
+ * \/ \/ \/ \/ \/
+ * $Id$
+ *
+ * Copyright (C) 2007-2010 Michael Sevakis (jhMikeS)
+ *
+ * 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.
+ *
+ ****************************************************************************/
+#if !SPC_NOECHO
+
+#define SPC_DSP_ECHO_APPLY
+
+enum
+{
+ FIR_BUF_CNT = FIR_BUF_HALF * 2 * 2,
+ FIR_BUF_SIZE = FIR_BUF_CNT * sizeof ( int32_t ),
+ FIR_BUF_ALIGN = FIR_BUF_SIZE,
+ FIR_BUF_MASK = ~((FIR_BUF_ALIGN / 2) | (sizeof ( int32_t ) * 2 - 1))
+};
+
+/* Echo filter structure embedded in struct Spc_Dsp */
+struct echo_filter
+{
+ /* fir_buf [i + 8] == fir_buf [i], to avoid wrap checking in FIR code */
+ int32_t* ptr;
+ /* FIR history is interleaved with guard to eliminate wrap checking
+ * when convolving.
+ * |LL|RR|LL|RR|LL|RR|LL|RR|LL|RR|LL|RR|LL|RR|LL|RR|...
+ * |--|--|--|--|--|--|--|--|--|--|--|--|--|--|--|--| */
+ /* copy of echo FIR constants as int32_t, for faster access */
+ int32_t coeff [VOICE_COUNT];
+};
+#endif /* SPC_NOECHO */
diff --git a/lib/rbcodec/codecs/libspc/cpu/spc_dsp_armv6.c b/lib/rbcodec/codecs/libspc/cpu/spc_dsp_armv6.c
new file mode 100644
index 0000000000..2e3de87613
--- /dev/null
+++ b/lib/rbcodec/codecs/libspc/cpu/spc_dsp_armv6.c
@@ -0,0 +1,244 @@
+/***************************************************************************
+ * __________ __ ___.
+ * Open \______ \ ____ ____ | | _\_ |__ _______ ___
+ * Source | _// _ \_/ ___\| |/ /| __ \ / _ \ \/ /
+ * Jukebox | | ( <_> ) \___| < | \_\ ( <_> > < <
+ * Firmware |____|_ /\____/ \___ >__|_ \|___ /\____/__/\_ \
+ * \/ \/ \/ \/ \/
+ * $Id$
+ *
+ * Copyright (C) 2010 Michael Sevakis (jhMikeS)
+ *
+ * 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.
+ *
+ ****************************************************************************/
+#if !SPC_NOINTERP
+
+#define SPC_GAUSSIAN_FAST_INTERP
+static inline int gaussian_fast_interp( int16_t const* samples,
+ int32_t position,
+ int16_t const* fwd,
+ int16_t const* rev )
+{
+ int output;
+ int t0, t1, t2, t3;
+
+ asm volatile (
+ /* NOTE: often-unaligned accesses */
+ "ldr %[t0], [%[samp]] \n" /* t0=i0i1 */
+ "ldr %[t2], [%[fwd]] \n" /* t2=f0f1 */
+ "ldr %[t1], [%[samp], #4] \n" /* t1=i2i3 */
+ "ldr %[t3], [%[rev]] \n" /* t3=r0r1 */
+ "smuad %[out], %[t0], %[t2] \n" /* out=f0*i0+f1*i1 */
+ "smladx %[out], %[t1], %[t3], %[out] \n" /* out+=r1*i2+r0*i3 */
+ : [out]"=r"(output),
+ [t0]"=&r"(t0), [t1]"=&r"(t1), [t2]"=&r"(t2), [t3]"=r"(t3)
+ : [fwd]"r"(fwd), [rev]"r"(rev),
+ [samp]"r"(samples + (position >> 12)));
+
+ return output;
+}
+
+#define SPC_GAUSSIAN_FAST_AMP
+static inline int gaussian_fast_amp( struct voice_t* voice, int output,
+ int* amp_0, int* amp_1 )
+{
+ int t0;
+
+ asm volatile (
+ "mov %[t0], %[out], asr #(11-5) \n" /* To do >> 16 below */
+ "mul %[out], %[t0], %[envx] \n"
+ : [out]"+r"(output), [t0]"=&r"(t0)
+ : [envx]"r"((int) voice->envx));
+
+ asm volatile (
+ "smulwb %[a0], %[out], %[v0] \n" /* amp * vol >> 16 */
+ "smulwb %[a1], %[out], %[v1] \n"
+ : [a0]"=&r"(*amp_0), [a1]"=r"(*amp_1)
+ : [out]"r"(output),
+ [v0]"r"(voice->volume [0]),
+ [v1]"r"(voice->volume [1]));
+
+ return output >> 5; /* 'output' still 5 bits too big */
+}
+
+#define SPC_GAUSSIAN_SLOW_INTERP
+static inline int gaussian_slow_interp( int16_t const* samples,
+ int32_t position,
+ int16_t const* fwd,
+ int16_t const* rev )
+{
+ int output;
+ int t0, t1, t2, t3;
+
+ asm volatile (
+ /* NOTE: often-unaligned accesses */
+ "ldr %[t0], [%[samp]] \n" /* t0=i0i1 */
+ "ldr %[t2], [%[fwd]] \n" /* t2=f0f1 */
+ "ldr %[t1], [%[samp], #4] \n" /* t1=i2i3 */
+ "ldr %[t3], [%[rev]] \n" /* t3=f2f3 */
+ "smulbb %[out], %[t0], %[t2] \n" /* out=f0*i0 */
+ "smultt %[t0], %[t0], %[t2] \n" /* t0=f1*i1 */
+ "smulbt %[t2], %[t1], %[t3] \n" /* t2=r1*i2 */
+ "smultb %[t3], %[t1], %[t3] \n" /* t3=r0*i3 */
+ : [out]"=r"(output),
+ [t0]"=&r"(t0), [t1]"=&r"(t1), [t2]"=&r"(t2), [t3]"=r"(t3)
+ : [fwd]"r"(fwd), [rev]"r"(rev),
+ [samp]"r"(samples + (position >> 12)));
+
+ asm volatile (
+ "mov %[out], %[out], asr #12 \n"
+ "add %[t0], %[out], %[t0], asr #12 \n"
+ "add %[t2], %[t0], %[t2], asr #12 \n"
+ "pkhbt %[t0], %[t2], %[t3], asl #4 \n" /* t3[31:16], t2[15:0] */
+ "sadd16 %[t0], %[t0], %[t0] \n" /* t3[31:16]*2, t2[15:0]*2 */
+ "qsubaddx %[out], %[t0], %[t0] \n" /* out[15:0]=
+ * sat16(t3[31:16]+t2[15:0]) */
+ : [out]"+r"(output),
+ [t0]"+r"(t0), [t2]"+r"(t2), [t3]"+r"(t3));
+
+ /* output will be sign-extended in next step */
+ return output;
+}
+
+#define SPC_GAUSSIAN_SLOW_AMP
+static inline int gaussian_slow_amp( struct voice_t* voice, int output,
+ int* amp_0, int* amp_1 )
+{
+ asm volatile (
+ "smulbb %[out], %[out], %[envx]"
+ : [out]"+r"(output)
+ : [envx]"r"(voice->envx));
+
+ asm volatile (
+ "mov %[out], %[out], asr #11 \n"
+ "bic %[out], %[out], #0x1 \n"
+ "smulbb %[amp_0], %[out], %[v0] \n"
+ "smulbb %[amp_1], %[out], %[v1] \n"
+ : [out]"+r"(output),
+ [amp_0]"=&r"(*amp_0), [amp_1]"=r"(*amp_1)
+ : [v0]"r"(voice->volume[0]), [v1]"r"(voice->volume[1]));
+
+ return output;
+}
+
+#endif /* !SPC_NOINTERP */
+
+#if !SPC_NOECHO
+
+#define SPC_DSP_ECHO_APPLY
+
+/* Echo filter history */
+static int32_t fir_buf[FIR_BUF_CNT] IBSS_ATTR_SPC
+ __attribute__(( aligned(FIR_BUF_ALIGN*1) ));
+
+static inline void echo_init( struct Spc_Dsp* this )
+{
+ this->fir.ptr = fir_buf;
+ ci->memset( fir_buf, 0, sizeof fir_buf );
+}
+
+static inline void echo_apply(struct Spc_Dsp* this,
+ uint8_t* const echo_ptr, int* out_0, int* out_1)
+{
+ /* Keep last 8 samples */
+ int32_t* fir_ptr;
+ int t0;
+ asm volatile (
+ "ldr %[t0], [%[ep]] \n"
+ "add %[p], %[t_p], #4 \n"
+ "bic %[t_p], %[p], %[mask] \n"
+ "str %[t0], [%[p], #-4] \n"
+ /* duplicate at +8 eliminates wrap checking below */
+ "str %[t0], [%[p], #28] \n"
+ : [p]"=&r"(fir_ptr), [t_p]"+r"(this->fir.ptr),
+ [t0]"=&r"(t0)
+ : [ep]"r"(echo_ptr), [mask]"i"(~FIR_BUF_MASK));
+
+ int32_t* fir_coeff = (int32_t *)this->fir.coeff;
+
+ asm volatile ( /* L0R0 = acc0 */
+ "ldmia %[p]!, { r2-r5 } \n" /* L1R1-L4R4 = r2-r5 */
+ "ldmia %[c]!, { r0-r1 } \n" /* C0C1-C2C3 = r0-r1 */
+ "pkhbt %[acc0], %[t0], r2, asl #16 \n" /* L0R0,L1R1->L0L1,R0R1 */
+ "pkhtb r2, r2, %[t0], asr #16 \n"
+ "smuad %[acc0], %[acc0], r0 \n" /* acc0=L0*C0+L1*C1 */
+ "smuad %[acc1], r2, r0 \n" /* acc1=R0*C0+R1*C1 */
+ "pkhbt %[t0], r3, r4, asl #16 \n" /* L2R2,L3R3->L2L3,R2R3 */
+ "pkhtb r4, r4, r3, asr #16 \n"
+ "smlad %[acc0], %[t0], r1, %[acc0] \n" /* acc0+=L2*C2+L3*C3 */
+ "smlad %[acc1], r4, r1, %[acc1] \n" /* acc1+=R2*C2+R3*C3 */
+ "ldmia %[p], { r2-r4 } \n" /* L5R5-L7R7 = r2-r4 */
+ "ldmia %[c], { r0-r1 } \n" /* C4C5-C6C7 = r0-r1 */
+ "pkhbt %[t0], r5, r2, asl #16 \n" /* L4R4,L5R5->L4L5,R4R5 */
+ "pkhtb r2, r2, r5, asr #16 \n"
+ "smlad %[acc0], %[t0], r0, %[acc0] \n" /* acc0+=L4*C4+L5*C5 */
+ "smlad %[acc1], r2, r0, %[acc1] \n" /* acc1+=R4*C4+R5*C5 */
+ "pkhbt %[t0], r3, r4, asl #16 \n" /* L6R6,L7R7->L6L7,R6R7 */
+ "pkhtb r4, r4, r3, asr #16 \n"
+ "smlad %[acc0], %[t0], r1, %[acc0] \n" /* acc0+=L6*C6+L7*C7 */
+ "smlad %[acc1], r4, r1, %[acc1] \n" /* acc1+=R6*C6+R7*C7 */
+ : [t0]"+r"(t0), [acc0]"=&r"(*out_0), [acc1]"=&r"(*out_1),
+ [p]"+r"(fir_ptr), [c]"+r"(fir_coeff)
+ :
+ : "r0", "r1", "r2", "r3", "r4", "r5");
+}
+
+#define SPC_DSP_ECHO_FEEDBACK
+static inline void echo_feedback(struct Spc_Dsp* this, uint8_t* echo_ptr,
+ int echo_0, int echo_1, int fb_0, int fb_1)
+{
+ int e0, e1;
+ asm volatile (
+ "mov %[e0], %[ei0], asl #7 \n"
+ "mov %[e1], %[ei1], asl #7 \n"
+ "mla %[e0], %[fb0], %[ef], %[e0] \n"
+ "mla %[e1], %[fb1], %[ef], %[e1] \n"
+ : [e0]"=&r"(e0), [e1]"=&r"(e1)
+ : [ei0]"r"(echo_0), [ei1]"r"(echo_1),
+ [fb0]"r"(fb_0), [fb1]"r"(fb_1),
+ [ef]"r"((int)this->r.g.echo_feedback));
+ asm volatile (
+ "ssat %[e0], #16, %[e0], asr #14 \n"
+ "ssat %[e1], #16, %[e1], asr #14 \n"
+ "pkhbt %[e0], %[e0], %[e1], lsl #16 \n"
+ "str %[e0], [%[ep]] \n"
+ : [e0]"+r"(e0), [e1]"+r"(e1)
+ : [ep]"r"((int32_t *)echo_ptr));
+}
+
+#define SPC_DSP_GENERATE_OUTPUT
+static inline void echo_output( struct Spc_Dsp* this, int global_muting,
+ int global_vol_0, int global_vol_1, int chans_0, int chans_1,
+ int fb_0, int fb_1, int* out_0, int* out_1 )
+{
+ int t0, t1;
+
+ asm volatile (
+ "mul %[t0], %[gv0], %[ch0] \n"
+ "mul %[t1], %[gv1], %[ch1] \n"
+ : [t0]"=&r"(t0), [t1]"=r"(t1)
+ : [gv0]"r"(global_vol_0), [gv1]"r"(global_vol_1),
+ [ch0]"r"(chans_0), [ch1]"r"(chans_1));
+ asm volatile (
+ "mla %[t0], %[i0], %[ev0], %[t0] \n"
+ "mla %[t1], %[i1], %[ev1], %[t1] \n"
+ : [t0]"+r"(t0), [t1]"+r"(t1)
+ : [i0]"r"(fb_0), [i1]"r"(fb_1),
+ [ev0]"r"((int)this->r.g.echo_volume_0),
+ [ev1]"r"((int)this->r.g.echo_volume_1));
+ asm volatile (
+ "mov %[o0], %[t0], asr %[gm] \n"
+ "mov %[o1], %[t1], asr %[gm] \n"
+ : [o0]"=&r"(*out_0), [o1]"=r"(*out_1)
+ : [t0]"r"(t0), [t1]"r"(t1),
+ [gm]"r"(global_muting));
+}
+
+#endif /* SPC_NOECHO */
diff --git a/lib/rbcodec/codecs/libspc/cpu/spc_dsp_armv6.h b/lib/rbcodec/codecs/libspc/cpu/spc_dsp_armv6.h
new file mode 100644
index 0000000000..a36d8166c2
--- /dev/null
+++ b/lib/rbcodec/codecs/libspc/cpu/spc_dsp_armv6.h
@@ -0,0 +1,45 @@
+/***************************************************************************
+ * __________ __ ___.
+ * Open \______ \ ____ ____ | | _\_ |__ _______ ___
+ * Source | _// _ \_/ ___\| |/ /| __ \ / _ \ \/ /
+ * Jukebox | | ( <_> ) \___| < | \_\ ( <_> > < <
+ * Firmware |____|_ /\____/ \___ >__|_ \|___ /\____/__/\_ \
+ * \/ \/ \/ \/ \/
+ * $Id$
+ *
+ * Copyright (C) 2010 Michael Sevakis (jhMikeS)
+ *
+ * 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.
+ *
+ ****************************************************************************/
+#if !SPC_NOECHO
+
+#define SPC_DSP_ECHO_APPLY
+
+enum
+{
+ FIR_BUF_CNT = FIR_BUF_HALF * 2,
+ FIR_BUF_SIZE = FIR_BUF_CNT * sizeof ( int32_t ),
+ FIR_BUF_ALIGN = FIR_BUF_SIZE,
+ FIR_BUF_MASK = ~((FIR_BUF_ALIGN / 2) | (sizeof ( int32_t ) - 1))
+};
+
+/* Echo filter structure embedded in struct Spc_Dsp */
+struct echo_filter
+{ /* fir_buf [i + 8] == fir_buf [i], to avoid wrap checking in FIR code */
+ int32_t* ptr;
+ /* FIR history is interleaved with guard to eliminate wrap checking
+ * when convolving.
+ * |LR|LR|LR|LR|LR|LR|LR|LR|--|--|--|--|--|--|--|--| */
+ /* copy of echo FIR constants as int16_t, loaded as int32 for
+ * halfword, packed multiples */
+ int16_t coeff [VOICE_COUNT];
+};
+
+#endif /* SPC_NOECHO */
diff --git a/lib/rbcodec/codecs/libspc/cpu/spc_dsp_coldfire.c b/lib/rbcodec/codecs/libspc/cpu/spc_dsp_coldfire.c
new file mode 100644
index 0000000000..b0d14d157e
--- /dev/null
+++ b/lib/rbcodec/codecs/libspc/cpu/spc_dsp_coldfire.c
@@ -0,0 +1,198 @@
+/***************************************************************************
+ * __________ __ ___.
+ * Open \______ \ ____ ____ | | _\_ |__ _______ ___
+ * Source | _// _ \_/ ___\| |/ /| __ \ / _ \ \/ /
+ * Jukebox | | ( <_> ) \___| < | \_\ ( <_> > < <
+ * Firmware |____|_ /\____/ \___ >__|_ \|___ /\____/__/\_ \
+ * \/ \/ \/ \/ \/
+ * $Id$
+ *
+ * Copyright (C) 2007 Michael Sevakis (jhMikeS)
+ *
+ * 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.
+ *
+ ****************************************************************************/
+#if SPC_NOINTERP
+
+#define SPC_LINEAR_INTERP
+static inline int linear_interp( int16_t const* samples, int32_t position )
+{
+ uint32_t f = position;
+ int32_t y0, y1;
+
+ /**
+ * output = y0 + f*y1 - f*y0
+ */
+ asm volatile (
+ "move.l %[f], %[y1] \n" /* separate frac and whole */
+ "and.l #0xfff, %[f] \n"
+ "asr.l %[sh], %[y1] \n"
+ "move.l 2(%[s], %[y1].l*2), %[y1] \n" /* y0=upper, y1=lower */
+ "mac.w %[f]l, %[y1]l, %%acc0 \n" /* %acc0 = f*y1 */
+ "msac.w %[f]l, %[y1]u, %%acc0 \n" /* %acc0 -= f*y0 */
+ "swap %[y1] \n" /* separate out y0 and sign extend */
+ "movea.w %[y1], %[y0] \n"
+ "movclr.l %%acc0, %[y1] \n" /* fetch, scale down, add y0 */
+ "asr.l %[sh], %[y1] \n" /* output = y0 + (result >> 12) */
+ "add.l %[y0], %[y1] \n"
+ : [f]"+d"(f), [y0]"=&a"(y0), [y1]"=&d"(y1)
+ : [s]"a"(samples), [sh]"d"(12));
+
+ return y1;
+}
+
+#define SPC_LINEAR_AMP
+static inline int linear_amp( struct voice_t* voice, int output,
+ int* amp_0, int* amp_1 )
+{
+ asm volatile (
+ "mac.w %[out]l, %[envx]l, %%acc0"
+ :
+ : [out]"r"(output), [envx]"r"(voice->envx));
+ asm volatile (
+ "movclr.l %%acc0, %[out] \n"
+ "asr.l #8, %[out] \n"
+ "mac.l %[v0], %[out], %%acc0 \n"
+ "mac.l %[v1], %[out], %%acc1 \n"
+ "asr.l #3, %[out] \n"
+ : [out]"+r"(output)
+ : [v0]"r"((int) voice->volume [0]),
+ [v1]"r"((int) voice->volume [1]));
+ asm volatile (
+ "movclr.l %%acc0, %[a0] \n"
+ "asr.l #3, %[a0] \n"
+ "movclr.l %%acc1, %[a1] \n"
+ "asr.l #3, %[a1] \n"
+ : [a0]"=d"(*amp_0), [a1]"=d"(*amp_1));
+
+ return output;
+}
+
+#endif /* SPC_NOINTERP */
+
+
+#if !SPC_NOECHO
+
+#define SPC_DSP_ECHO_APPLY
+
+/* Echo filter history */
+static int32_t fir_buf[FIR_BUF_CNT] IBSS_ATTR_SPC
+ __attribute__(( aligned(FIR_BUF_ALIGN*1) ));
+
+static inline void echo_init( struct Spc_Dsp* this )
+{
+ /* Initialize mask register with the buffer address mask */
+ asm volatile ("move.l %0, %%mask" : : "i"(FIR_BUF_MASK));
+ this->fir.ptr = fir_buf;
+ this->fir.hist_ptr = &fir_buf [1];
+ ci->memset( fir_buf, 0, sizeof fir_buf );
+}
+
+static inline void echo_apply( struct Spc_Dsp* this, uint8_t* echo_ptr,
+ int* out_0, int* out_1 )
+{
+ int t0, t1, t2;
+
+ t1 = swap_odd_even32( *(int32_t *)echo_ptr );
+
+ /* Keep last 8 samples */
+ *this->fir.ptr = t1;
+ this->fir.ptr = this->fir.hist_ptr;
+
+ asm volatile (
+ "move.l (%[c]) , %[t2] \n"
+ "mac.w %[t1]u, %[t2]u, <<, (%[p])+&, %[t0], %%acc0 \n"
+ "mac.w %[t1]l, %[t2]u, <<, (%[p])& , %[t1], %%acc1 \n"
+ "mac.w %[t0]u, %[t2]l, << , %%acc0 \n"
+ "mac.w %[t0]l, %[t2]l, <<, 4(%[c]) , %[t2], %%acc1 \n"
+ "mac.w %[t1]u, %[t2]u, <<, 4(%[p])& , %[t0], %%acc0 \n"
+ "mac.w %[t1]l, %[t2]u, <<, 8(%[p])& , %[t1], %%acc1 \n"
+ "mac.w %[t0]u, %[t2]l, << , %%acc0 \n"
+ "mac.w %[t0]l, %[t2]l, <<, 8(%[c]) , %[t2], %%acc1 \n"
+ "mac.w %[t1]u, %[t2]u, <<, 12(%[p])& , %[t0], %%acc0 \n"
+ "mac.w %[t1]l, %[t2]u, <<, 16(%[p])& , %[t1], %%acc1 \n"
+ "mac.w %[t0]u, %[t2]l, << , %%acc0 \n"
+ "mac.w %[t0]l, %[t2]l, <<, 12(%[c]) , %[t2], %%acc1 \n"
+ "mac.w %[t1]u, %[t2]u, <<, 20(%[p])& , %[t0], %%acc0 \n"
+ "mac.w %[t1]l, %[t2]u, << , %%acc1 \n"
+ "mac.w %[t0]u, %[t2]l, << , %%acc0 \n"
+ "mac.w %[t0]l, %[t2]l, << , %%acc1 \n"
+ : [t0]"=&r"(t0), [t1]"+r"(t1), [t2]"=&r"(t2),
+ [p]"+a"(this->fir.hist_ptr)
+ : [c]"a"(this->fir.coeff));
+ asm volatile (
+ "movclr.l %%acc0, %[o0] \n"
+ "movclr.l %%acc1, %[o1] \n"
+ "mac.l %[ev0], %[o0], >>, %%acc2 \n" /* echo volume */
+ "mac.l %[ev1], %[o1], >>, %%acc3 \n"
+ : [o0]"=&r"(*out_0), [o1]"=&r"(*out_1)
+ : [ev0]"r"((int) this->r.g.echo_volume_0),
+ [ev1]"r"((int) this->r.g.echo_volume_1));
+}
+
+#define SPC_DSP_ECHO_FEEDBACK
+static inline void echo_feedback( struct Spc_Dsp* this, uint8_t* echo_ptr,
+ int echo_0, int echo_1, int fb_0, int fb_1 )
+{
+ asm volatile (
+ /* scale echo voices; saturate if overflow */
+ "mac.l %[sh], %[e1] , %%acc1 \n"
+ "mac.l %[sh], %[e0] , %%acc0 \n"
+ /* add scaled output from FIR filter */
+ "mac.l %[fb1], %[ef], <<, %%acc1 \n"
+ "mac.l %[fb0], %[ef], <<, %%acc0 \n"
+ :
+ : [e0]"d"(echo_0), [e1]"d"(echo_1),
+ [fb0]"r"(fb_0), [fb1]"r"(fb_1),
+ [ef]"r"((int)this->r.g.echo_feedback),
+ [sh]"r"(1 << 9));
+ /* swap and fetch feedback results */
+ int t0;
+ asm volatile(
+ "move.l #0x00ff00ff, %[t0] \n"
+ "movclr.l %%acc1, %[e1] \n"
+ "swap.w %[e1] \n"
+ "movclr.l %%acc0, %[e0] \n"
+ "move.w %[e1], %[e0] \n"
+ "and.l %[e0], %[t0] \n"
+ "eor.l %[t0], %[e0] \n"
+ "lsl.l #8, %[t0] \n"
+ "lsr.l #8, %[e0] \n"
+ "or.l %[e0], %[t0] \n"
+ : [e0]"=&d"(echo_0), [e1]"=&d"(echo_1),
+ [t0]"=&d"(t0));
+
+ /* save final feedback into echo buffer */
+ *(int32_t *)echo_ptr = t0;
+}
+
+#define SPC_DSP_GENERATE_OUTPUT
+static inline void echo_output( struct Spc_Dsp* this, int global_muting,
+ int global_vol_0, int global_vol_1, int chans_0, int chans_1,
+ int fb_0, int fb_1, int* out_0, int* out_1 )
+{
+ asm volatile (
+ "mac.l %[ch0], %[gv0], %%acc2 \n" /* global volume */
+ "mac.l %[ch1], %[gv1], %%acc3 \n"
+ :
+ : [ch0]"r"(chans_0), [gv0]"r"(global_vol_0),
+ [ch1]"r"(chans_1), [gv1]"r"(global_vol_1));
+ asm volatile (
+ "movclr.l %%acc2, %[a0] \n" /* fetch mixed output */
+ "movclr.l %%acc3, %[a1] \n"
+ "asr.l %[gm], %[a0] \n" /* scale by global_muting shift */
+ "asr.l %[gm], %[a1] \n"
+ : [a0]"=&d"(*out_0), [a1]"=&d"(*out_1)
+ : [gm]"d"(global_muting));
+
+ /* scaled echo is stored in %acc2 and %acc3 */
+ (void)this; (void)fb_0; (void)fb_1;
+}
+
+#endif /* !SPC_NOECHO */
diff --git a/lib/rbcodec/codecs/libspc/cpu/spc_dsp_coldfire.h b/lib/rbcodec/codecs/libspc/cpu/spc_dsp_coldfire.h
new file mode 100644
index 0000000000..f9aafabd18
--- /dev/null
+++ b/lib/rbcodec/codecs/libspc/cpu/spc_dsp_coldfire.h
@@ -0,0 +1,45 @@
+/***************************************************************************
+ * __________ __ ___.
+ * Open \______ \ ____ ____ | | _\_ |__ _______ ___
+ * Source | _// _ \_/ ___\| |/ /| __ \ / _ \ \/ /
+ * Jukebox | | ( <_> ) \___| < | \_\ ( <_> > < <
+ * Firmware |____|_ /\____/ \___ >__|_ \|___ /\____/__/\_ \
+ * \/ \/ \/ \/ \/
+ * $Id$
+ *
+ * Copyright (C) 2007 Michael Sevakis (jhMikeS)
+ *
+ * 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.
+ *
+ ****************************************************************************/
+#if !SPC_NOECHO
+
+#define SPC_DSP_ECHO_APPLY
+
+enum
+{
+ FIR_BUF_CNT = FIR_BUF_HALF,
+ FIR_BUF_SIZE = FIR_BUF_CNT * sizeof ( int32_t ),
+ FIR_BUF_ALIGN = FIR_BUF_SIZE * 2,
+ FIR_BUF_MASK = ~((FIR_BUF_ALIGN / 2) | (sizeof ( int32_t ) - 1))
+};
+
+/* Echo filter structure embedded in struct Spc_Dsp */
+struct echo_filter
+{
+ /* FIR history is interleaved. Hardware handles wrapping by mask.
+ * |LR|LR|LR|LR|LR|LR|LR|LR| */
+ int32_t* ptr;
+ /* wrapped address just behind current position -
+ allows mac.w to increment and mask ptr */
+ int32_t* hist_ptr;
+ /* copy of echo FIR constants as int16_t for use with mac.w */
+ int16_t coeff [VOICE_COUNT];
+};
+#endif /* !SPC_NOECHO */
diff --git a/lib/rbcodec/codecs/libspc/spc_codec.h b/lib/rbcodec/codecs/libspc/spc_codec.h
index 7f6b6e2e9f..a8eee6bfef 100644
--- a/lib/rbcodec/codecs/libspc/spc_codec.h
+++ b/lib/rbcodec/codecs/libspc/spc_codec.h
@@ -213,7 +213,9 @@ struct cpu_ram_t
#define RAM ram.ram
extern struct cpu_ram_t ram;
-long CPU_run( THIS, long start_time ) ICODE_ATTR_SPC;
+long CPU_run( THIS, long start_time )
+ ICODE_ATTR_SPC;
+
void CPU_Init( THIS );
/* The DSP portion (awe!) */
@@ -261,6 +263,7 @@ struct globals_t
char unused9 [2];
};
+enum { ENV_RATE_INIT = 0x7800 };
enum state_t
{ /* -1, 0, +1 allows more efficient if statements */
state_decay = -1,
@@ -278,64 +281,61 @@ struct cache_entry_t
};
enum { BRR_BLOCK_SIZE = 16 };
-enum { BRR_CACHE_SIZE = 0x20000 + 32} ;
+enum { BRR_CACHE_SIZE = 0x20000 + 32};
+
+#if SPC_BRRCACHE
+struct voice_wave_t
+{
+ int16_t const* samples; /* decoded samples in cache */
+ long position; /* position in samples buffer, 12-bit frac */
+ long end; /* end position in samples buffer */
+ int loop; /* length of looping area */
+ unsigned block_header; /* header byte from current BRR block */
+ uint8_t const* addr; /* BRR waveform address in RAM */
+};
+#else /* !SPC_BRRCACHE */
+struct voice_wave_t
+{
+ int16_t samples [3 + BRR_BLOCK_SIZE + 1]; /* last decoded block */
+ int32_t position; /* position in samples buffer, 12-bit frac */
+ unsigned block_header; /* header byte from current BRR block */
+ uint8_t const* addr; /* BRR waveform address in RAM */
+};
+#endif /* SPC_BRRCACHE */
struct voice_t
{
-#if SPC_BRRCACHE
- int16_t const* samples;
- long wave_end;
- int wave_loop;
-#else
- int16_t samples [3 + BRR_BLOCK_SIZE + 1];
- int block_header; /* header byte from current block */
-#endif
- uint8_t const* addr;
+ struct voice_wave_t wave;
short volume [2];
- long position;/* position in samples buffer, with 12-bit fraction */
short envx;
short env_mode;
short env_timer;
short key_on_delay;
+ short rate;
};
-#if SPC_BRRCACHE
-/* a little extra for samples that go past end */
-extern int16_t BRRcache [BRR_CACHE_SIZE];
+#if !SPC_NOECHO
+enum { FIR_BUF_HALF = 8 };
#endif
-enum { FIR_BUF_HALF = 8 };
+struct Spc_Dsp;
-#if defined(CPU_COLDFIRE)
-/* global because of the large aligment requirement for hardware masking -
- * L-R interleaved 16-bit samples for easy loading and mac.w use.
- */
-enum
-{
- FIR_BUF_CNT = FIR_BUF_HALF,
- FIR_BUF_SIZE = FIR_BUF_CNT * sizeof ( int32_t ),
- FIR_BUF_ALIGN = FIR_BUF_SIZE * 2,
- FIR_BUF_MASK = ~((FIR_BUF_ALIGN / 2) | (sizeof ( int32_t ) - 1))
-};
-#elif defined (CPU_ARM)
+/* These must go before the definition of struct Spc_Dsp because a
+ definition of struct echo_filter is required. Only declarations
+ are created unless SPC_DSP_C is defined before including these. */
+#if defined(CPU_ARM)
#if ARM_ARCH >= 6
-enum
-{
- FIR_BUF_CNT = FIR_BUF_HALF * 2,
- FIR_BUF_SIZE = FIR_BUF_CNT * sizeof ( int32_t ),
- FIR_BUF_ALIGN = FIR_BUF_SIZE,
- FIR_BUF_MASK = ~((FIR_BUF_ALIGN / 2) | (sizeof ( int32_t ) - 1))
-};
+#include "cpu/spc_dsp_armv6.h"
#else
-enum
-{
- FIR_BUF_CNT = FIR_BUF_HALF * 2 * 2,
- FIR_BUF_SIZE = FIR_BUF_CNT * sizeof ( int32_t ),
- FIR_BUF_ALIGN = FIR_BUF_SIZE,
- FIR_BUF_MASK = ~((FIR_BUF_ALIGN / 2) | (sizeof ( int32_t ) * 2 - 1))
-};
-#endif /* ARM_ARCH */
-#endif /* CPU_* */
+#include "cpu/spc_dsp_armv4.h"
+#endif
+#elif defined (CPU_COLDFIRE)
+#include "cpu/spc_dsp_coldfire.h"
+#endif
+
+/* Above may still use generic implementations. Also defines final
+ function names. */
+#include "spc_dsp_generic.h"
struct Spc_Dsp
{
@@ -347,47 +347,15 @@ struct Spc_Dsp
int16_t align;
} r;
- unsigned echo_pos;
int keys_down;
int noise_count;
uint16_t noise; /* also read as int16_t */
-
-#if defined(CPU_COLDFIRE)
- /* FIR history is interleaved. Hardware handles wrapping by mask.
- * |LR|LR|LR|LR|LR|LR|LR|LR| */
- int32_t *fir_ptr;
- /* wrapped address just behind current position -
- allows mac.w to increment and mask fir_ptr */
- int32_t *last_fir_ptr;
- /* copy of echo FIR constants as int16_t for use with mac.w */
- int16_t fir_coeff [VOICE_COUNT];
-#elif defined (CPU_ARM)
- /* fir_buf [i + 8] == fir_buf [i], to avoid wrap checking in FIR code */
- int32_t *fir_ptr;
-#if ARM_ARCH >= 6
- /* FIR history is interleaved with guard to eliminate wrap checking
- * when convolving.
- * |LR|LR|LR|LR|LR|LR|LR|LR|--|--|--|--|--|--|--|--| */
- /* copy of echo FIR constants as int16_t, loaded as int32 for
- * halfword, packed multiples */
- int16_t fir_coeff [VOICE_COUNT];
-#else
- /* FIR history is interleaved with guard to eliminate wrap checking
- * when convolving.
- * |LL|RR|LL|RR|LL|RR|LL|RR|LL|RR|LL|RR|LL|RR|LL|RR|...
- * |--|--|--|--|--|--|--|--|--|--|--|--|--|--|--|--| */
- /* copy of echo FIR constants as int32_t, for faster access */
- int32_t fir_coeff [VOICE_COUNT];
-#endif /* ARM_ARCH */
-#else /* Unoptimized CPU */
- /* fir_buf [i + 8] == fir_buf [i], to avoid wrap checking in FIR code */
- int fir_pos; /* (0 to 7) */
- int fir_buf [FIR_BUF_HALF * 2] [2];
- /* copy of echo FIR constants as int, for faster access */
- int fir_coeff [VOICE_COUNT];
-#endif
-
struct voice_t voice_state [VOICE_COUNT];
+
+#if !SPC_NOECHO
+ unsigned echo_pos;
+ struct echo_filter fir;
+#endif /* !SPC_NOECHO */
#if SPC_BRRCACHE
uint8_t oldsize;
@@ -396,7 +364,9 @@ struct Spc_Dsp
#endif
};
-void DSP_run_( struct Spc_Dsp* this, long count, int32_t* out_buf ) ICODE_ATTR_SPC;
+void DSP_run_( struct Spc_Dsp* this, long count, int32_t* out_buf )
+ ICODE_ATTR_SPC;
+
void DSP_reset( struct Spc_Dsp* this );
static inline void DSP_run( struct Spc_Dsp* this, long count, int32_t* out )
@@ -474,7 +444,8 @@ void SPC_Init( THIS );
int SPC_load_spc( THIS, const void* data, long size );
/**************** DSP interaction ****************/
-void DSP_write( struct Spc_Dsp* this, int i, int data ) ICODE_ATTR_SPC;
+void DSP_write( struct Spc_Dsp* this, int i, int data )
+ ICODE_ATTR_SPC;
static inline int DSP_read( struct Spc_Dsp* this, int i )
{
@@ -482,10 +453,14 @@ static inline int DSP_read( struct Spc_Dsp* this, int i )
return this->r.reg [i];
}
-int SPC_read( THIS, unsigned addr, long const time ) ICODE_ATTR_SPC;
-void SPC_write( THIS, unsigned addr, int data, long const time ) ICODE_ATTR_SPC;
+int SPC_read( THIS, unsigned addr, long const time )
+ ICODE_ATTR_SPC;
+
+void SPC_write( THIS, unsigned addr, int data, long const time )
+ ICODE_ATTR_SPC;
/**************** Sample generation ****************/
-int SPC_play( THIS, long count, int32_t* out ) ICODE_ATTR_SPC;
+int SPC_play( THIS, long count, int32_t* out )
+ ICODE_ATTR_SPC;
#endif /* _SPC_CODEC_H_ */
diff --git a/lib/rbcodec/codecs/libspc/spc_cpu.c b/lib/rbcodec/codecs/libspc/spc_cpu.c
index 23dcc257de..dbbc6cda0f 100644
--- a/lib/rbcodec/codecs/libspc/spc_cpu.c
+++ b/lib/rbcodec/codecs/libspc/spc_cpu.c
@@ -113,9 +113,7 @@ enum { st_c = 0x01 };
long CPU_run( THIS, long start_time )
{
-#if 0
ENTER_TIMER(cpu);
-#endif
register long spc_time_ = start_time;
@@ -1036,9 +1034,7 @@ out_of_time:
this->r.x = (uint8_t) x;
this->r.y = (uint8_t) y;
-#if 0
EXIT_TIMER(cpu);
-#endif
return spc_time_;
}
diff --git a/lib/rbcodec/codecs/libspc/spc_dsp.c b/lib/rbcodec/codecs/libspc/spc_dsp.c
index 6350c4c331..c94fbc990e 100644
--- a/lib/rbcodec/codecs/libspc/spc_dsp.c
+++ b/lib/rbcodec/codecs/libspc/spc_dsp.c
@@ -27,15 +27,103 @@
#include "spc_codec.h"
#include "spc_profiler.h"
-#if defined(CPU_COLDFIRE) || defined (CPU_ARM)
-int32_t fir_buf[FIR_BUF_CNT] IBSS_ATTR_SPC
- __attribute__((aligned(FIR_BUF_ALIGN*1)));
+#define CLAMP16( n ) clip_sample_16( n )
+
+#if defined(CPU_ARM)
+#if ARM_ARCH >= 6
+#include "cpu/spc_dsp_armv6.c"
+#else
+#include "cpu/spc_dsp_armv4.c"
#endif
-#if SPC_BRRCACHE
-/* a little extra for samples that go past end */
-int16_t BRRcache [BRR_CACHE_SIZE] CACHEALIGN_ATTR;
+#elif defined (CPU_COLDFIRE)
+#include "cpu/spc_dsp_coldfire.c"
#endif
+/* Above may still use generic implementations. Also defines final
+ function names. */
+#include "spc_dsp_generic.c"
+
+/* each rate divides exactly into 0x7800 without remainder */
+static unsigned short const env_rates [0x20] ICONST_ATTR_SPC =
+{
+ 0x0000, 0x000F, 0x0014, 0x0018, 0x001E, 0x0028, 0x0030, 0x003C,
+ 0x0050, 0x0060, 0x0078, 0x00A0, 0x00C0, 0x00F0, 0x0140, 0x0180,
+ 0x01E0, 0x0280, 0x0300, 0x03C0, 0x0500, 0x0600, 0x0780, 0x0A00,
+ 0x0C00, 0x0F00, 0x1400, 0x1800, 0x1E00, 0x2800, 0x3C00, 0x7800
+};
+
+#if !SPC_NOINTERP
+/* Interleved gauss table (to improve cache coherency). */
+/* gauss [i * 2 + j] = normal_gauss [(1 - j) * 256 + i] */
+static int16_t const gauss_table [512] ICONST_ATTR_SPC MEM_ALIGN_ATTR =
+{
+ 370,1305, 366,1305, 362,1304, 358,1304,
+ 354,1304, 351,1304, 347,1304, 343,1303,
+ 339,1303, 336,1303, 332,1302, 328,1302,
+ 325,1301, 321,1300, 318,1300, 314,1299,
+ 311,1298, 307,1297, 304,1297, 300,1296,
+ 297,1295, 293,1294, 290,1293, 286,1292,
+ 283,1291, 280,1290, 276,1288, 273,1287,
+ 270,1286, 267,1284, 263,1283, 260,1282,
+ 257,1280, 254,1279, 251,1277, 248,1275,
+ 245,1274, 242,1272, 239,1270, 236,1269,
+ 233,1267, 230,1265, 227,1263, 224,1261,
+ 221,1259, 218,1257, 215,1255, 212,1253,
+ 210,1251, 207,1248, 204,1246, 201,1244,
+ 199,1241, 196,1239, 193,1237, 191,1234,
+ 188,1232, 186,1229, 183,1227, 180,1224,
+ 178,1221, 175,1219, 173,1216, 171,1213,
+ 168,1210, 166,1207, 163,1205, 161,1202,
+ 159,1199, 156,1196, 154,1193, 152,1190,
+ 150,1186, 147,1183, 145,1180, 143,1177,
+ 141,1174, 139,1170, 137,1167, 134,1164,
+ 132,1160, 130,1157, 128,1153, 126,1150,
+ 124,1146, 122,1143, 120,1139, 118,1136,
+ 117,1132, 115,1128, 113,1125, 111,1121,
+ 109,1117, 107,1113, 106,1109, 104,1106,
+ 102,1102, 100,1098, 99,1094, 97,1090,
+ 95,1086, 94,1082, 92,1078, 90,1074,
+ 89,1070, 87,1066, 86,1061, 84,1057,
+ 83,1053, 81,1049, 80,1045, 78,1040,
+ 77,1036, 76,1032, 74,1027, 73,1023,
+ 71,1019, 70,1014, 69,1010, 67,1005,
+ 66,1001, 65, 997, 64, 992, 62, 988,
+ 61, 983, 60, 978, 59, 974, 58, 969,
+ 56, 965, 55, 960, 54, 955, 53, 951,
+ 52, 946, 51, 941, 50, 937, 49, 932,
+ 48, 927, 47, 923, 46, 918, 45, 913,
+ 44, 908, 43, 904, 42, 899, 41, 894,
+ 40, 889, 39, 884, 38, 880, 37, 875,
+ 36, 870, 36, 865, 35, 860, 34, 855,
+ 33, 851, 32, 846, 32, 841, 31, 836,
+ 30, 831, 29, 826, 29, 821, 28, 816,
+ 27, 811, 27, 806, 26, 802, 25, 797,
+ 24, 792, 24, 787, 23, 782, 23, 777,
+ 22, 772, 21, 767, 21, 762, 20, 757,
+ 20, 752, 19, 747, 19, 742, 18, 737,
+ 17, 732, 17, 728, 16, 723, 16, 718,
+ 15, 713, 15, 708, 15, 703, 14, 698,
+ 14, 693, 13, 688, 13, 683, 12, 678,
+ 12, 674, 11, 669, 11, 664, 11, 659,
+ 10, 654, 10, 649, 10, 644, 9, 640,
+ 9, 635, 9, 630, 8, 625, 8, 620,
+ 8, 615, 7, 611, 7, 606, 7, 601,
+ 6, 596, 6, 592, 6, 587, 6, 582,
+ 5, 577, 5, 573, 5, 568, 5, 563,
+ 4, 559, 4, 554, 4, 550, 4, 545,
+ 4, 540, 3, 536, 3, 531, 3, 527,
+ 3, 522, 3, 517, 2, 513, 2, 508,
+ 2, 504, 2, 499, 2, 495, 2, 491,
+ 2, 486, 1, 482, 1, 477, 1, 473,
+ 1, 469, 1, 464, 1, 460, 1, 456,
+ 1, 451, 1, 447, 1, 443, 1, 439,
+ 0, 434, 0, 430, 0, 426, 0, 422,
+ 0, 418, 0, 414, 0, 410, 0, 405,
+ 0, 401, 0, 397, 0, 393, 0, 389,
+ 0, 385, 0, 381, 0, 378, 0, 374,
+};
+#endif /* !SPC_NOINTERP */
+
void DSP_write( struct Spc_Dsp* this, int i, int data )
{
assert( (unsigned) i < REGISTER_COUNT );
@@ -51,230 +139,395 @@ void DSP_write( struct Spc_Dsp* this, int i, int data )
v->volume [0] = left;
v->volume [1] = right;
}
+ else if ( low < 4 ) /* voice rates */
+ {
+ struct voice_t* v = this->voice_state + high;
+ v->rate = GET_LE16A( this->r.voice[high].rate ) & 0x3fff;
+ }
+#if !SPC_NOECHO
else if ( low == 0x0F ) /* fir coefficients */
{
- this->fir_coeff [7 - high] = (int8_t) data; /* sign-extend */
+ this->fir.coeff [7 - high] = (int8_t) data; /* sign-extend */
}
+#endif /* !SPC_NOECHO */
}
-#define CLAMP16( n ) clip_sample_16( n )
+/* Decode BRR block */
+static inline void
+decode_brr_block( struct voice_t* voice, uint8_t const* addr, int16_t* out )
+{
+ /* header */
+ unsigned block_header = *addr;
+ voice->wave.block_header = block_header;
+
+ /* point to next header */
+ addr += 9;
+ voice->wave.addr = addr;
+
+ /* previous samples */
+ int smp2 = out [0];
+ int smp1 = out [1];
+
+ int offset = -BRR_BLOCK_SIZE * 4;
+
+#if !SPC_BRRCACHE
+ out [-(BRR_BLOCK_SIZE + 1)] = out [-1];
+
+ /* if next block has end flag set,
+ this block ends early (verified) */
+ if ( (block_header & 3) != 3 && (*addr & 3) == 1 )
+ {
+ /* arrange for last 9 samples to be skipped */
+ int const skip = 9;
+ out [skip - (BRR_BLOCK_SIZE + 1)] = out [-1];
+ out += (skip & 1);
+ voice->wave.position += skip * 0x1000;
+ offset = (-BRR_BLOCK_SIZE + (skip & ~1)) * 4;
+ addr -= skip / 2;
+ /* force sample to end on next decode */
+ voice->wave.block_header = 1;
+ }
+#endif /* !SPC_BRRCACHE */
+
+ int const filter = block_header & 0x0c;
+ int const scale = block_header >> 4;
+
+ if ( filter == 0x08 ) /* filter 2 (30-90% of the time) */
+ {
+ /* y[n] = x[n] + 61/32 * y[n-1] - 15/16 * y[n-2] */
+ do /* decode and filter 16 samples */
+ {
+ /* Get nybble, sign-extend, then scale
+ get byte, select which nybble, sign-extend, then shift
+ based on scaling. */
+ int delta = (int8_t)(addr [offset >> 3] << (offset & 4)) >> 4;
+ delta = (delta << scale) >> 1;
+
+ if (scale > 0xc)
+ delta = (delta >> 17) << 11;
+
+ out [offset >> 2] = smp2;
+
+ delta -= smp2 >> 1;
+ delta += smp2 >> 5;
+ delta += smp1;
+ delta += (-smp1 - (smp1 >> 1)) >> 5;
+
+ delta = CLAMP16( delta );
+ smp2 = smp1;
+ smp1 = (int16_t) (delta * 2); /* sign-extend */
+ }
+ while ( (offset += 4) != 0 );
+ }
+ else if ( filter == 0x04 ) /* filter 1 */
+ {
+ /* y[n] = x[n] + 15/16 * y[n-1] */
+ do /* decode and filter 16 samples */
+ {
+ /* Get nybble, sign-extend, then scale
+ get byte, select which nybble, sign-extend, then shift
+ based on scaling. */
+ int delta = (int8_t)(addr [offset >> 3] << (offset & 4)) >> 4;
+ delta = (delta << scale) >> 1;
+
+ if (scale > 0xc)
+ delta = (delta >> 17) << 11;
+
+ out [offset >> 2] = smp2;
+
+ delta += smp1 >> 1;
+ delta += (-smp1) >> 5;
+
+ delta = CLAMP16( delta );
+ smp2 = smp1;
+ smp1 = (int16_t) (delta * 2); /* sign-extend */
+ }
+ while ( (offset += 4) != 0 );
+ }
+ else if ( filter == 0x0c ) /* filter 3 */
+ {
+ /* y[n] = x[n] + 115/64 * y[n-1] - 13/16 * y[n-2] */
+ do /* decode and filter 16 samples */
+ {
+ /* Get nybble, sign-extend, then scale
+ get byte, select which nybble, sign-extend, then shift
+ based on scaling. */
+ int delta = (int8_t)(addr [offset >> 3] << (offset & 4)) >> 4;
+ delta = (delta << scale) >> 1;
+
+ if (scale > 0xc)
+ delta = (delta >> 17) << 11;
+
+ out [offset >> 2] = smp2;
+
+ delta -= smp2 >> 1;
+ delta += (smp2 + (smp2 >> 1)) >> 4;
+ delta += smp1;
+ delta += (-smp1 * 13) >> 7;
+
+ delta = CLAMP16( delta );
+ smp2 = smp1;
+ smp1 = (int16_t) (delta * 2); /* sign-extend */
+ }
+ while ( (offset += 4) != 0 );
+ }
+ else /* filter 0 */
+ {
+ /* y[n] = x[n] */
+ do /* decode and filter 16 samples */
+ {
+ /* Get nybble, sign-extend, then scale
+ get byte, select which nybble, sign-extend, then shift
+ based on scaling. */
+ int delta = (int8_t)(addr [offset >> 3] << (offset & 4)) >> 4;
+ delta = (delta << scale) >> 1;
+
+ if (scale > 0xc)
+ delta = (delta >> 17) << 11;
+
+ out [offset >> 2] = smp2;
+
+ smp2 = smp1;
+ smp1 = delta * 2;
+ }
+ while ( (offset += 4) != 0 );
+ }
#if SPC_BRRCACHE
-static void decode_brr( struct Spc_Dsp* this, unsigned start_addr,
- struct voice_t* voice,
- struct raw_voice_t const* const raw_voice ) ICODE_ATTR_SPC;
-static void decode_brr( struct Spc_Dsp* this, unsigned start_addr,
- struct voice_t* voice,
- struct raw_voice_t const* const raw_voice )
+ if ( !(block_header & 1) )
+ {
+ /* save to end of next block (for next call) */
+ out [BRR_BLOCK_SIZE ] = smp2;
+ out [BRR_BLOCK_SIZE + 1] = smp1;
+ }
+ else
+#endif /* SPC_BRRCACHE */
+ {
+ /* save to end of this block */
+ out [0] = smp2;
+ out [1] = smp1;
+ }
+}
+
+#if SPC_BRRCACHE
+static void NO_INLINE ICODE_ATTR_SPC
+brr_decode_cache( struct Spc_Dsp* this, struct src_dir const* sd,
+ unsigned start_addr, struct voice_t* voice,
+ struct raw_voice_t const* raw_voice )
{
- /* setup same variables as where decode_brr() is called from */
- #undef RAM
- #define RAM ram.ram
+ /* a little extra for samples that go past end */
+ static int16_t BRRcache [BRR_CACHE_SIZE] CACHEALIGN_ATTR;
+
+ DEBUGF( "decode at %08x (wave #%d)\n",
+ start_addr, raw_voice->waveform );
- struct src_dir const* const sd =
- &ram.sd[this->r.g.wave_page * 0x100/sizeof(struct src_dir)];
struct cache_entry_t* const wave_entry =
&this->wave_entry [raw_voice->waveform];
- /* the following block can be put in place of the call to
- decode_brr() below
- */
+ wave_entry->start_addr = start_addr;
+
+ uint8_t const* const loop_ptr =
+ ram.ram + letoh16( sd [raw_voice->waveform].loop );
+
+ int16_t* loop_start = NULL;
+
+ uint8_t const* addr = ram.ram + start_addr;
+
+ int16_t* out = BRRcache + start_addr * 2;
+ wave_entry->samples = out;
+
+ /* BRR filter uses previous samples */
+ out [BRR_BLOCK_SIZE + 1] = 0;
+ out [BRR_BLOCK_SIZE + 2] = 0;
+ *out++ = 0;
+
+ unsigned block_header;
+
+ do
+ {
+ if ( addr == loop_ptr )
+ {
+ loop_start = out;
+ DEBUGF( "loop at %08lx (wave #%d)\n",
+ (unsigned long)(addr - RAM), raw_voice->waveform );
+ }
+
+ /* output position - preincrement */
+ out += BRR_BLOCK_SIZE;
+
+ decode_brr_block( voice, addr, out );
+
+ block_header = voice->wave.block_header;
+ addr = voice->wave.addr;
+
+ /* if next block has end flag set, this block ends early */
+ /* (verified) */
+ if ( (block_header & 3) != 3 && (*addr & 3) == 1 )
+ {
+ /* skip last 9 samples */
+ DEBUGF( "block early end\n" );
+ out -= 9;
+ break;
+ }
+ }
+ while ( !(block_header & 1) && addr < RAM + 0x10000 );
+
+ wave_entry->end = (out - 1 - wave_entry->samples) << 12;
+ wave_entry->loop = 0;
+
+ if ( (block_header & 2) )
+ {
+ if ( loop_start )
+ {
+ wave_entry->loop = out - loop_start;
+ wave_entry->end += 0x3000;
+
+ out [2] = loop_start [2];
+ out [3] = loop_start [3];
+ out [4] = loop_start [4];
+ }
+ else
+ {
+ DEBUGF( "loop point outside initial wave\n" );
+ }
+ }
+
+ DEBUGF( "end at %08lx (wave #%d)\n",
+ (unsigned long)(addr - RAM), raw_voice->waveform );
+
+ /* add to cache */
+ this->wave_entry_old [this->oldsize++] = *wave_entry;
+}
+
+static inline void
+brr_key_on( struct Spc_Dsp* this, struct src_dir const* sd,
+ struct voice_t* voice, struct raw_voice_t const* raw_voice )
+{
+ unsigned start_addr = letoh16( sd [raw_voice->waveform].start );
+ struct cache_entry_t* const wave_entry =
+ &this->wave_entry [raw_voice->waveform];
+
+ /* predecode BRR if not already */
+ if ( wave_entry->start_addr != start_addr )
{
- DEBUGF( "decode at %08x (wave #%d)\n",
- start_addr, raw_voice->waveform );
-
/* see if in cache */
- int i;
- for ( i = 0; i < this->oldsize; i++ )
+ for ( int i = 0; i < this->oldsize; i++ )
{
struct cache_entry_t* e = &this->wave_entry_old [i];
+
if ( e->start_addr == start_addr )
{
DEBUGF( "found in wave_entry_old (oldsize=%d)\n",
this->oldsize );
*wave_entry = *e;
- goto wave_in_cache;
+ goto wave_in_cache; /* Wave in cache */
}
}
-
- wave_entry->start_addr = start_addr;
-
- uint8_t const* const loop_ptr =
- RAM + letoh16(sd[raw_voice->waveform].loop);
- short* loop_start = 0;
-
- short* out = BRRcache + start_addr * 2;
- wave_entry->samples = out;
- *out++ = 0;
- int smp1 = 0;
- int smp2 = 0;
-
- uint8_t const* addr = RAM + start_addr;
- int block_header;
- do
- {
- if ( addr == loop_ptr )
- {
- loop_start = out;
- DEBUGF( "loop at %08lx (wave #%d)\n",
- (unsigned long)(addr - RAM), raw_voice->waveform );
- }
-
- /* header */
- block_header = *addr;
- addr += 9;
- voice->addr = addr;
- int const filter = (block_header & 0x0C) - 0x08;
-
- /* scaling
- (invalid scaling gives -4096 for neg nybble, 0 for pos) */
- static unsigned char const right_shifts [16] = {
- 5, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 29, 29, 29,
- };
- static unsigned char const left_shifts [16] = {
- 0, 0, 1, 2, 3, 4, 5, 6, 7, 8, 9, 10, 11, 11, 11, 11
- };
- int const scale = block_header >> 4;
- int const right_shift = right_shifts [scale];
- int const left_shift = left_shifts [scale];
-
- /* output position */
- out += BRR_BLOCK_SIZE;
- int offset = -BRR_BLOCK_SIZE << 2;
-
- do /* decode and filter 16 samples */
- {
- /* Get nybble, sign-extend, then scale
- get byte, select which nybble, sign-extend, then shift based
- on scaling. also handles invalid scaling values. */
- int delta = (int) (int8_t) (addr [offset >> 3] << (offset & 4))
- >> right_shift << left_shift;
-
- out [offset >> 2] = smp2;
-
- if ( filter == 0 ) /* mode 0x08 (30-90% of the time) */
- {
- delta -= smp2 >> 1;
- delta += smp2 >> 5;
- smp2 = smp1;
- delta += smp1;
- delta += (-smp1 - (smp1 >> 1)) >> 5;
- }
- else
- {
- if ( filter == -4 ) /* mode 0x04 */
- {
- delta += smp1 >> 1;
- delta += (-smp1) >> 5;
- }
- else if ( filter > -4 ) /* mode 0x0C */
- {
- delta -= smp2 >> 1;
- delta += (smp2 + (smp2 >> 1)) >> 4;
- delta += smp1;
- delta += (-smp1 * 13) >> 7;
- }
- smp2 = smp1;
- }
-
- delta = CLAMP16( delta );
- smp1 = (int16_t) (delta * 2); /* sign-extend */
- }
- while ( (offset += 4) != 0 );
-
- /* if next block has end flag set, this block ends early */
- /* (verified) */
- if ( (block_header & 3) != 3 && (*addr & 3) == 1 )
- {
- /* skip last 9 samples */
- out -= 9;
- goto early_end;
- }
- }
- while ( !(block_header & 1) && addr < RAM + 0x10000 );
-
- out [0] = smp2;
- out [1] = smp1;
-
- early_end:
- wave_entry->end = (out - 1 - wave_entry->samples) << 12;
-
- wave_entry->loop = 0;
- if ( (block_header & 2) )
- {
- if ( loop_start )
- {
- int loop = out - loop_start;
- wave_entry->loop = loop;
- wave_entry->end += 0x3000;
- out [2] = loop_start [2];
- out [3] = loop_start [3];
- out [4] = loop_start [4];
- }
- else
- {
- DEBUGF( "loop point outside initial wave\n" );
- }
- }
-
- DEBUGF( "end at %08lx (wave #%d)\n",
- (unsigned long)(addr - RAM), raw_voice->waveform );
-
- /* add to cache */
- this->wave_entry_old [this->oldsize++] = *wave_entry;
-wave_in_cache:;
+
+ /* actually decode it */
+ brr_decode_cache( this, sd, start_addr, voice, raw_voice );
}
+
+wave_in_cache:
+ voice->wave.position = 3 * 0x1000 - 1; /* 0x2fff */
+ voice->wave.samples = wave_entry->samples;
+ voice->wave.end = wave_entry->end;
+ voice->wave.loop = wave_entry->loop;
+}
+
+static inline int brr_decode( struct src_dir const* sd, struct voice_t* voice,
+ struct raw_voice_t const* raw_voice )
+{
+ if ( voice->wave.position < voice->wave.end )
+ return 0;
+
+ long loop_len = voice->wave.loop << 12;
+
+ if ( !loop_len )
+ return 2;
+
+ voice->wave.position -= loop_len;
+ return 1;
+
+ (void)sd; (void)raw_voice;
}
-#endif
-static void key_on(struct Spc_Dsp* const this, struct voice_t* const voice,
- struct src_dir const* const sd,
- struct raw_voice_t const* const raw_voice,
- const int key_on_delay, const int vbit) ICODE_ATTR_SPC;
-static void key_on(struct Spc_Dsp* const this, struct voice_t* const voice,
- struct src_dir const* const sd,
- struct raw_voice_t const* const raw_voice,
- const int key_on_delay, const int vbit) {
+#else /* !SPC_BRRCACHE */
+
+static inline void
+brr_key_on( struct Spc_Dsp* this, struct src_dir const* sd,
+ struct voice_t* voice, struct raw_voice_t const* raw_voice )
+{
+ voice->wave.addr = ram.ram + letoh16( sd [raw_voice->waveform].start );
+ /* BRR filter uses previous samples */
+ voice->wave.samples [BRR_BLOCK_SIZE + 1] = 0;
+ voice->wave.samples [BRR_BLOCK_SIZE + 2] = 0;
+ /* force decode on next brr_decode call */
+ voice->wave.position = (BRR_BLOCK_SIZE + 3) * 0x1000 - 1; /* 0x12fff */
+ voice->wave.block_header = 0; /* "previous" BRR header */
+ (void)this;
+}
+
+static inline int brr_decode( struct src_dir const* sd, struct voice_t* voice,
+ struct raw_voice_t const* raw_voice )
+{
#undef RAM
+#if defined(CPU_ARM) && !SPC_BRRCACHE
+ uint8_t* const ram_ = ram.ram;
+ #define RAM ram_
+#else
#define RAM ram.ram
- int const env_rate_init = 0x7800;
+#endif
+
+ if ( voice->wave.position < BRR_BLOCK_SIZE * 0x1000 )
+ return 0;
+
+ voice->wave.position -= BRR_BLOCK_SIZE * 0x1000;
+
+ uint8_t const* addr = voice->wave.addr;
+
+ if ( addr >= RAM + 0x10000 )
+ addr -= 0x10000;
+
+ unsigned block_header = voice->wave.block_header;
+
+ /* action based on previous block's header */
+ int dec = 0;
+
+ if ( block_header & 1 )
+ {
+ addr = RAM + letoh16( sd [raw_voice->waveform].loop );
+ dec = 1;
+
+ if ( !(block_header & 2) ) /* 1% of the time */
+ {
+ /* first block was end block;
+ don't play anything (verified) */
+ return 2;
+ }
+ }
+
+ decode_brr_block( voice, addr, &voice->wave.samples [1 + BRR_BLOCK_SIZE] );
+
+ return dec;
+}
+#endif /* SPC_BRRCACHE */
+
+static void NO_INLINE ICODE_ATTR_SPC
+key_on( struct Spc_Dsp* const this, struct voice_t* const voice,
+ struct src_dir const* const sd,
+ struct raw_voice_t const* const raw_voice,
+ const int key_on_delay, const int vbit )
+{
voice->key_on_delay = key_on_delay;
+
if ( key_on_delay == 0 )
{
this->keys_down |= vbit;
- voice->envx = 0;
- voice->env_mode = state_attack;
- voice->env_timer = env_rate_init; /* TODO: inaccurate? */
- unsigned start_addr = letoh16(sd[raw_voice->waveform].start);
- #if !SPC_BRRCACHE
- {
- voice->addr = RAM + start_addr;
- /* BRR filter uses previous samples */
- voice->samples [BRR_BLOCK_SIZE + 1] = 0;
- voice->samples [BRR_BLOCK_SIZE + 2] = 0;
- /* decode three samples immediately */
- voice->position = (BRR_BLOCK_SIZE + 3) * 0x1000 - 1;
- voice->block_header = 0; /* "previous" BRR header */
- }
- #else
- {
- voice->position = 3 * 0x1000 - 1;
- struct cache_entry_t* const wave_entry =
- &this->wave_entry [raw_voice->waveform];
-
- /* predecode BRR if not already */
- if ( wave_entry->start_addr != start_addr )
- {
- /* the following line can be replaced by the indicated block
- in decode_brr() */
- decode_brr( this, start_addr, voice, raw_voice );
- }
-
- voice->samples = wave_entry->samples;
- voice->wave_end = wave_entry->end;
- voice->wave_loop = wave_entry->loop;
- }
- #endif
+ voice->envx = 0;
+ voice->env_mode = state_attack;
+ voice->env_timer = ENV_RATE_INIT; /* TODO: inaccurate? */
+ brr_key_on( this, sd, voice, raw_voice );
}
}
@@ -287,10 +540,8 @@ void DSP_run_( struct Spc_Dsp* this, long count, int32_t* out_buf )
#else
#define RAM ram.ram
#endif
-#if 0
EXIT_TIMER(cpu);
ENTER_TIMER(dsp);
-#endif
/* Here we check for keys on/off. Docs say that successive writes
to KON/KOF must be separated by at least 2 Ts periods or risk
@@ -327,98 +578,60 @@ void DSP_run_( struct Spc_Dsp* this, long count, int32_t* out_buf )
}
struct src_dir const* const sd =
- &ram.sd[this->r.g.wave_page * 0x100/sizeof(struct src_dir)];
+ &ram.sd [this->r.g.wave_page * 0x100/sizeof(struct src_dir)];
- #ifdef ROCKBOX_BIG_ENDIAN
- /* Convert endiannesses before entering loops - these
- get used alot */
- const uint32_t rates[VOICE_COUNT] =
- {
- GET_LE16A( this->r.voice[0].rate ) & 0x3FFF,
- GET_LE16A( this->r.voice[1].rate ) & 0x3FFF,
- GET_LE16A( this->r.voice[2].rate ) & 0x3FFF,
- GET_LE16A( this->r.voice[3].rate ) & 0x3FFF,
- GET_LE16A( this->r.voice[4].rate ) & 0x3FFF,
- GET_LE16A( this->r.voice[5].rate ) & 0x3FFF,
- GET_LE16A( this->r.voice[6].rate ) & 0x3FFF,
- GET_LE16A( this->r.voice[7].rate ) & 0x3FFF,
- };
- #define VOICE_RATE(x) *(x)
- #define IF_RBE(...) __VA_ARGS__
- #ifdef CPU_COLDFIRE
- /* Initialize mask register with the buffer address mask */
- asm volatile ("move.l %[m], %%mask" : : [m]"i"(FIR_BUF_MASK));
- const int echo_wrap = (this->r.g.echo_delay & 15) * 0x800;
- const int echo_start = this->r.g.echo_page * 0x100;
- #endif /* CPU_COLDFIRE */
- #else
- #define VOICE_RATE(x) (GET_LE16(raw_voice->rate) & 0x3FFF)
- #define IF_RBE(...)
- #endif /* ROCKBOX_BIG_ENDIAN */
-
#if !SPC_NOINTERP
int const slow_gaussian = (this->r.g.pitch_mods >> 1) |
- this->r.g.noise_enables;
+ this->r.g.noise_enables;
+#endif
+#if !SPC_NOECHO
+ int const echo_start = this->r.g.echo_page * 0x100;
+ int const echo_delay = (this->r.g.echo_delay & 15) * 0x800;
#endif
/* (g.flags & 0x40) ? 30 : 14 */
int const global_muting = ((this->r.g.flags & 0x40) >> 2) + 14 - 8;
int const global_vol_0 = this->r.g.volume_0;
int const global_vol_1 = this->r.g.volume_1;
- /* each rate divides exactly into 0x7800 without remainder */
- int const env_rate_init = 0x7800;
- static unsigned short const env_rates [0x20] ICONST_ATTR_SPC =
- {
- 0x0000, 0x000F, 0x0014, 0x0018, 0x001E, 0x0028, 0x0030, 0x003C,
- 0x0050, 0x0060, 0x0078, 0x00A0, 0x00C0, 0x00F0, 0x0140, 0x0180,
- 0x01E0, 0x0280, 0x0300, 0x03C0, 0x0500, 0x0600, 0x0780, 0x0A00,
- 0x0C00, 0x0F00, 0x1400, 0x1800, 0x1E00, 0x2800, 0x3C00, 0x7800
- };
-
do /* one pair of output samples per iteration */
{
/* Noise */
if ( this->r.g.noise_enables )
{
- if ( (this->noise_count -=
- env_rates [this->r.g.flags & 0x1F]) <= 0 )
+ this->noise_count -= env_rates [this->r.g.flags & 0x1F];
+
+ if ( this->noise_count <= 0 )
{
- this->noise_count = env_rate_init;
+ this->noise_count = ENV_RATE_INIT;
int feedback = (this->noise << 13) ^ (this->noise << 14);
this->noise = (feedback & 0x8000) ^ (this->noise >> 1 & ~1);
}
}
-#if !SPC_NOECHO
- int echo_0 = 0;
- int echo_1 = 0;
-#endif
+ #if !SPC_NOECHO
+ int echo_0 = 0, echo_1 = 0;
+ #endif /* !SPC_NOECHO */
long prev_outx = 0; /* TODO: correct value for first channel? */
- int chans_0 = 0;
- int chans_1 = 0;
+ int chans_0 = 0, chans_1 = 0;
+
/* TODO: put raw_voice pointer in voice_t? */
struct raw_voice_t * raw_voice = this->r.voice;
struct voice_t* voice = this->voice_state;
- int vbit = 1;
- IF_RBE( const uint32_t* vr = rates; )
- for ( ; vbit < 0x100; vbit <<= 1, ++voice, ++raw_voice IF_RBE( , ++vr ) )
+
+ for (int vbit = 1; vbit < 0x100; vbit <<= 1, ++voice, ++raw_voice )
{
/* pregen involves checking keyon, etc */
-#if 0
ENTER_TIMER(dsp_pregen);
-#endif
/* Key on events are delayed */
int key_on_delay = voice->key_on_delay;
if ( UNLIKELY ( --key_on_delay >= 0 ) ) /* <1% of the time */
- {
- key_on(this,voice,sd,raw_voice,key_on_delay,vbit);
- }
+ key_on( this, voice, sd, raw_voice, key_on_delay, vbit );
if ( !(this->keys_down & vbit) ) /* Silent channel */
{
- silent_chan:
+ silent_chan:
raw_voice->envx = 0;
raw_voice->outx = 0;
prev_outx = 0;
@@ -461,7 +674,7 @@ void DSP_run_( struct Spc_Dsp* this, long count, int32_t* out_buf )
voice->envx = envx;
/* TODO: should this be 8? */
raw_voice->envx = envx >> 4;
- env_timer = env_rate_init;
+ env_timer = ENV_RATE_INIT;
}
int sustain_level = adsr1 >> 5;
@@ -561,994 +774,131 @@ void DSP_run_( struct Spc_Dsp* this, long count, int32_t* out_buf )
}
}
init_env_timer:
- env_timer = env_rate_init;
+ env_timer = ENV_RATE_INIT;
write_env_timer:
voice->env_timer = env_timer;
env_end:;
}
-#if 0
+
EXIT_TIMER(dsp_pregen);
ENTER_TIMER(dsp_gen);
-#endif
- #if !SPC_BRRCACHE
- /* Decode BRR block */
- if ( voice->position >= BRR_BLOCK_SIZE * 0x1000 )
- {
- voice->position -= BRR_BLOCK_SIZE * 0x1000;
-
- uint8_t const* addr = voice->addr;
- if ( addr >= RAM + 0x10000 )
- addr -= 0x10000;
-
- /* action based on previous block's header */
- if ( voice->block_header & 1 )
- {
- addr = RAM + letoh16(sd[raw_voice->waveform].loop);
- this->r.g.wave_ended |= vbit;
- if ( !(voice->block_header & 2) ) /* 1% of the time */
- {
- /* first block was end block;
- don't play anything (verified) */
- /* bit was set, so this clears it */
- this->keys_down ^= vbit;
-
- /* since voice->envx is 0,
- samples and position don't matter */
- raw_voice->envx = 0;
- voice->envx = 0;
- goto skip_decode;
- }
- }
-
- /* header */
- int const block_header = *addr;
- addr += 9;
- voice->addr = addr;
- voice->block_header = block_header;
-
- /* previous samples */
- int smp2 = voice->samples [BRR_BLOCK_SIZE + 1];
- int smp1 = voice->samples [BRR_BLOCK_SIZE + 2];
- voice->samples [0] = voice->samples [BRR_BLOCK_SIZE];
-
- /* output position */
- short* out = voice->samples + (1 + BRR_BLOCK_SIZE);
- int offset = -BRR_BLOCK_SIZE << 2;
-
- /* if next block has end flag set,
- this block ends early (verified) */
- if ( (block_header & 3) != 3 && (*addr & 3) == 1 )
- {
- /* arrange for last 9 samples to be skipped */
- int const skip = 9;
- out += (skip & 1);
- voice->samples [skip] = voice->samples [BRR_BLOCK_SIZE];
- voice->position += skip * 0x1000;
- offset = (-BRR_BLOCK_SIZE + (skip & ~1)) << 2;
- addr -= skip / 2;
- /* force sample to end on next decode */
- voice->block_header = 1;
- }
-
- int const filter = block_header & 0x0c;
- int const scale = block_header >> 4;
-
- if ( filter == 0x08 ) /* filter 2 (30-90% of the time) */
- {
- /* y[n] = x[n] + 61/32 * y[n-1] - 15/16 * y[n-2] */
- do /* decode and filter 16 samples */
- {
- /* Get nybble, sign-extend, then scale
- get byte, select which nybble, sign-extend, then shift
- based on scaling. */
- int delta = (int8_t)(addr [offset >> 3] << (offset & 4)) >> 4;
- delta = (delta << scale) >> 1;
- if (scale > 0xc)
- delta = (delta >> 17) << 11;
-
- out [offset >> 2] = smp2;
-
- delta -= smp2 >> 1;
- delta += smp2 >> 5;
- delta += smp1;
- delta += (-smp1 - (smp1 >> 1)) >> 5;
-
- delta = CLAMP16( delta );
- smp2 = smp1;
- smp1 = (int16_t) (delta * 2); /* sign-extend */
- }
- while ( (offset += 4) != 0 );
- }
- else if ( filter == 0x04 ) /* filter 1 */
- {
- /* y[n] = x[n] + 15/16 * y[n-1] */
- do /* decode and filter 16 samples */
- {
- /* Get nybble, sign-extend, then scale
- get byte, select which nybble, sign-extend, then shift
- based on scaling. */
- int delta = (int8_t)(addr [offset >> 3] << (offset & 4)) >> 4;
- delta = (delta << scale) >> 1;
-
- if (scale > 0xc)
- delta = (delta >> 17) << 11;
-
- out [offset >> 2] = smp2;
-
- delta += smp1 >> 1;
- delta += (-smp1) >> 5;
-
- delta = CLAMP16( delta );
- smp2 = smp1;
- smp1 = (int16_t) (delta * 2); /* sign-extend */
- }
- while ( (offset += 4) != 0 );
- }
- else if ( filter == 0x0c ) /* filter 3 */
- {
- /* y[n] = x[n] + 115/64 * y[n-1] - 13/16 * y[n-2] */
- do /* decode and filter 16 samples */
- {
- /* Get nybble, sign-extend, then scale
- get byte, select which nybble, sign-extend, then shift
- based on scaling. */
- int delta = (int8_t)(addr [offset >> 3] << (offset & 4)) >> 4;
- delta = (delta << scale) >> 1;
-
- if (scale > 0xc)
- delta = (delta >> 17) << 11;
-
- out [offset >> 2] = smp2;
-
- delta -= smp2 >> 1;
- delta += (smp2 + (smp2 >> 1)) >> 4;
- delta += smp1;
- delta += (-smp1 * 13) >> 7;
-
- delta = CLAMP16( delta );
- smp2 = smp1;
- smp1 = (int16_t) (delta * 2); /* sign-extend */
- }
- while ( (offset += 4) != 0 );
- }
- else /* filter 0 */
- {
- /* y[n] = x[n] */
- do /* decode and filter 16 samples */
- {
- /* Get nybble, sign-extend, then scale
- get byte, select which nybble, sign-extend, then shift
- based on scaling. */
- int delta = (int8_t)(addr [offset >> 3] << (offset & 4)) >> 4;
- delta = (delta << scale) >> 1;
-
- if (scale > 0xc)
- delta = (delta >> 17) << 11;
-
- out [offset >> 2] = smp2;
-
- smp2 = smp1;
- smp1 = delta * 2;
- }
- while ( (offset += 4) != 0 );
- }
+ switch ( brr_decode( sd, voice, raw_voice ) )
+ {
+ case 2:
+ /* bit was set, so this clears it */
+ this->keys_down ^= vbit;
- out [0] = smp2;
- out [1] = smp1;
-
- skip_decode:;
+ /* since voice->envx is 0,
+ samples and position don't matter */
+ raw_voice->envx = 0;
+ voice->envx = 0;
+ case 1:
+ this->r.g.wave_ended |= vbit;
}
- #endif /* !SPC_BRRCACHE */
+
/* Get rate (with possible modulation) */
- int rate = VOICE_RATE(vr);
+ int rate = voice->rate;
if ( this->r.g.pitch_mods & vbit )
rate = (rate * (prev_outx + 32768)) >> 15;
+ uint32_t position = voice->wave.position;
+ voice->wave.position += rate;
+
+ int output;
+ int amp_0, amp_1;
+
#if !SPC_NOINTERP
- /* Interleved gauss table (to improve cache coherency). */
- /* gauss [i * 2 + j] = normal_gauss [(1 - j) * 256 + i] */
- static short const gauss [512] ICONST_ATTR_SPC MEM_ALIGN_ATTR =
- {
-370,1305, 366,1305, 362,1304, 358,1304, 354,1304, 351,1304, 347,1304, 343,1303,
-339,1303, 336,1303, 332,1302, 328,1302, 325,1301, 321,1300, 318,1300, 314,1299,
-311,1298, 307,1297, 304,1297, 300,1296, 297,1295, 293,1294, 290,1293, 286,1292,
-283,1291, 280,1290, 276,1288, 273,1287, 270,1286, 267,1284, 263,1283, 260,1282,
-257,1280, 254,1279, 251,1277, 248,1275, 245,1274, 242,1272, 239,1270, 236,1269,
-233,1267, 230,1265, 227,1263, 224,1261, 221,1259, 218,1257, 215,1255, 212,1253,
-210,1251, 207,1248, 204,1246, 201,1244, 199,1241, 196,1239, 193,1237, 191,1234,
-188,1232, 186,1229, 183,1227, 180,1224, 178,1221, 175,1219, 173,1216, 171,1213,
-168,1210, 166,1207, 163,1205, 161,1202, 159,1199, 156,1196, 154,1193, 152,1190,
-150,1186, 147,1183, 145,1180, 143,1177, 141,1174, 139,1170, 137,1167, 134,1164,
-132,1160, 130,1157, 128,1153, 126,1150, 124,1146, 122,1143, 120,1139, 118,1136,
-117,1132, 115,1128, 113,1125, 111,1121, 109,1117, 107,1113, 106,1109, 104,1106,
-102,1102, 100,1098, 99,1094, 97,1090, 95,1086, 94,1082, 92,1078, 90,1074,
- 89,1070, 87,1066, 86,1061, 84,1057, 83,1053, 81,1049, 80,1045, 78,1040,
- 77,1036, 76,1032, 74,1027, 73,1023, 71,1019, 70,1014, 69,1010, 67,1005,
- 66,1001, 65, 997, 64, 992, 62, 988, 61, 983, 60, 978, 59, 974, 58, 969,
- 56, 965, 55, 960, 54, 955, 53, 951, 52, 946, 51, 941, 50, 937, 49, 932,
- 48, 927, 47, 923, 46, 918, 45, 913, 44, 908, 43, 904, 42, 899, 41, 894,
- 40, 889, 39, 884, 38, 880, 37, 875, 36, 870, 36, 865, 35, 860, 34, 855,
- 33, 851, 32, 846, 32, 841, 31, 836, 30, 831, 29, 826, 29, 821, 28, 816,
- 27, 811, 27, 806, 26, 802, 25, 797, 24, 792, 24, 787, 23, 782, 23, 777,
- 22, 772, 21, 767, 21, 762, 20, 757, 20, 752, 19, 747, 19, 742, 18, 737,
- 17, 732, 17, 728, 16, 723, 16, 718, 15, 713, 15, 708, 15, 703, 14, 698,
- 14, 693, 13, 688, 13, 683, 12, 678, 12, 674, 11, 669, 11, 664, 11, 659,
- 10, 654, 10, 649, 10, 644, 9, 640, 9, 635, 9, 630, 8, 625, 8, 620,
- 8, 615, 7, 611, 7, 606, 7, 601, 6, 596, 6, 592, 6, 587, 6, 582,
- 5, 577, 5, 573, 5, 568, 5, 563, 4, 559, 4, 554, 4, 550, 4, 545,
- 4, 540, 3, 536, 3, 531, 3, 527, 3, 522, 3, 517, 2, 513, 2, 508,
- 2, 504, 2, 499, 2, 495, 2, 491, 2, 486, 1, 482, 1, 477, 1, 473,
- 1, 469, 1, 464, 1, 460, 1, 456, 1, 451, 1, 447, 1, 443, 1, 439,
- 0, 434, 0, 430, 0, 426, 0, 422, 0, 418, 0, 414, 0, 410, 0, 405,
- 0, 401, 0, 397, 0, 393, 0, 389, 0, 385, 0, 381, 0, 378, 0, 374,
- };
/* Gaussian interpolation using most recent 4 samples */
- long position = voice->position;
- voice->position += rate;
- short const* interp = voice->samples + (position >> 12);
- int offset = position >> 4 & 0xFF;
-
+
/* Only left half of gaussian kernel is in table, so we must mirror
for right half */
- short const* fwd = gauss + offset * 2;
- short const* rev = gauss + 510 - offset * 2;
+ int offset = ( position >> 4 ) & 0xFF;
+ int16_t const* fwd = gauss_table + offset * 2;
+ int16_t const* rev = gauss_table + 510 - offset * 2;
/* Use faster gaussian interpolation when exact result isn't needed
by pitch modulator of next channel */
- int amp_0, amp_1; /* Also serve as temps _0, and _1 */
if ( LIKELY ( !(slow_gaussian & vbit) ) ) /* 99% of the time */
{
/* Main optimization is lack of clamping. Not a problem since
output never goes more than +/- 16 outside 16-bit range and
things are clamped later anyway. Other optimization is to
preserve fractional accuracy, eliminating several masks. */
- #if defined (CPU_ARM)
- int output;
- int _2, _3; /* All-purpose temps */
- /* Multiple ASM blocks keep regs free and reduce result
- * latency issues. */
- #if ARM_ARCH >= 6
- /* Interpolate */
- asm volatile (
- "ldr %[_0], [%[interp]] \r\n" /* _0=i0i1 */
- "ldr %[_2], [%[fwd]] \r\n" /* _2=f0f1 */
- "ldr %[_1], [%[interp], #4] \r\n" /* _1=i2i3 */
- "ldr %[_3], [%[rev]] \r\n" /* _3=r0r1 */
- "smuad %[out], %[_0], %[_2] \r\n" /* out=f0*i0 + f1*i1 */
- "smladx %[out], %[_1], %[_3], %[out] \r\n" /* out+=r1*i2 + r0*i3 */
- : [out]"=r"(output),
- [_0]"=&r"(amp_0), [_1]"=&r"(amp_1),
- [_2]"=&r"(_2), [_3]"=r"(_3)
- : [fwd]"r"(fwd), [rev]"r"(rev),
- [interp]"r"(interp));
- /* Apply voice envelope */
- asm volatile (
- "mov %[_2], %[out], asr #(11-5) \r\n" /* To do >> 16 later */
- "mul %[out], %[_2], %[envx] \r\n" /* and avoid exp. shift */
- : [out]"+r"(output), [_2]"=&r"(_2)
- : [envx]"r"((int)voice->envx));
- /* Apply left and right volume */
- asm volatile (
- "smulwb %[amp_0], %[out], %[vvol_0] \r\n" /* (32x16->48)[47:16]->[31:0] */
- "smulwb %[amp_1], %[out], %[vvol_1] \r\n"
- : [out]"+r"(output),
- [amp_0]"=&r"(amp_0), [amp_1]"=r"(amp_1)
- : [vvol_0]"r"(voice->volume[0]),
- [vvol_1]"r"(voice->volume[1]));
-
- raw_voice->outx = output >> (8+5); /* 'output' still 5 bits too big */
- #else /* ARM_ARCH < 6 */
- /* Perform gaussian interpolation on four samples */
- asm volatile (
- "ldrsh %[_0], [%[interp]] \r\n"
- "ldrsh %[_2], [%[fwd]] \r\n"
- "ldrsh %[_1], [%[interp], #2] \r\n"
- "ldrsh %[_3], [%[fwd], #2] \r\n"
- "mul %[out], %[_0], %[_2] \r\n" /* out= fwd[0]*interp[0] */
- "ldrsh %[_0], [%[interp], #4] \r\n"
- "ldrsh %[_2], [%[rev], #2] \r\n"
- "mla %[out], %[_1], %[_3], %[out] \r\n" /* out+=fwd[1]*interp[1] */
- "ldrsh %[_1], [%[interp], #6] \r\n"
- "ldrsh %[_3], [%[rev]] \r\n"
- "mla %[out], %[_0], %[_2], %[out] \r\n" /* out+=rev[1]*interp[2] */
- "mla %[out], %[_1], %[_3], %[out] \r\n" /* out+=rev[0]*interp[3] */
- : [out]"=&r"(output),
- [_0]"=&r"(amp_0), [_1]"=&r"(amp_1),
- [_2]"=&r"(_2), [_3]"=&r"(_3)
- : [fwd]"r"(fwd), [rev]"r"(rev),
- [interp]"r"(interp));
- /* Apply voice envelope */
- asm volatile (
- "mov %[_2], %[out], asr #11 \r\n"
- "mul %[out], %[_2], %[envx] \r\n"
- : [out]"+r"(output), [_2]"=&r"(_2)
- : [envx]"r"((int)voice->envx));
- /* Reduce and apply left and right volume */
- asm volatile (
- "mov %[out], %[out], asr #11 \r\n"
- "mul %[amp_0], %[out], %[vvol_0] \r\n"
- "mul %[amp_1], %[out], %[vvol_1] \r\n"
- : [out]"+r"(output),
- [amp_0]"=&r"(amp_0), [amp_1]"=r"(amp_1)
- : [vvol_0]"r"((int)voice->volume[0]),
- [vvol_1]"r"((int)voice->volume[1]));
-
- raw_voice->outx = output >> 8;
- #endif /* ARM_ARCH */
- #else /* Unoptimized CPU */
- int output = (((fwd [0] * interp [0] +
- fwd [1] * interp [1] +
- rev [1] * interp [2] +
- rev [0] * interp [3] ) >> 11) * voice->envx) >> 11;
-
- /* duplicated here to give compiler more to run in parallel */
- amp_0 = voice->volume [0] * output;
- amp_1 = voice->volume [1] * output;
-
- raw_voice->outx = output >> 8;
- #endif /* CPU_* */
+ output = gaussian_fast_interp( voice->wave.samples, position,
+ fwd, rev );
+ output = gaussian_fast_amp( voice, output, &amp_0, &amp_1 );
}
else /* slow gaussian */
+ #endif /* !SPC_NOINTERP (else two-point linear interpolation) */
{
- #if defined(CPU_ARM)
- #if ARM_ARCH >= 6
- int output = *(int16_t*) &this->noise;
-
- if ( !(this->r.g.noise_enables & vbit) )
- {
- /* Interpolate */
- int _2, _3;
- asm volatile (
- /* NOTE: often-unaligned accesses */
- "ldr %[_0], [%[interp]] \r\n" /* _0=i0i1 */
- "ldr %[_2], [%[fwd]] \r\n" /* _2=f0f1 */
- "ldr %[_1], [%[interp], #4] \r\n" /* _1=i2i3 */
- "ldr %[_3], [%[rev]] \r\n" /* _3=f2f3 */
- "smulbb %[out], %[_0], %[_2] \r\n" /* out=f0*i0 */
- "smultt %[_0], %[_0], %[_2] \r\n" /* _0=f1*i1 */
- "smulbt %[_2], %[_1], %[_3] \r\n" /* _2=r1*i2 */
- "smultb %[_3], %[_1], %[_3] \r\n" /* _3=r0*i3 */
- : [out]"=r"(output),
- [_0]"=&r"(amp_0), [_1]"=&r"(amp_1),
- [_2]"=&r"(_2), [_3]"=r"(_3)
- : [fwd]"r"(fwd), [rev]"r"(rev),
- [interp]"r"(interp));
- asm volatile (
- "mov %[out], %[out], asr#12 \r\n"
- "add %[_0], %[out], %[_0], asr #12 \r\n"
- "add %[_2], %[_0], %[_2], asr #12 \r\n"
- "pkhbt %[_0], %[_2], %[_3], asl #4 \r\n" /* _3[31:16], _2[15:0] */
- "sadd16 %[_0], %[_0], %[_0] \r\n" /* _3[31:16]*2, _2[15:0]*2 */
- "qsubaddx %[out], %[_0], %[_0] \r\n" /* out[15:0]=
- * sat16(_3[31:16]+_2[15:0]) */
- : [out]"+r"(output),
- [_0]"+r"(amp_0), [_2]"+r"(_2), [_3]"+r"(_3));
- }
- /* Apply voice envelope */
- asm volatile (
- "smulbb %[out], %[out], %[envx] \r\n"
- : [out]"+r"(output)
- : [envx]"r"(voice->envx));
- /* Reduce and apply left and right volume */
- asm volatile (
- "mov %[out], %[out], asr #11 \r\n"
- "bic %[out], %[out], #0x1 \r\n"
- "mul %[amp_0], %[out], %[vvol_0] \r\n"
- "mul %[amp_1], %[out], %[vvol_1] \r\n"
- : [out]"+r"(output),
- [amp_0]"=&r"(amp_0), [amp_1]"=r"(amp_1)
- : [vvol_0]"r"((int)voice->volume[0]),
- [vvol_1]"r"((int)voice->volume[1]));
-
- prev_outx = output;
- raw_voice->outx = output >> 8;
- #else /* ARM_ARCH < 6 */
- int output = *(int16_t*) &this->noise;
-
- if ( !(this->r.g.noise_enables & vbit) )
- {
- /* Interpolate */
- int _2, _3;
- asm volatile (
- "ldrsh %[_0], [%[interp]] \r\n"
- "ldrsh %[_2], [%[fwd]] \r\n"
- "ldrsh %[_1], [%[interp], #2] \r\n"
- "ldrsh %[_3], [%[fwd], #2] \r\n"
- "mul %[out], %[_2], %[_0] \r\n" /* fwd[0]*interp[0] */
- "ldrsh %[_2], [%[rev], #2] \r\n"
- "mul %[_0], %[_3], %[_1] \r\n" /* fwd[1]*interp[1] */
- "ldrsh %[_1], [%[interp], #4] \r\n"
- "mov %[out], %[out], asr #12 \r\n"
- "ldrsh %[_3], [%[rev]] \r\n"
- "mul %[_2], %[_1], %[_2] \r\n" /* rev[1]*interp[2] */
- "ldrsh %[_1], [%[interp], #6] \r\n"
- "add %[_0], %[out], %[_0], asr #12 \r\n"
- "mul %[_3], %[_1], %[_3] \r\n" /* rev[0]*interp[3] */
- "add %[_2], %[_0], %[_2], asr #12 \r\n"
- "mov %[_2], %[_2], lsl #17 \r\n"
- "mov %[_3], %[_3], asr #12 \r\n"
- "mov %[_3], %[_3], asl #1 \r\n"
- "add %[out], %[_3], %[_2], asr #16 \r\n"
- : [out]"=&r"(output),
- [_0]"=&r"(amp_0), [_1]"=&r"(amp_1),
- [_2]"=&r"(_2), [_3]"=&r"(_3)
- : [fwd]"r"(fwd), [rev]"r"(rev),
- [interp]"r"(interp));
-
- output = CLAMP16(output);
- }
- /* Apply voice envelope */
- asm volatile (
- "mul %[_0], %[out], %[envx] \r\n"
- : [_0]"=r"(amp_0)
- : [out]"r"(output), [envx]"r"((int)voice->envx));
- /* Reduce and apply left and right volume */
- asm volatile (
- "mov %[out], %[amp_0], asr #11 \r\n" /* amp_0 = _0 */
- "bic %[out], %[out], #0x1 \r\n"
- "mul %[amp_0], %[out], %[vvol_0] \r\n"
- "mul %[amp_1], %[out], %[vvol_1] \r\n"
- : [out]"+r"(output),
- [amp_0]"+r"(amp_0), [amp_1]"=r"(amp_1)
- : [vvol_0]"r"((int)voice->volume[0]),
- [vvol_1]"r"((int)voice->volume[1]));
-
- prev_outx = output;
- raw_voice->outx = output >> 8;
- #endif /* ARM_ARCH >= 6 */
- #else /* Unoptimized CPU */
- int output = *(int16_t*) &this->noise;
+ output = *(int16_t *)&this->noise;
if ( !(this->r.g.noise_enables & vbit) )
- {
- output = (fwd [0] * interp [0]) & ~0xFFF;
- output = (output + fwd [1] * interp [1]) & ~0xFFF;
- output = (output + rev [1] * interp [2]) >> 12;
- output = (int16_t) (output * 2);
- output += ((rev [0] * interp [3]) >> 12) * 2;
- output = CLAMP16( output );
- }
- output = (output * voice->envx) >> 11 & ~1;
-
- /* duplicated here to give compiler more to run in parallel */
- amp_0 = voice->volume [0] * output;
- amp_1 = voice->volume [1] * output;
-
- prev_outx = output;
- raw_voice->outx = output >> 8;
- #endif /* CPU_* */
- }
- #else /* SPCNOINTERP */
- /* two-point linear interpolation */
- #ifdef CPU_COLDFIRE
- int amp_0 = (int16_t)this->noise;
- int amp_1;
-
- if ( (this->r.g.noise_enables & vbit) == 0 )
- {
- uint32_t f = voice->position;
- int32_t y0;
-
- /**
- * Formula (fastest found so far of MANY):
- * output = y0 + f*y1 - f*y0
- */
- asm volatile (
- /* separate fractional and whole parts */
- "move.l %[f], %[y1] \r\n"
- "and.l #0xfff, %[f] \r\n"
- "lsr.l %[sh], %[y1] \r\n"
- /* load samples y0 (upper) & y1 (lower) */
- "move.l 2(%[s], %[y1].l*2), %[y1] \r\n"
- /* %acc0 = f*y1 */
- "mac.w %[f]l, %[y1]l, %%acc0 \r\n"
- /* %acc0 -= f*y0 */
- "msac.w %[f]l, %[y1]u, %%acc0 \r\n"
- /* separate out y0 and sign extend */
- "swap %[y1] \r\n"
- "movea.w %[y1], %[y0] \r\n"
- /* fetch result, scale down and add y0 */
- "movclr.l %%acc0, %[y1] \r\n"
- /* output = y0 + (result >> 12) */
- "asr.l %[sh], %[y1] \r\n"
- "add.l %[y0], %[y1] \r\n"
- : [f]"+d"(f), [y0]"=&a"(y0), [y1]"=&d"(amp_0)
- : [s]"a"(voice->samples), [sh]"d"(12));
- }
+ output = interp( voice->wave.samples, position, fwd, rev );
- /* apply voice envelope to output */
- asm volatile (
- "mac.w %[out]l, %[envx]l, %%acc0 \r\n"
- :
- : [out]"r"(amp_0), [envx]"r"(voice->envx));
-
- /* advance voice position */
- voice->position += rate;
-
- /* fetch output, scale and apply left and right
- voice volume */
- asm volatile (
- "movclr.l %%acc0, %[out] \r\n"
- "asr.l %[sh], %[out] \r\n"
- "mac.l %[vvol_0], %[out], %%acc0 \r\n"
- "mac.l %[vvol_1], %[out], %%acc1 \r\n"
- : [out]"=&d"(amp_0)
- : [vvol_0]"r"((int)voice->volume[0]),
- [vvol_1]"r"((int)voice->volume[1]),
- [sh]"d"(11));
-
- /* save this output into previous, scale and save in
- output register */
- prev_outx = amp_0;
- raw_voice->outx = amp_0 >> 8;
-
- /* fetch final voice output */
- asm volatile (
- "movclr.l %%acc0, %[amp_0] \r\n"
- "movclr.l %%acc1, %[amp_1] \r\n"
- : [amp_0]"=r"(amp_0), [amp_1]"=r"(amp_1));
- #elif defined (CPU_ARM)
- int amp_0, amp_1;
-
- if ( (this->r.g.noise_enables & vbit) != 0 )
- {
- amp_0 = *(int16_t *)&this->noise;
- }
- else
- {
- uint32_t f = voice->position;
- amp_0 = (uint32_t)voice->samples;
-
- asm volatile(
- "mov %[y1], %[f], lsr #12 \r\n"
- "eor %[f], %[f], %[y1], lsl #12 \r\n"
- "add %[y1], %[y0], %[y1], lsl #1 \r\n"
- "ldrsh %[y0], [%[y1], #2] \r\n"
- "ldrsh %[y1], [%[y1], #4] \r\n"
- "sub %[y1], %[y1], %[y0] \r\n"
- "mul %[f], %[y1], %[f] \r\n"
- "add %[y0], %[y0], %[f], asr #12 \r\n"
- : [f]"+r"(f), [y0]"+r"(amp_0), [y1]"=&r"(amp_1));
- }
-
- voice->position += rate;
-
- asm volatile(
- "mul %[amp_1], %[amp_0], %[envx] \r\n"
- "mov %[amp_0], %[amp_1], asr #11 \r\n"
- "mov %[amp_1], %[amp_0], asr #8 \r\n"
- : [amp_0]"+r"(amp_0), [amp_1]"=r"(amp_1)
- : [envx]"r"(voice->envx));
-
- prev_outx = amp_0;
- raw_voice->outx = (int8_t)amp_1;
-
- asm volatile(
- "mul %[amp_1], %[amp_0], %[vol_1] \r\n"
- "mul %[amp_0], %[vol_0], %[amp_0] \r\n"
- : [amp_0]"+r"(amp_0), [amp_1]"=&r"(amp_1)
- : [vol_0]"r"((int)voice->volume[0]),
- [vol_1]"r"((int)voice->volume[1]));
- #else /* Unoptimized CPU */
- int output;
-
- if ( (this->r.g.noise_enables & vbit) == 0 )
- {
- int const fraction = voice->position & 0xfff;
- short const* const pos = (voice->samples + (voice->position >> 12)) + 1;
- output = pos[0] + ((fraction * (pos[1] - pos[0])) >> 12);
- } else {
- output = *(int16_t *)&this->noise;
+ /* Apply envelope and volume */
+ output = apply_amp( voice, output, &amp_0, &amp_1 );
}
- voice->position += rate;
-
- output = (output * voice->envx) >> 11;
-
- /* duplicated here to give compiler more to run in parallel */
- int amp_0 = voice->volume [0] * output;
- int amp_1 = voice->volume [1] * output;
-
prev_outx = output;
- raw_voice->outx = (int8_t) (output >> 8);
- #endif /* CPU_* */
- #endif /* SPCNOINTERP */
+ raw_voice->outx = output >> 8;
- #if SPC_BRRCACHE
- if ( voice->position >= voice->wave_end )
- {
- long loop_len = voice->wave_loop << 12;
- voice->position -= loop_len;
- this->r.g.wave_ended |= vbit;
- if ( !loop_len )
- {
- this->keys_down ^= vbit;
- raw_voice->envx = 0;
- voice->envx = 0;
- }
- }
- #endif
-#if 0
EXIT_TIMER(dsp_gen);
ENTER_TIMER(dsp_mix);
-#endif
+
chans_0 += amp_0;
chans_1 += amp_1;
- #if !SPC_NOECHO
- if ( this->r.g.echo_ons & vbit )
- {
- echo_0 += amp_0;
- echo_1 += amp_1;
- }
- #endif
-#if 0
+ #if !SPC_NOECHO
+ if ( this->r.g.echo_ons & vbit )
+ {
+ echo_0 += amp_0;
+ echo_1 += amp_1;
+ }
+ #endif /* !SPC_NOECHO */
+
EXIT_TIMER(dsp_mix);
-#endif
}
/* end of voice loop */
+ /* Generate output */
+ int amp_0, amp_1;
#if !SPC_NOECHO
- #ifdef CPU_COLDFIRE
/* Read feedback from echo buffer */
int echo_pos = this->echo_pos;
uint8_t* const echo_ptr = RAM + ((echo_start + echo_pos) & 0xFFFF);
- echo_pos += 4;
- if ( echo_pos >= echo_wrap )
- echo_pos = 0;
- this->echo_pos = echo_pos;
- int fb = swap_odd_even32(*(int32_t *)echo_ptr);
- int out_0, out_1;
-
- /* Keep last 8 samples */
- *this->last_fir_ptr = fb;
- this->last_fir_ptr = this->fir_ptr;
-
- /* Apply echo FIR filter to output samples read from echo buffer -
- circular buffer is hardware incremented and masked; FIR
- coefficients and buffer history are loaded in parallel with
- multiply accumulate operations. Shift left by one here and once
- again when calculating feedback to have sample values justified
- to bit 31 in the output to ease endian swap, interleaving and
- clamping before placing result in the program's echo buffer. */
- int _0, _1, _2;
- asm volatile (
- "move.l (%[fir_c]) , %[_2] \r\n"
- "mac.w %[fb]u, %[_2]u, <<, (%[fir_p])+&, %[_0], %%acc0 \r\n"
- "mac.w %[fb]l, %[_2]u, <<, (%[fir_p])& , %[_1], %%acc1 \r\n"
- "mac.w %[_0]u, %[_2]l, << , %%acc0 \r\n"
- "mac.w %[_0]l, %[_2]l, <<, 4(%[fir_c]) , %[_2], %%acc1 \r\n"
- "mac.w %[_1]u, %[_2]u, <<, 4(%[fir_p])& , %[_0], %%acc0 \r\n"
- "mac.w %[_1]l, %[_2]u, <<, 8(%[fir_p])& , %[_1], %%acc1 \r\n"
- "mac.w %[_0]u, %[_2]l, << , %%acc0 \r\n"
- "mac.w %[_0]l, %[_2]l, <<, 8(%[fir_c]) , %[_2], %%acc1 \r\n"
- "mac.w %[_1]u, %[_2]u, <<, 12(%[fir_p])& , %[_0], %%acc0 \r\n"
- "mac.w %[_1]l, %[_2]u, <<, 16(%[fir_p])& , %[_1], %%acc1 \r\n"
- "mac.w %[_0]u, %[_2]l, << , %%acc0 \r\n"
- "mac.w %[_0]l, %[_2]l, <<, 12(%[fir_c]) , %[_2], %%acc1 \r\n"
- "mac.w %[_1]u, %[_2]u, <<, 20(%[fir_p])& , %[_0], %%acc0 \r\n"
- "mac.w %[_1]l, %[_2]u, << , %%acc1 \r\n"
- "mac.w %[_0]u, %[_2]l, << , %%acc0 \r\n"
- "mac.w %[_0]l, %[_2]l, << , %%acc1 \r\n"
- : [_0]"=&r"(_0), [_1]"=&r"(_1), [_2]"=&r"(_2),
- [fir_p]"+a"(this->fir_ptr)
- : [fir_c]"a"(this->fir_coeff), [fb]"r"(fb)
- );
-
- /* Generate output */
- asm volatile (
- /* fetch filter results _after_ gcc loads asm
- block parameters to eliminate emac stalls */
- "movclr.l %%acc0, %[out_0] \r\n"
- "movclr.l %%acc1, %[out_1] \r\n"
- /* apply global volume */
- "mac.l %[chans_0], %[gv_0] , %%acc2 \r\n"
- "mac.l %[chans_1], %[gv_1] , %%acc3 \r\n"
- /* apply echo volume and add to final output */
- "mac.l %[ev_0], %[out_0], >>, %%acc2 \r\n"
- "mac.l %[ev_1], %[out_1], >>, %%acc3 \r\n"
- : [out_0]"=&r"(out_0), [out_1]"=&r"(out_1)
- : [chans_0]"r"(chans_0), [gv_0]"r"(global_vol_0),
- [ev_0]"r"((int)this->r.g.echo_volume_0),
- [chans_1]"r"(chans_1), [gv_1]"r"(global_vol_1),
- [ev_1]"r"((int)this->r.g.echo_volume_1)
- );
-
- /* Feedback into echo buffer */
- if ( !(this->r.g.flags & 0x20) )
- {
- int sh = 1 << 9;
-
- asm volatile (
- /* scale echo voices; saturate if overflow */
- "mac.l %[sh], %[e1] , %%acc1 \r\n"
- "mac.l %[sh], %[e0] , %%acc0 \r\n"
- /* add scaled output from FIR filter */
- "mac.l %[out_1], %[ef], <<, %%acc1 \r\n"
- "mac.l %[out_0], %[ef], <<, %%acc0 \r\n"
- /* swap and fetch feedback results - simply
- swap_odd_even32 mixed in between macs and
- movclrs to mitigate stall issues */
- "move.l #0x00ff00ff, %[sh] \r\n"
- "movclr.l %%acc1, %[e1] \r\n"
- "swap %[e1] \r\n"
- "movclr.l %%acc0, %[e0] \r\n"
- "move.w %[e1], %[e0] \r\n"
- "and.l %[e0], %[sh] \r\n"
- "eor.l %[sh], %[e0] \r\n"
- "lsl.l #8, %[sh] \r\n"
- "lsr.l #8, %[e0] \r\n"
- "or.l %[sh], %[e0] \r\n"
- /* save final feedback into echo buffer */
- "move.l %[e0], (%[echo_ptr]) \r\n"
- : [e0]"+d"(echo_0), [e1]"+d"(echo_1), [sh]"+d"(sh)
- : [out_0]"r"(out_0), [out_1]"r"(out_1),
- [ef]"r"((int)this->r.g.echo_feedback),
- [echo_ptr]"a"((int32_t *)echo_ptr)
- );
- }
- /* Output final samples */
- asm volatile (
- /* fetch output saved in %acc2 and %acc3 */
- "movclr.l %%acc2, %[out_0] \r\n"
- "movclr.l %%acc3, %[out_1] \r\n"
- /* scale right by global_muting shift */
- "asr.l %[gm], %[out_0] \r\n"
- "asr.l %[gm], %[out_1] \r\n"
- : [out_0]"=&d"(out_0), [out_1]"=&d"(out_1)
- : [gm]"d"(global_muting)
- );
-
- out_buf [ 0] = out_0;
- out_buf [WAV_CHUNK_SIZE] = out_1;
- out_buf ++;
- #elif defined (CPU_ARM)
- /* Read feedback from echo buffer */
- int echo_pos = this->echo_pos;
- uint8_t* const echo_ptr = RAM +
- ((this->r.g.echo_page * 0x100 + echo_pos) & 0xFFFF);
echo_pos += 4;
- if ( echo_pos >= (this->r.g.echo_delay & 15) * 0x800 )
+
+ if ( echo_pos >= echo_delay )
echo_pos = 0;
- this->echo_pos = echo_pos;
- #if ARM_ARCH >= 6
- int32_t *fir_ptr, *fir_coeff;
- int fb_0, fb_1;
+ this->echo_pos = echo_pos;
/* Apply FIR */
-
- /* Keep last 8 samples */
- asm volatile (
- "ldr %[fb_0], [%[echo_p]] \r\n"
- "add %[fir_p], %[t_fir_p], #4 \r\n"
- "bic %[t_fir_p], %[fir_p], %[mask] \r\n"
- "str %[fb_0], [%[fir_p], #-4] \r\n"
- /* duplicate at +8 eliminates wrap checking below */
- "str %[fb_0], [%[fir_p], #28] \r\n"
- : [fir_p]"=&r"(fir_ptr), [t_fir_p]"+r"(this->fir_ptr),
- [fb_0]"=&r"(fb_0)
- : [echo_p]"r"(echo_ptr), [mask]"i"(~FIR_BUF_MASK));
-
- fir_coeff = (int32_t *)this->fir_coeff;
-
- /* Fugly, but the best version found. */
- int _0;
- asm volatile ( /* L0R0 = acc0 */
- "ldmia %[fir_p]!, { r2-r5 } \r\n" /* L1R1-L4R4 = r2-r5 */
- "ldmia %[fir_c]!, { r0-r1 } \r\n" /* C0C1-C2C3 = r0-r1 */
- "pkhbt %[_0], %[acc0], r2, asl #16 \r\n" /* L0R0,L1R1->L0L1,R0R1 */
- "pkhtb r2, r2, %[acc0], asr #16 \r\n"
- "smuad %[acc0], %[_0], r0 \r\n" /* acc0=L0*C0+L1*C1 */
- "smuad %[acc1], r2, r0 \r\n" /* acc1=R0*C0+R1*C1 */
- "pkhbt %[_0], r3, r4, asl #16 \r\n" /* L2R2,L3R3->L2L3,R2R3 */
- "pkhtb r4, r4, r3, asr #16 \r\n"
- "smlad %[acc0], %[_0], r1, %[acc0] \r\n" /* acc0+=L2*C2+L3*C3 */
- "smlad %[acc1], r4, r1, %[acc1] \r\n" /* acc1+=R2*C2+R3*C3 */
- "ldmia %[fir_p], { r2-r4 } \r\n" /* L5R5-L7R7 = r2-r4 */
- "ldmia %[fir_c], { r0-r1 } \r\n" /* C4C5-C6C7 = r0-r1 */
- "pkhbt %[_0], r5, r2, asl #16 \r\n" /* L4R4,L5R5->L4L5,R4R5 */
- "pkhtb r2, r2, r5, asr #16 \r\n"
- "smlad %[acc0], %[_0], r0, %[acc0] \r\n" /* acc0+=L4*C4+L5*C5 */
- "smlad %[acc1], r2, r0, %[acc1] \r\n" /* acc1+=R4*C4+R5*C5 */
- "pkhbt %[_0], r3, r4, asl #16 \r\n" /* L6R6,L7R7->L6L7,R6R7 */
- "pkhtb r4, r4, r3, asr #16 \r\n"
- "smlad %[acc0], %[_0], r1, %[acc0] \r\n" /* acc0+=L6*C6+L7*C7 */
- "smlad %[acc1], r4, r1, %[acc1] \r\n" /* acc1+=R6*C6+R7*C7 */
- : [acc0]"+r"(fb_0), [acc1]"=&r"(fb_1), [_0]"=&r"(_0),
- [fir_p]"+r"(fir_ptr), [fir_c]"+r"(fir_coeff)
- :
- : "r0", "r1", "r2", "r3", "r4", "r5");
-
- /* Generate output */
- int amp_0, amp_1;
-
- asm volatile (
- "mul %[amp_0], %[gvol_0], %[chans_0] \r\n"
- "mul %[amp_1], %[gvol_1], %[chans_1] \r\n"
- : [amp_0]"=&r"(amp_0), [amp_1]"=r"(amp_1)
- : [gvol_0]"r"(global_vol_0), [gvol_1]"r"(global_vol_1),
- [chans_0]"r"(chans_0), [chans_1]"r"(chans_1));
- asm volatile (
- "mla %[amp_0], %[fb_0], %[ev_0], %[amp_0] \r\n"
- "mla %[amp_1], %[fb_1], %[ev_1], %[amp_1] \r\n"
- : [amp_0]"+r"(amp_0), [amp_1]"+r"(amp_1)
- : [fb_0]"r"(fb_0), [fb_1]"r"(fb_1),
- [ev_0]"r"((int)this->r.g.echo_volume_0),
- [ev_1]"r"((int)this->r.g.echo_volume_1));
-
- out_buf [ 0] = amp_0 >> global_muting;
- out_buf [WAV_CHUNK_SIZE] = amp_1 >> global_muting;
- out_buf ++;
+ int fb_0, fb_1;
+ echo_apply( this, echo_ptr, &fb_0, &fb_1 );
if ( !(this->r.g.flags & 0x20) )
{
/* Feedback into echo buffer */
- int e0, e1;
-
- asm volatile (
- "mov %[e0], %[echo_0], asl #7 \r\n"
- "mov %[e1], %[echo_1], asl #7 \r\n"
- "mla %[e0], %[fb_0], %[efb], %[e0] \r\n"
- "mla %[e1], %[fb_1], %[efb], %[e1] \r\n"
- : [e0]"=&r"(e0), [e1]"=&r"(e1)
- : [echo_0]"r"(echo_0), [echo_1]"r"(echo_1),
- [fb_0]"r"(fb_0), [fb_1]"r"(fb_1),
- [efb]"r"((int)this->r.g.echo_feedback));
- asm volatile (
- "ssat %[e0], #16, %[e0], asr #14 \r\n"
- "ssat %[e1], #16, %[e1], asr #14 \r\n"
- "pkhbt %[e0], %[e0], %[e1], lsl #16 \r\n"
- "str %[e0], [%[echo_p]] \r\n"
- : [e0]"+r"(e0), [e1]"+r"(e1)
- : [echo_p]"r"(echo_ptr));
+ echo_feedback( this, echo_ptr, echo_0, echo_1, fb_0, fb_1 );
}
- #else /* ARM_ARCH < 6 */
- int fb_0 = GET_LE16SA( echo_ptr );
- int fb_1 = GET_LE16SA( echo_ptr + 2 );
- int32_t *fir_ptr, *fir_coeff;
-
- /* Keep last 8 samples */
-
- /* Apply FIR */
- asm volatile (
- "add %[fir_p], %[t_fir_p], #8 \r\n"
- "bic %[t_fir_p], %[fir_p], %[mask] \r\n"
- "str %[fb_0], [%[fir_p], #-8] \r\n"
- "str %[fb_1], [%[fir_p], #-4] \r\n"
- /* duplicate at +8 eliminates wrap checking below */
- "str %[fb_0], [%[fir_p], #56] \r\n"
- "str %[fb_1], [%[fir_p], #60] \r\n"
- : [fir_p]"=&r"(fir_ptr), [t_fir_p]"+r"(this->fir_ptr)
- : [fb_0]"r"(fb_0), [fb_1]"r"(fb_1), [mask]"i"(~FIR_BUF_MASK));
-
- fir_coeff = this->fir_coeff;
-
- asm volatile (
- "ldmia %[fir_c]!, { r0-r1 } \r\n"
- "ldmia %[fir_p]!, { r4-r5 } \r\n"
- "mul %[fb_0], r0, %[fb_0] \r\n"
- "mul %[fb_1], r0, %[fb_1] \r\n"
- "mla %[fb_0], r4, r1, %[fb_0] \r\n"
- "mla %[fb_1], r5, r1, %[fb_1] \r\n"
- "ldmia %[fir_c]!, { r0-r1 } \r\n"
- "ldmia %[fir_p]!, { r2-r5 } \r\n"
- "mla %[fb_0], r2, r0, %[fb_0] \r\n"
- "mla %[fb_1], r3, r0, %[fb_1] \r\n"
- "mla %[fb_0], r4, r1, %[fb_0] \r\n"
- "mla %[fb_1], r5, r1, %[fb_1] \r\n"
- "ldmia %[fir_c]!, { r0-r1 } \r\n"
- "ldmia %[fir_p]!, { r2-r5 } \r\n"
- "mla %[fb_0], r2, r0, %[fb_0] \r\n"
- "mla %[fb_1], r3, r0, %[fb_1] \r\n"
- "mla %[fb_0], r4, r1, %[fb_0] \r\n"
- "mla %[fb_1], r5, r1, %[fb_1] \r\n"
- "ldmia %[fir_c]!, { r0-r1 } \r\n"
- "ldmia %[fir_p]!, { r2-r5 } \r\n"
- "mla %[fb_0], r2, r0, %[fb_0] \r\n"
- "mla %[fb_1], r3, r0, %[fb_1] \r\n"
- "mla %[fb_0], r4, r1, %[fb_0] \r\n"
- "mla %[fb_1], r5, r1, %[fb_1] \r\n"
- : [fb_0]"+r"(fb_0), [fb_1]"+r"(fb_1),
- [fir_p]"+r"(fir_ptr), [fir_c]"+r"(fir_coeff)
- :
- : "r0", "r1", "r2", "r3", "r4", "r5");
-
- /* Generate output */
- int amp_0 = (chans_0 * global_vol_0 + fb_0 * this->r.g.echo_volume_0)
- >> global_muting;
- int amp_1 = (chans_1 * global_vol_1 + fb_1 * this->r.g.echo_volume_1)
- >> global_muting;
-
- out_buf [ 0] = amp_0;
- out_buf [WAV_CHUNK_SIZE] = amp_1;
- out_buf ++;
+ #endif /* !SPC_NOECHO */
- if ( !(this->r.g.flags & 0x20) )
- {
- /* Feedback into echo buffer */
- int e0 = (echo_0 >> 7) + ((fb_0 * this->r.g.echo_feedback) >> 14);
- int e1 = (echo_1 >> 7) + ((fb_1 * this->r.g.echo_feedback) >> 14);
- e0 = CLAMP16( e0 );
- SET_LE16A( echo_ptr , e0 );
- e1 = CLAMP16( e1 );
- SET_LE16A( echo_ptr + 2, e1 );
- }
- #endif /* ARM_ARCH */
- #else /* Unoptimized CPU */
- /* Read feedback from echo buffer */
- int echo_pos = this->echo_pos;
- uint8_t* const echo_ptr = RAM +
- ((this->r.g.echo_page * 0x100 + echo_pos) & 0xFFFF);
- echo_pos += 4;
- if ( echo_pos >= (this->r.g.echo_delay & 15) * 0x800 )
- echo_pos = 0;
- this->echo_pos = echo_pos;
- int fb_0 = GET_LE16SA( echo_ptr );
- int fb_1 = GET_LE16SA( echo_ptr + 2 );
-
- /* Keep last 8 samples */
- int (* const fir_ptr) [2] = this->fir_buf + this->fir_pos;
- this->fir_pos = (this->fir_pos + 1) & (FIR_BUF_HALF - 1);
- fir_ptr [ 0] [0] = fb_0;
- fir_ptr [ 0] [1] = fb_1;
- /* duplicate at +8 eliminates wrap checking below */
- fir_ptr [FIR_BUF_HALF] [0] = fb_0;
- fir_ptr [FIR_BUF_HALF] [1] = fb_1;
-
- /* Apply FIR */
- fb_0 *= this->fir_coeff [0];
- fb_1 *= this->fir_coeff [0];
+ mix_output( this, global_muting, global_vol_0, global_vol_1,
+ chans_0, chans_1, fb_0, fb_1, &amp_0, &amp_1 );
- #define DO_PT( i )\
- fb_0 += fir_ptr [i] [0] * this->fir_coeff [i];\
- fb_1 += fir_ptr [i] [1] * this->fir_coeff [i];
-
- DO_PT( 1 )
- DO_PT( 2 )
- DO_PT( 3 )
- DO_PT( 4 )
- DO_PT( 5 )
- DO_PT( 6 )
- DO_PT( 7 )
-
- /* Generate output */
- int amp_0 = (chans_0 * global_vol_0 + fb_0 * this->r.g.echo_volume_0)
- >> global_muting;
- int amp_1 = (chans_1 * global_vol_1 + fb_1 * this->r.g.echo_volume_1)
- >> global_muting;
- out_buf [ 0] = amp_0;
- out_buf [WAV_CHUNK_SIZE] = amp_1;
- out_buf ++;
-
- if ( !(this->r.g.flags & 0x20) )
- {
- /* Feedback into echo buffer */
- int e0 = (echo_0 >> 7) + ((fb_0 * this->r.g.echo_feedback) >> 14);
- int e1 = (echo_1 >> 7) + ((fb_1 * this->r.g.echo_feedback) >> 14);
- e0 = CLAMP16( e0 );
- SET_LE16A( echo_ptr , e0 );
- e1 = CLAMP16( e1 );
- SET_LE16A( echo_ptr + 2, e1 );
- }
- #endif /* CPU_* */
- #else /* SPCNOECHO == 1*/
- /* Generate output */
- int amp_0 = (chans_0 * global_vol_0) >> global_muting;
- int amp_1 = (chans_1 * global_vol_1) >> global_muting;
out_buf [ 0] = amp_0;
out_buf [WAV_CHUNK_SIZE] = amp_1;
out_buf ++;
- #endif /* SPCNOECHO */
}
while ( --count );
-#if 0
+
EXIT_TIMER(dsp);
ENTER_TIMER(cpu);
-#endif
}
void DSP_reset( struct Spc_Dsp* this )
@@ -1563,31 +913,22 @@ void DSP_reset( struct Spc_Dsp* this )
ci->memset( this->voice_state, 0, sizeof this->voice_state );
- int i;
- for ( i = VOICE_COUNT; --i >= 0; )
+ for ( int i = VOICE_COUNT; --i >= 0; )
{
struct voice_t* v = this->voice_state + i;
v->env_mode = state_release;
- v->addr = ram.ram;
+ v->wave.addr = ram.ram;
}
- #if SPC_BRRCACHE
- this->oldsize = 0;
- for ( i = 0; i < 256; i++ )
- this->wave_entry [i].start_addr = -1;
- #endif
-
-#if defined(CPU_COLDFIRE)
- this->fir_ptr = fir_buf;
- this->last_fir_ptr = &fir_buf [7];
- ci->memset( fir_buf, 0, sizeof fir_buf );
-#elif defined (CPU_ARM)
- this->fir_ptr = fir_buf;
- ci->memset( fir_buf, 0, sizeof fir_buf );
-#else
- this->fir_pos = 0;
- ci->memset( this->fir_buf, 0, sizeof this->fir_buf );
-#endif
+#if SPC_BRRCACHE
+ this->oldsize = 0;
+ for ( int i = 0; i < 256; i++ )
+ this->wave_entry [i].start_addr = -1;
+#endif /* SPC_BRRCACHE */
+
+#if !SPC_NOECHO
+ echo_init(this);
+#endif /* SPC_NOECHO */
assert( offsetof (struct globals_t,unused9 [2]) == REGISTER_COUNT );
assert( sizeof (this->r.voice) == REGISTER_COUNT );
diff --git a/lib/rbcodec/codecs/libspc/spc_dsp_generic.c b/lib/rbcodec/codecs/libspc/spc_dsp_generic.c
new file mode 100644
index 0000000000..60e79f8763
--- /dev/null
+++ b/lib/rbcodec/codecs/libspc/spc_dsp_generic.c
@@ -0,0 +1,211 @@
+/***************************************************************************
+ * __________ __ ___.
+ * Open \______ \ ____ ____ | | _\_ |__ _______ ___
+ * Source | _// _ \_/ ___\| |/ /| __ \ / _ \ \/ /
+ * Jukebox | | ( <_> ) \___| < | \_\ ( <_> > < <
+ * Firmware |____|_ /\____/ \___ >__|_ \|___ /\____/__/\_ \
+ * \/ \/ \/ \/ \/
+ * $Id$
+ *
+ * Copyright (C) 2006-2007 Adam Gashlin (hcs)
+ * Copyright (C) 2004-2007 Shay Green (blargg)
+ * Copyright (C) 2002 Brad Martin
+ *
+ * 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.
+ *
+ ****************************************************************************/
+static inline int apply_gen_envx( struct voice_t* voice, int output )
+{
+ return (output * voice->envx) >> 11;
+}
+
+static inline int apply_gen_volume( struct voice_t* voice, int output,
+ int* amp_0, int* amp_1 )
+{
+ *amp_0 = voice->volume [0] * output;
+ *amp_1 = voice->volume [1] * output;
+ return output;
+}
+
+static inline int apply_gen_amp( struct voice_t* voice, int output,
+ int* amp_0, int* amp_1)
+{
+ output = apply_gen_envx( voice, output );
+ output = apply_gen_volume( voice, output, amp_0, amp_1 );
+ return output;
+}
+
+#if !SPC_NOINTERP
+
+#ifndef SPC_GAUSSIAN_FAST_INTERP
+static inline int gaussian_fast_interp( int16_t const* samples,
+ int32_t position,
+ int16_t const* fwd,
+ int16_t const* rev )
+{
+ samples += position >> 12;
+ return (fwd [0] * samples [0] +
+ fwd [1] * samples [1] +
+ rev [1] * samples [2] +
+ rev [0] * samples [3]) >> 11;
+}
+#endif /* SPC_GAUSSIAN_FAST_INTERP */
+
+#ifndef SPC_GAUSSIAN_FAST_AMP
+#define gaussian_fast_amp apply_amp
+#endif /* SPC_GAUSSIAN_FAST_AMP */
+
+#ifndef SPC_GAUSSIAN_SLOW_INTERP
+static inline int gaussian_slow_interp( int16_t const* samples,
+ int32_t position,
+ int16_t const* fwd,
+ int16_t const* rev )
+{
+ int output;
+ samples += position >> 12;
+ output = (fwd [0] * samples [0]) & ~0xFFF;
+ output = (output + fwd [1] * samples [1]) & ~0xFFF;
+ output = (output + rev [1] * samples [2]) >> 12;
+ output = (int16_t) (output * 2);
+ output += ((rev [0] * samples [3]) >> 12) * 2;
+ return CLAMP16( output );
+}
+#endif /* SPC_GAUSSIAN_SLOW_INTERP */
+
+#ifndef SPC_GAUSSIAN_SLOW_AMP
+static inline int gaussian_slow_amp( struct voice_t* voice, int output,
+ int *amp_0, int *amp_1 )
+{
+ output = apply_gen_envx( voice, output ) & ~1;
+ output = apply_gen_volume( voice, output, amp_0, amp_1 );
+ return output;
+}
+#endif /* SPC_GAUSSIAN_SLOW_AMP */
+
+#define interp gaussian_slow_interp
+#define apply_amp gaussian_slow_amp
+
+#else /* SPC_NOINTERP */
+
+#ifndef SPC_LINEAR_INTERP
+static inline int linear_interp( int16_t const* samples, int32_t position )
+{
+ int32_t fraction = position & 0xfff;
+ int16_t const* pos = (samples + (position >> 12)) + 1;
+ return pos[0] + ((fraction * (pos[1] - pos[0])) >> 12);
+}
+#endif /* SPC_LINEAR_INTERP */
+
+#define interp( samp, pos, fwd, rev ) \
+ linear_interp( (samp), (pos) )
+
+#ifndef SPC_LINEAR_AMP
+#define linear_amp apply_gen_amp
+#endif /* SPC_LINEAR_AMP */
+
+#define apply_amp linear_amp
+#endif /* SPC_NOINTERP */
+
+
+#if !SPC_NOECHO
+
+#ifndef SPC_DSP_ECHO_APPLY
+/* Init FIR filter */
+static inline void echo_init( struct Spc_Dsp* this )
+{
+ this->fir.pos = 0;
+ ci->memset( this->fir.buf, 0, sizeof this->fir.buf );
+}
+
+/* Apply FIR filter */
+static inline void echo_apply(struct Spc_Dsp* this,
+ uint8_t* const echo_ptr, int* out_0, int* out_1)
+{
+ int fb_0 = GET_LE16SA( echo_ptr );
+ int fb_1 = GET_LE16SA( echo_ptr + 2 );
+
+ /* Keep last 8 samples */
+ int (* const fir_ptr) [2] = this->fir.buf + this->fir.pos;
+ this->fir.pos = (this->fir.pos + 1) & (FIR_BUF_HALF - 1);
+
+ fir_ptr [ 0] [0] = fb_0;
+ fir_ptr [ 0] [1] = fb_1;
+ /* duplicate at +8 eliminates wrap checking below */
+ fir_ptr [FIR_BUF_HALF] [0] = fb_0;
+ fir_ptr [FIR_BUF_HALF] [1] = fb_1;
+
+ fb_0 *= this->fir.coeff [0];
+ fb_1 *= this->fir.coeff [0];
+
+ #define DO_PT( i ) \
+ fb_0 += fir_ptr [i] [0] * this->fir.coeff [i]; \
+ fb_1 += fir_ptr [i] [1] * this->fir.coeff [i];
+
+ DO_PT( 1 )
+ DO_PT( 2 )
+ DO_PT( 3 )
+ DO_PT( 4 )
+ DO_PT( 5 )
+ DO_PT( 6 )
+ DO_PT( 7 )
+
+ #undef DO_PT
+
+ *out_0 = fb_0;
+ *out_1 = fb_1;
+}
+#endif /* SPC_DSP_ECHO_APPLY */
+
+#ifndef SPC_DSP_ECHO_FEEDBACK
+/* Feedback into echo buffer */
+static inline void echo_feedback( struct Spc_Dsp* this, uint8_t *echo_ptr,
+ int echo_0, int echo_1, int fb_0, int fb_1 )
+{
+ int e0 = (echo_0 >> 7) + ((fb_0 * this->r.g.echo_feedback) >> 14);
+ int e1 = (echo_1 >> 7) + ((fb_1 * this->r.g.echo_feedback) >> 14);
+ e0 = CLAMP16( e0 );
+ SET_LE16A( echo_ptr , e0 );
+ e1 = CLAMP16( e1 );
+ SET_LE16A( echo_ptr + 2, e1 );
+}
+#endif /* SPC_DSP_ECHO_FEEDBACK */
+
+#ifndef SPC_DSP_GENERATE_OUTPUT
+/* Generate final output */
+static inline void echo_output( struct Spc_Dsp* this, int global_muting,
+ int global_vol_0, int global_vol_1, int chans_0, int chans_1,
+ int fb_0, int fb_1, int* out_0, int* out_1 )
+{
+ *out_0 = (chans_0 * global_vol_0 + fb_0 * this->r.g.echo_volume_0)
+ >> global_muting;
+ *out_1 = (chans_1 * global_vol_1 + fb_1 * this->r.g.echo_volume_1)
+ >> global_muting;
+}
+#endif /* SPC_DSP_GENERATE_OUTPUT */
+
+#define mix_output echo_output
+
+#else /* SPC_NOECHO */
+
+#ifndef SPC_DSP_GENERATE_OUTPUT
+/* Generate final output */
+static inline void noecho_output( struct Spc_Dsp* this, int global_muting,
+ int global_vol_0, int global_vol_1, int chans_0, int chans_1,
+ int* out_0, int* out_1 )
+{
+ *out_0 = (chans_0 * global_vol_0) >> global_muting;
+ *out_1 = (chans_1 * global_vol_1) >> global_muting;
+ (void)this;
+}
+#endif /* SPC_DSP_GENERATE_OUTPUT */
+
+#define mix_output(this, gm, gv0, gv1, ch0, ch1, fb_0, fb_1, o0, o1) \
+ noecho_output( (this), (gm), (gv0), (gv1), (ch0), (ch1), (o0), (o1) )
+
+#endif /* !SPC_NOECHO */
diff --git a/lib/rbcodec/codecs/libspc/spc_dsp_generic.h b/lib/rbcodec/codecs/libspc/spc_dsp_generic.h
new file mode 100644
index 0000000000..beeb87deb2
--- /dev/null
+++ b/lib/rbcodec/codecs/libspc/spc_dsp_generic.h
@@ -0,0 +1,45 @@
+/***************************************************************************
+ * __________ __ ___.
+ * Open \______ \ ____ ____ | | _\_ |__ _______ ___
+ * Source | _// _ \_/ ___\| |/ /| __ \ / _ \ \/ /
+ * Jukebox | | ( <_> ) \___| < | \_\ ( <_> > < <
+ * Firmware |____|_ /\____/ \___ >__|_ \|___ /\____/__/\_ \
+ * \/ \/ \/ \/ \/
+ * $Id$
+ *
+ * Copyright (C) 2006-2007 Adam Gashlin (hcs)
+ * Copyright (C) 2004-2007 Shay Green (blargg)
+ * Copyright (C) 2002 Brad Martin
+ *
+ * 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.
+ *
+ ****************************************************************************/
+#if !SPC_NOECHO
+
+#ifndef SPC_DSP_ECHO_APPLY
+enum
+{
+ FIR_BUF_CNT = FIR_BUF_HALF * 2 * 2,
+ FIR_BUF_SIZE = FIR_BUF_CNT * sizeof ( int32_t ),
+ FIR_BUF_ALIGN = FIR_BUF_SIZE,
+ FIR_BUF_MASK = ~((FIR_BUF_ALIGN / 2) | (sizeof ( int32_t ) * 2 - 1))
+};
+
+/* Echo filter structure embedded in struct Spc_Dsp */
+struct echo_filter
+{
+ /* fir_buf [i + 8] == fir_buf [i], to avoid wrap checking in FIR code */
+ int pos; /* (0 to 7) */
+ int buf [FIR_BUF_HALF * 2] [2];
+ /* copy of echo FIR constants as int, for faster access */
+ int coeff [VOICE_COUNT];
+};
+#endif /* SPC_DSP_ECHO_APPLY */
+
+#endif /* !SPC_NOECHO */
diff --git a/lib/rbcodec/codecs/libspc/spc_emu.c b/lib/rbcodec/codecs/libspc/spc_emu.c
index 5ea5b0cdeb..dab4199ef0 100644
--- a/lib/rbcodec/codecs/libspc/spc_emu.c
+++ b/lib/rbcodec/codecs/libspc/spc_emu.c
@@ -32,8 +32,8 @@ struct cpu_ram_t ram IBSS_ATTR_SPC_LARGE_IRAM CACHEALIGN_ATTR;
/**************** Timers ****************/
-static void Timer_run_( struct Timer* t, long time ) ICODE_ATTR_SPC;
-static void Timer_run_( struct Timer* t, long time )
+static void NO_INLINE ICODE_ATTR_SPC
+Timer_run_( struct Timer* t, long time )
{
/* when disabled, next_tick should always be in the future */
assert( t->enabled );
@@ -60,7 +60,7 @@ static inline void Timer_run( struct Timer* t, long time )
/**************** SPC emulator ****************/
/* 1.024 MHz clock / 32000 samples per second */
-static void SPC_enable_rom( THIS, int enable )
+static void NO_INLINE SPC_enable_rom( THIS, int enable )
{
if ( this->rom_enabled != enable )
{
@@ -186,8 +186,8 @@ int SPC_load_spc( THIS, const void* data, long size )
}
/**************** DSP interaction ****************/
-static void SPC_run_dsp_( THIS, long time ) ICODE_ATTR_SPC;
-static void SPC_run_dsp_( THIS, long time )
+static void NO_INLINE ICODE_ATTR_SPC
+SPC_run_dsp_( THIS, long time )
{
/* divide by CLOCKS_PER_SAMPLE */
int count = ((time - this->next_dsp) >> 5) + 1;
@@ -383,13 +383,10 @@ int SPC_play( THIS, long count, int32_t* out )
}
/* Catch DSP up to present */
-#if 0
ENTER_TIMER(cpu);
-#endif
SPC_run_dsp( this, -EXTRA_CLOCKS );
-#if 0
EXIT_TIMER(cpu);
-#endif
+
assert( this->next_dsp == CLOCKS_PER_SAMPLE - EXTRA_CLOCKS );
assert( this->sample_buf - out == count );
generated by cgit (git 2.34.1) at 2024-04-19 06:24:05 +0000