NetBSD/sys/dev/audio/linear.c

/*	$NetBSD: linear.c,v 1.3 2020/01/11 04:06:13 isaki Exp $	*/

/*
 * Copyright (C) 2017 Tetsuya Isaki. All rights reserved.
 * Copyright (C) 2017 Y.Sugahara (moveccr). All rights reserved.
 *
 * Redistribution and use in source and binary forms, with or without
 * modification, are permitted provided that the following conditions
 * are met:
 * 1. Redistributions of source code must retain the above copyright
 *    notice, this list of conditions and the following disclaimer.
 * 2. Redistributions in binary form must reproduce the above copyright
 *    notice, this list of conditions and the following disclaimer in the
 *    documentation and/or other materials provided with the distribution.
 *
 * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR
 * IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES
 * OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED.
 * IN NO EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT,
 * INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING,
 * BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES;
 * LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED
 * AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY,
 * OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY
 * OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF
 * SUCH DAMAGE.
 */

#include <sys/cdefs.h>
__KERNEL_RCSID(0, "$NetBSD: linear.c,v 1.3 2020/01/11 04:06:13 isaki Exp $");

#include <sys/types.h>
#include <sys/systm.h>
#include <sys/device.h>
#include <dev/audio/audiovar.h>
#include <dev/audio/linear.h>

/*
 * audio_linear8_to_internal:
 *	This filter performs conversion from [US]LINEAR8 to internal format.
 */
void
audio_linear8_to_internal(audio_filter_arg_t *arg)
{
	const uint8_t *s;
	aint_t *d;
	uint8_t xor;
	u_int sample_count;
	u_int i;

	DIAGNOSTIC_filter_arg(arg);
	KASSERT(audio_format2_is_linear(arg->srcfmt));
	KASSERT(arg->srcfmt->precision == 8);
	KASSERT(arg->srcfmt->stride == 8);
	KASSERT(audio_format2_is_internal(arg->dstfmt));
	KASSERT(arg->srcfmt->channels == arg->dstfmt->channels);

	s = arg->src;
	d = arg->dst;
	sample_count = arg->count * arg->srcfmt->channels;
	xor = audio_format2_is_signed(arg->srcfmt) ? 0 : 0x80;

	for (i = 0; i < sample_count; i++) {
		uint8_t val;
		val = *s++;
		val ^= xor;
		*d++ = (auint_t)val << (AUDIO_INTERNAL_BITS - 8);
	}
}

/*
 * audio_internal_to_linear8:
 *	This filter performs conversion from internal format to [US]LINEAR8.
 */
void
audio_internal_to_linear8(audio_filter_arg_t *arg)
{
	const aint_t *s;
	uint8_t *d;
	uint8_t xor;
	u_int sample_count;
	u_int i;

	DIAGNOSTIC_filter_arg(arg);
	KASSERT(audio_format2_is_linear(arg->dstfmt));
	KASSERT(arg->dstfmt->precision == 8);
	KASSERT(arg->dstfmt->stride == 8);
	KASSERT(audio_format2_is_internal(arg->srcfmt));
	KASSERT(arg->srcfmt->channels == arg->dstfmt->channels);

	s = arg->src;
	d = arg->dst;
	sample_count = arg->count * arg->srcfmt->channels;
	xor = audio_format2_is_signed(arg->dstfmt) ? 0 : 0x80;

	for (i = 0; i < sample_count; i++) {
		uint8_t val;
		val = (*s++) >> (AUDIO_INTERNAL_BITS - 8);
		val ^= xor;
		*d++ = val;
	}
}

/*
 * audio_linear16_to_internal:
 *	This filter performs conversion from [US]LINEAR16{LE,BE} to internal
 *	format.
 */
void
audio_linear16_to_internal(audio_filter_arg_t *arg)
{
	const uint16_t *s;
	aint_t *d;
	uint16_t xor;
	u_int sample_count;
	u_int shift;
	u_int i;
	bool is_src_NE;

	DIAGNOSTIC_filter_arg(arg);
	KASSERT(audio_format2_is_linear(arg->srcfmt));
	KASSERT(arg->srcfmt->precision == 16);
	KASSERT(arg->srcfmt->stride == 16);
	KASSERT(audio_format2_is_internal(arg->dstfmt));
	KASSERT(arg->srcfmt->channels == arg->dstfmt->channels);

	s = arg->src;
	d = arg->dst;
	sample_count = arg->count * arg->srcfmt->channels;

	shift = AUDIO_INTERNAL_BITS - 16;
	xor = audio_format2_is_signed(arg->srcfmt) ? 0 : 0x8000;
	is_src_NE = (audio_format2_endian(arg->srcfmt) == BYTE_ORDER);

	/*
	 * Since slinear16_OppositeEndian to slinear_NativeEndian is used
	 * so much especially on big endian machines, so it's expanded.
	 * Other conversions are rarely used, so they are compressed.
	 */
	if (__predict_true(xor == 0) && is_src_NE == false) {
		/* slinear16_OE to slinear<AI>_NE */
		for (i = 0; i < sample_count; i++) {
			uint16_t val;
			val = *s++;
			val = bswap16(val);
			*d++ = (auint_t)val << shift;
		}
	} else {
		/* slinear16_NE      to slinear<AI>_NE */
		/* ulinear16_{NE,OE} to slinear<AI>_NE */
		for (i = 0; i < sample_count; i++) {
			uint16_t val;
			val = *s++;
			if (!is_src_NE)
				val = bswap16(val);
			val ^= xor;
			*d++ = (auint_t)val << shift;
		}
	}
}

/*
 * audio_internal_to_linear16:
 *	This filter performs conversion from internal format to
 *	[US]LINEAR16{LE,BE}.
 */
void
audio_internal_to_linear16(audio_filter_arg_t *arg)
{
	const aint_t *s;
	uint16_t *d;
	uint16_t xor;
	u_int sample_count;
	u_int shift;
	u_int i;
	bool is_dst_NE;

	DIAGNOSTIC_filter_arg(arg);
	KASSERT(audio_format2_is_linear(arg->dstfmt));
	KASSERT(arg->dstfmt->precision == 16);
	KASSERT(arg->dstfmt->stride == 16);
	KASSERT(audio_format2_is_internal(arg->srcfmt));
	KASSERT(arg->srcfmt->channels == arg->dstfmt->channels);

	s = arg->src;
	d = arg->dst;
	sample_count = arg->count * arg->srcfmt->channels;

	shift = AUDIO_INTERNAL_BITS - 16;
	xor = audio_format2_is_signed(arg->dstfmt) ? 0 : 0x8000;
	is_dst_NE = (audio_format2_endian(arg->dstfmt) == BYTE_ORDER);

	/*
	 * Since slinear_NativeEndian to slinear16_OppositeEndian is used
	 * so much especially on big endian machines, so it's expanded.
	 * Other conversions are rarely used, so they are compressed.
	 */
	if (__predict_true(xor == 0) && is_dst_NE == false) {
		/* slinear<AI>_NE -> slinear16_OE */
		for (i = 0; i < sample_count; i++) {
			uint16_t val;
			val = (*s++) >> shift;
			val = bswap16(val);
			*d++ = val;
		}
	} else {
		/* slinear<AI>_NE -> slinear16_NE */
		/* slinear<AI>_NE -> ulinear16_{NE,OE} */
		for (i = 0; i < sample_count; i++) {
			uint16_t val;
			val = (*s++) >> shift;
			val ^= xor;
			if (!is_dst_NE)
				val = bswap16(val);
			*d++ = val;
		}
	}
}

#if defined(AUDIO_SUPPORT_LINEAR24)
/*
 * audio_linear24_to_internal:
 *	This filter performs conversion from [US]LINEAR24/24{LE,BE} to
 *	internal format.  Since it's rerely used, it's size optimized.
 */
void
audio_linear24_to_internal(audio_filter_arg_t *arg)
{
	const uint8_t *s;
	aint_t *d;
	auint_t xor;
	u_int sample_count;
	u_int i;
	bool is_src_LE;

	DIAGNOSTIC_filter_arg(arg);
	KASSERT(audio_format2_is_linear(arg->srcfmt));
	KASSERT(arg->srcfmt->precision == 24);
	KASSERT(arg->srcfmt->stride == 24);
	KASSERT(audio_format2_is_internal(arg->dstfmt));
	KASSERT(arg->srcfmt->channels == arg->dstfmt->channels);

	s = arg->src;
	d = arg->dst;
	sample_count = arg->count * arg->srcfmt->channels;
	xor = audio_format2_is_signed(arg->srcfmt)
	    ? 0 : (1 << (AUDIO_INTERNAL_BITS - 1));
	is_src_LE = (audio_format2_endian(arg->srcfmt) == LITTLE_ENDIAN);

	for (i = 0; i < sample_count; i++) {
		uint32_t val;
		if (is_src_LE) {
			val = s[0] | (s[1] << 8) | (s[2] << 16);
		} else {
			val = (s[0] << 16) | (s[1] << 8) | s[2];
		}
		s += 3;

#if AUDIO_INTERNAL_BITS < 24
		val >>= 24 - AUDIO_INTERNAL_BITS;
#else
		val <<= AUDIO_INTERNAL_BITS - 24;
#endif
		val ^= xor;
		*d++ = val;
	}
}

/*
 * audio_internal_to_linear24:
 *	This filter performs conversion from internal format to
 *	[US]LINEAR24/24{LE,BE}.  Since it's rarely used, it's size optimized.
 */
void
audio_internal_to_linear24(audio_filter_arg_t *arg)
{
	const aint_t *s;
	uint8_t *d;
	auint_t xor;
	u_int sample_count;
	u_int i;
	bool is_dst_LE;

	DIAGNOSTIC_filter_arg(arg);
	KASSERT(audio_format2_is_linear(arg->dstfmt));
	KASSERT(arg->dstfmt->precision == 24);
	KASSERT(arg->dstfmt->stride == 24);
	KASSERT(audio_format2_is_internal(arg->srcfmt));
	KASSERT(arg->srcfmt->channels == arg->dstfmt->channels);

	s = arg->src;
	d = arg->dst;
	sample_count = arg->count * arg->srcfmt->channels;
	xor = audio_format2_is_signed(arg->dstfmt)
	    ? 0 : (1 << (AUDIO_INTERNAL_BITS - 1));
	is_dst_LE = (audio_format2_endian(arg->dstfmt) == LITTLE_ENDIAN);

	for (i = 0; i < sample_count; i++) {
		uint32_t val;
		val = *s++;
		val ^= xor;
#if AUDIO_INTERNAL_BITS < 24
		val <<= 24 - AUDIO_INTERNAL_BITS;
#else
		val >>= AUDIO_INTERNAL_BITS - 24;
#endif
		if (is_dst_LE) {
			d[0] = val & 0xff;
			d[1] = (val >> 8) & 0xff;
			d[2] = (val >> 16) & 0xff;
		} else {
			d[0] = (val >> 16) & 0xff;
			d[1] = (val >> 8) & 0xff;
			d[2] = val & 0xff;
		}
		d += 3;
	}
}
#endif /* AUDIO_SUPPORT_LINEAR24 */

/*
 * audio_linear32_to_internal:
 *	This filter performs conversion from [US]LINEAR32{LE,BE} to internal
 *	format.  Since it's rarely used, it's size optimized.
 */
void
audio_linear32_to_internal(audio_filter_arg_t *arg)
{
	const uint32_t *s;
	aint_t *d;
	auint_t xor;
	u_int sample_count;
	u_int i;
	bool is_src_NE;

	DIAGNOSTIC_filter_arg(arg);
	KASSERT(audio_format2_is_linear(arg->srcfmt));
	KASSERT(arg->srcfmt->precision == 32);
	KASSERT(arg->srcfmt->stride == 32);
	KASSERT(audio_format2_is_internal(arg->dstfmt));
	KASSERT(arg->srcfmt->channels == arg->dstfmt->channels);

	s = arg->src;
	d = arg->dst;
	sample_count = arg->count * arg->srcfmt->channels;
	xor = audio_format2_is_signed(arg->srcfmt)
	    ? 0 : (1 << (AUDIO_INTERNAL_BITS - 1));
	is_src_NE = (audio_format2_endian(arg->srcfmt) == BYTE_ORDER);

	for (i = 0; i < sample_count; i++) {
		uint32_t val;
		val = *s++;
		if (!is_src_NE)
			val = bswap32(val);
		val >>= 32 - AUDIO_INTERNAL_BITS;
		val ^= xor;
		*d++ = val;
	}
}

/*
 * audio_internal_to_linear32:
 *	This filter performs conversion from internal format to
 *	[US]LINEAR32{LE,BE}.  Since it's rarely used, it's size optimized.
 */
void
audio_internal_to_linear32(audio_filter_arg_t *arg)
{
	const aint_t *s;
	uint32_t *d;
	auint_t xor;
	u_int sample_count;
	u_int i;
	bool is_dst_NE;

	DIAGNOSTIC_filter_arg(arg);
	KASSERT(audio_format2_is_linear(arg->dstfmt));
	KASSERT(arg->dstfmt->precision == 32);
	KASSERT(arg->dstfmt->stride == 32);
	KASSERT(audio_format2_is_internal(arg->srcfmt));
	KASSERT(arg->srcfmt->channels == arg->dstfmt->channels);

	s = arg->src;
	d = arg->dst;
	sample_count = arg->count * arg->srcfmt->channels;
	xor = audio_format2_is_signed(arg->dstfmt)
	    ? 0 : (1 << (AUDIO_INTERNAL_BITS - 1));
	is_dst_NE = (audio_format2_endian(arg->dstfmt) == BYTE_ORDER);

	for (i = 0; i < sample_count; i++) {
		uint32_t val;
		val = *s++;
		val ^= xor;
		val <<= 32 - AUDIO_INTERNAL_BITS;
		if (!is_dst_NE)
			val = bswap32(val);
		*d++ = val;
	}
}