470bcabd8f
Apparently we don't use __MSC_VER as a compiler anymore and we always require a C99 compiler (which means we always have __func__) so we don't need a special AUDIO_FUNC macro. We can just replace AUDIO_FUNC with __func__ instead. Checkpatch failures were manually fixed. Signed-off-by: Alistair Francis <alistair.francis@xilinx.com> Cc: Gerd Hoffmann <kraxel@redhat.com> Reviewed-by: Thomas Huth <thuth@redhat.com> Reviewed-by: Eric Blake <eblake@redhat.com> Reviewed-by: Gerd Hoffmann <kraxel@redhat.com> Signed-off-by: Markus Armbruster <armbru@redhat.com> Reviewed-by: Philippe Mathieu-Daudé <f4bug@amsat.org> Message-Id: <20180203084315.20497-2-armbru@redhat.com>
401 lines
9.9 KiB
C
401 lines
9.9 KiB
C
/*
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* QEMU Mixing engine
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*
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* Copyright (c) 2004-2005 Vassili Karpov (malc)
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* Copyright (c) 1998 Fabrice Bellard
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*
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* Permission is hereby granted, free of charge, to any person obtaining a copy
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* of this software and associated documentation files (the "Software"), to deal
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* in the Software without restriction, including without limitation the rights
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* to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
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* copies of the Software, and to permit persons to whom the Software is
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* furnished to do so, subject to the following conditions:
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*
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* The above copyright notice and this permission notice shall be included in
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* all copies or substantial portions of the Software.
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*
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* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
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* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
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* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
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* THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
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* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
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* OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
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* THE SOFTWARE.
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*/
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#include "qemu/osdep.h"
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#include "qemu-common.h"
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#include "qemu/bswap.h"
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#include "qemu/error-report.h"
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#include "audio.h"
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#define AUDIO_CAP "mixeng"
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#include "audio_int.h"
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/* 8 bit */
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#define ENDIAN_CONVERSION natural
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#define ENDIAN_CONVERT(v) (v)
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/* Signed 8 bit */
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#define BSIZE 8
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#define ITYPE int
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#define IN_MIN SCHAR_MIN
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#define IN_MAX SCHAR_MAX
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#define SIGNED
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#define SHIFT 8
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#include "mixeng_template.h"
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#undef SIGNED
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#undef IN_MAX
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#undef IN_MIN
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#undef BSIZE
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#undef ITYPE
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#undef SHIFT
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/* Unsigned 8 bit */
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#define BSIZE 8
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#define ITYPE uint
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#define IN_MIN 0
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#define IN_MAX UCHAR_MAX
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#define SHIFT 8
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#include "mixeng_template.h"
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#undef IN_MAX
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#undef IN_MIN
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#undef BSIZE
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#undef ITYPE
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#undef SHIFT
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#undef ENDIAN_CONVERT
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#undef ENDIAN_CONVERSION
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/* Signed 16 bit */
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#define BSIZE 16
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#define ITYPE int
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#define IN_MIN SHRT_MIN
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#define IN_MAX SHRT_MAX
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#define SIGNED
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#define SHIFT 16
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#define ENDIAN_CONVERSION natural
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#define ENDIAN_CONVERT(v) (v)
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#include "mixeng_template.h"
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#undef ENDIAN_CONVERT
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#undef ENDIAN_CONVERSION
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#define ENDIAN_CONVERSION swap
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#define ENDIAN_CONVERT(v) bswap16 (v)
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#include "mixeng_template.h"
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#undef ENDIAN_CONVERT
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#undef ENDIAN_CONVERSION
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#undef SIGNED
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#undef IN_MAX
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#undef IN_MIN
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#undef BSIZE
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#undef ITYPE
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#undef SHIFT
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/* Unsigned 16 bit */
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#define BSIZE 16
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#define ITYPE uint
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#define IN_MIN 0
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#define IN_MAX USHRT_MAX
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#define SHIFT 16
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#define ENDIAN_CONVERSION natural
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#define ENDIAN_CONVERT(v) (v)
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#include "mixeng_template.h"
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#undef ENDIAN_CONVERT
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#undef ENDIAN_CONVERSION
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#define ENDIAN_CONVERSION swap
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#define ENDIAN_CONVERT(v) bswap16 (v)
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#include "mixeng_template.h"
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#undef ENDIAN_CONVERT
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#undef ENDIAN_CONVERSION
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#undef IN_MAX
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#undef IN_MIN
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#undef BSIZE
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#undef ITYPE
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#undef SHIFT
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/* Signed 32 bit */
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#define BSIZE 32
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#define ITYPE int
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#define IN_MIN INT32_MIN
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#define IN_MAX INT32_MAX
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#define SIGNED
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#define SHIFT 32
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#define ENDIAN_CONVERSION natural
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#define ENDIAN_CONVERT(v) (v)
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#include "mixeng_template.h"
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#undef ENDIAN_CONVERT
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#undef ENDIAN_CONVERSION
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#define ENDIAN_CONVERSION swap
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#define ENDIAN_CONVERT(v) bswap32 (v)
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#include "mixeng_template.h"
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#undef ENDIAN_CONVERT
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#undef ENDIAN_CONVERSION
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#undef SIGNED
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#undef IN_MAX
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#undef IN_MIN
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#undef BSIZE
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#undef ITYPE
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#undef SHIFT
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/* Unsigned 32 bit */
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#define BSIZE 32
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#define ITYPE uint
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#define IN_MIN 0
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#define IN_MAX UINT32_MAX
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#define SHIFT 32
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#define ENDIAN_CONVERSION natural
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#define ENDIAN_CONVERT(v) (v)
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#include "mixeng_template.h"
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#undef ENDIAN_CONVERT
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#undef ENDIAN_CONVERSION
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#define ENDIAN_CONVERSION swap
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#define ENDIAN_CONVERT(v) bswap32 (v)
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#include "mixeng_template.h"
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#undef ENDIAN_CONVERT
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#undef ENDIAN_CONVERSION
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#undef IN_MAX
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#undef IN_MIN
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#undef BSIZE
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#undef ITYPE
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#undef SHIFT
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t_sample *mixeng_conv[2][2][2][3] = {
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{
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{
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{
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conv_natural_uint8_t_to_mono,
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conv_natural_uint16_t_to_mono,
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conv_natural_uint32_t_to_mono
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},
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{
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conv_natural_uint8_t_to_mono,
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conv_swap_uint16_t_to_mono,
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conv_swap_uint32_t_to_mono,
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}
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},
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{
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{
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conv_natural_int8_t_to_mono,
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conv_natural_int16_t_to_mono,
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conv_natural_int32_t_to_mono
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},
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{
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conv_natural_int8_t_to_mono,
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conv_swap_int16_t_to_mono,
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conv_swap_int32_t_to_mono
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}
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}
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},
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{
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{
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{
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conv_natural_uint8_t_to_stereo,
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conv_natural_uint16_t_to_stereo,
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conv_natural_uint32_t_to_stereo
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},
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{
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conv_natural_uint8_t_to_stereo,
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conv_swap_uint16_t_to_stereo,
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conv_swap_uint32_t_to_stereo
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}
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},
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{
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{
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conv_natural_int8_t_to_stereo,
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conv_natural_int16_t_to_stereo,
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conv_natural_int32_t_to_stereo
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},
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{
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conv_natural_int8_t_to_stereo,
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conv_swap_int16_t_to_stereo,
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conv_swap_int32_t_to_stereo,
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}
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}
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}
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};
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f_sample *mixeng_clip[2][2][2][3] = {
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{
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{
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{
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clip_natural_uint8_t_from_mono,
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clip_natural_uint16_t_from_mono,
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clip_natural_uint32_t_from_mono
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},
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{
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clip_natural_uint8_t_from_mono,
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clip_swap_uint16_t_from_mono,
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clip_swap_uint32_t_from_mono
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}
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},
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{
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{
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clip_natural_int8_t_from_mono,
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clip_natural_int16_t_from_mono,
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clip_natural_int32_t_from_mono
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},
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{
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clip_natural_int8_t_from_mono,
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clip_swap_int16_t_from_mono,
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clip_swap_int32_t_from_mono
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}
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}
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},
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{
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{
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{
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clip_natural_uint8_t_from_stereo,
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clip_natural_uint16_t_from_stereo,
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clip_natural_uint32_t_from_stereo
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},
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{
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clip_natural_uint8_t_from_stereo,
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clip_swap_uint16_t_from_stereo,
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clip_swap_uint32_t_from_stereo
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}
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},
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{
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{
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clip_natural_int8_t_from_stereo,
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clip_natural_int16_t_from_stereo,
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clip_natural_int32_t_from_stereo
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},
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{
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clip_natural_int8_t_from_stereo,
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clip_swap_int16_t_from_stereo,
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clip_swap_int32_t_from_stereo
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}
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}
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}
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};
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void audio_sample_to_uint64(void *samples, int pos,
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uint64_t *left, uint64_t *right)
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{
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struct st_sample *sample = samples;
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sample += pos;
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#ifdef FLOAT_MIXENG
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error_report(
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"Coreaudio and floating point samples are not supported by replay yet");
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abort();
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#else
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*left = sample->l;
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*right = sample->r;
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#endif
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}
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void audio_sample_from_uint64(void *samples, int pos,
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uint64_t left, uint64_t right)
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{
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struct st_sample *sample = samples;
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sample += pos;
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#ifdef FLOAT_MIXENG
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error_report(
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"Coreaudio and floating point samples are not supported by replay yet");
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abort();
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#else
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sample->l = left;
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sample->r = right;
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#endif
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}
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/*
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* August 21, 1998
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* Copyright 1998 Fabrice Bellard.
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*
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* [Rewrote completely the code of Lance Norskog And Sundry
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* Contributors with a more efficient algorithm.]
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*
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* This source code is freely redistributable and may be used for
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* any purpose. This copyright notice must be maintained.
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* Lance Norskog And Sundry Contributors are not responsible for
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* the consequences of using this software.
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*/
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/*
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* Sound Tools rate change effect file.
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*/
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/*
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* Linear Interpolation.
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*
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* The use of fractional increment allows us to use no buffer. It
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* avoid the problems at the end of the buffer we had with the old
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* method which stored a possibly big buffer of size
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* lcm(in_rate,out_rate).
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*
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* Limited to 16 bit samples and sampling frequency <= 65535 Hz. If
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* the input & output frequencies are equal, a delay of one sample is
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* introduced. Limited to processing 32-bit count worth of samples.
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*
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* 1 << FRAC_BITS evaluating to zero in several places. Changed with
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* an (unsigned long) cast to make it safe. MarkMLl 2/1/99
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*/
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/* Private data */
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struct rate {
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uint64_t opos;
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uint64_t opos_inc;
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uint32_t ipos; /* position in the input stream (integer) */
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struct st_sample ilast; /* last sample in the input stream */
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};
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/*
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* Prepare processing.
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*/
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void *st_rate_start (int inrate, int outrate)
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{
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struct rate *rate = audio_calloc(__func__, 1, sizeof(*rate));
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if (!rate) {
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dolog ("Could not allocate resampler (%zu bytes)\n", sizeof (*rate));
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return NULL;
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}
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rate->opos = 0;
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/* increment */
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rate->opos_inc = ((uint64_t) inrate << 32) / outrate;
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rate->ipos = 0;
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rate->ilast.l = 0;
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rate->ilast.r = 0;
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return rate;
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}
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#define NAME st_rate_flow_mix
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#define OP(a, b) a += b
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#include "rate_template.h"
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#define NAME st_rate_flow
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#define OP(a, b) a = b
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#include "rate_template.h"
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void st_rate_stop (void *opaque)
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{
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g_free (opaque);
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}
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void mixeng_clear (struct st_sample *buf, int len)
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{
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memset (buf, 0, len * sizeof (struct st_sample));
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}
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void mixeng_volume (struct st_sample *buf, int len, struct mixeng_volume *vol)
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{
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if (vol->mute) {
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mixeng_clear (buf, len);
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return;
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}
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while (len--) {
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#ifdef FLOAT_MIXENG
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buf->l = buf->l * vol->l;
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buf->r = buf->r * vol->r;
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#else
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buf->l = (buf->l * vol->l) >> 32;
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buf->r = (buf->r * vol->r) >> 32;
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#endif
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buf += 1;
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}
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}
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