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NetBSD/sys/dev/mulaw.c

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Merge jmcneill-audiomp3 branch, which is derived from ad-audiomp2. From the original ad-audiomp branch notes: Add MP locking to the audio drivers. Making the audio drivers MP safe is necessary before efforts can be made to make the VM system MP safe. The are two locks per device instance, an ISR lock and a character device lock. The ISR lock replaces calls to splaudio()/splx(), and will be held across calls to device methods which were called at splaudio() before (e.g. trigger_output). The character device lock is held across calls to nearly all of the methods, excluding some only used for initialization, e.g. get_locks. Welcome to 5.99.57.
2011-11-24 03:07:28 +04:00
/* $NetBSD: mulaw.c,v 1.28 2011/11/23 23:07:31 jmcneill Exp $ */
RCS ID police.
1997-06-15 02:25:11 +04:00
GUS audio driver from John Kohl.
1995-07-19 23:58:09 +04:00
/*
* Copyright (c) 1991-1993 Regents of the University of California.
* 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.
* 3. All advertising materials mentioning features or use of this software
* must display the following acknowledgement:
* This product includes software developed by the Computer Systems
* Engineering Group at Lawrence Berkeley Laboratory.
* 4. Neither the name of the University nor of the Laboratory may be used
* to endorse or promote products derived from this software without
* specific prior written permission.
*
* THIS SOFTWARE IS PROVIDED BY THE REGENTS AND CONTRIBUTORS ``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 REGENTS OR CONTRIBUTORS 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.
*
*/
add RCSIDs
2001-11-13 08:32:49 +03:00
#include <sys/cdefs.h>
Merge jmcneill-audiomp3 branch, which is derived from ad-audiomp2. From the original ad-audiomp branch notes: Add MP locking to the audio drivers. Making the audio drivers MP safe is necessary before efforts can be made to make the VM system MP safe. The are two locks per device instance, an ISR lock and a character device lock. The ISR lock replaces calls to splaudio()/splx(), and will be held across calls to device methods which were called at splaudio() before (e.g. trigger_output). The character device lock is held across calls to nearly all of the methods, excluding some only used for initialization, e.g. get_locks. Welcome to 5.99.57.
2011-11-24 03:07:28 +04:00
__KERNEL_RCSID(0, "$NetBSD: mulaw.c,v 1.28 2011/11/23 23:07:31 jmcneill Exp $");
add RCSIDs
2001-11-13 08:32:49 +03:00
GUS audio driver from John Kohl.
1995-07-19 23:58:09 +04:00
#include <sys/types.h>
merge kent-audio1 branch, which introduces audio filter pipeline to the MI audio framework Summary of changes: * struct audio_params - remove sw_code, factor, factor_denom, hw_sample_rate, hw_encoding ,hw_precision, and hw_channels. Conversion information is conveyed by stream_filter_list_t. - change the type of sample_rate: u_long -> u_int - add `validbits,' which represents the valid data size in precision bits. It is required in order to distinguish 24/32bit from 24/24bit or 32/32bit. * audio_hw_if - add two parameters to set_params() stream_filter_list_t *pfil, stream_filter_list *rfil A HW driver should set filter recipes for requested formats - constify audio_params parameters of trigger_output() and trigger_input(). They represent audio formats for the hardware. - make open() and close() optional - add int (AUMODE_PLAY or AUMODE_RECORD) and audio_params_t parameters to round_blocksize() * sw_code is replaced with stream_filter_t. stream_filer_t converts audio data in an input buffer and writes into another output buffer unlike sw_code, which converts data in single buffer. converters in dev/auconv.c, dev/mulaw.c, dev/aurateconv.c, dev/tc/bba.c, dev/ic/msm6258.c, and arch/arm/iomd/vidcaudio.c are reimplemented as stream_filter_t * MI audio - audiosetinfo() builds filter pipelines from stream_filter_list_t filled by audio_hw_if::set_params() - audiosetinfo() returns with EINVAL if mmapped and set_params() requests filters - audio_write(), audio_pint(), and audio_rint() invoke a filter pipeline. - ioctl() for FIONREAD, AUDIO_WSEEK, AUDIO_GETIOFFS, AUDIO_GETOOFFS, and audio_prinfo::{seek,samples} for AUDIO_GETINFO handle values for a buffer nearest to userland. * add `struct device *' parameter to ac97_attach() * all of audio HW drivers follow audio_hw_if and ac97 changes
2005-01-11 01:01:36 +03:00
#include <sys/systm.h>
Split device_t and softc for audio(4), and other related cosmetic changes.
2008-03-04 21:23:44 +03:00
#include <sys/device.h>
merge kent-audio1 branch, which introduces audio filter pipeline to the MI audio framework Summary of changes: * struct audio_params - remove sw_code, factor, factor_denom, hw_sample_rate, hw_encoding ,hw_precision, and hw_channels. Conversion information is conveyed by stream_filter_list_t. - change the type of sample_rate: u_long -> u_int - add `validbits,' which represents the valid data size in precision bits. It is required in order to distinguish 24/32bit from 24/24bit or 32/32bit. * audio_hw_if - add two parameters to set_params() stream_filter_list_t *pfil, stream_filter_list *rfil A HW driver should set filter recipes for requested formats - constify audio_params parameters of trigger_output() and trigger_input(). They represent audio formats for the hardware. - make open() and close() optional - add int (AUMODE_PLAY or AUMODE_RECORD) and audio_params_t parameters to round_blocksize() * sw_code is replaced with stream_filter_t. stream_filer_t converts audio data in an input buffer and writes into another output buffer unlike sw_code, which converts data in single buffer. converters in dev/auconv.c, dev/mulaw.c, dev/aurateconv.c, dev/tc/bba.c, dev/ic/msm6258.c, and arch/arm/iomd/vidcaudio.c are reimplemented as stream_filter_t * MI audio - audiosetinfo() builds filter pipelines from stream_filter_list_t filled by audio_hw_if::set_params() - audiosetinfo() returns with EINVAL if mmapped and set_params() requests filters - audio_write(), audio_pint(), and audio_rint() invoke a filter pipeline. - ioctl() for FIONREAD, AUDIO_WSEEK, AUDIO_GETIOFFS, AUDIO_GETOOFFS, and audio_prinfo::{seek,samples} for AUDIO_GETINFO handle values for a buffer nearest to userland. * add `struct device *' parameter to ac97_attach() * all of audio HW drivers follow audio_hw_if and ac97 changes
2005-01-11 01:01:36 +03:00
#include <dev/auconv.h>
GUS audio driver from John Kohl.
1995-07-19 23:58:09 +04:00
#include <dev/mulaw.h>
merge kent-audio1 branch, which introduces audio filter pipeline to the MI audio framework Summary of changes: * struct audio_params - remove sw_code, factor, factor_denom, hw_sample_rate, hw_encoding ,hw_precision, and hw_channels. Conversion information is conveyed by stream_filter_list_t. - change the type of sample_rate: u_long -> u_int - add `validbits,' which represents the valid data size in precision bits. It is required in order to distinguish 24/32bit from 24/24bit or 32/32bit. * audio_hw_if - add two parameters to set_params() stream_filter_list_t *pfil, stream_filter_list *rfil A HW driver should set filter recipes for requested formats - constify audio_params parameters of trigger_output() and trigger_input(). They represent audio formats for the hardware. - make open() and close() optional - add int (AUMODE_PLAY or AUMODE_RECORD) and audio_params_t parameters to round_blocksize() * sw_code is replaced with stream_filter_t. stream_filer_t converts audio data in an input buffer and writes into another output buffer unlike sw_code, which converts data in single buffer. converters in dev/auconv.c, dev/mulaw.c, dev/aurateconv.c, dev/tc/bba.c, dev/ic/msm6258.c, and arch/arm/iomd/vidcaudio.c are reimplemented as stream_filter_t * MI audio - audiosetinfo() builds filter pipelines from stream_filter_list_t filled by audio_hw_if::set_params() - audiosetinfo() returns with EINVAL if mmapped and set_params() requests filters - audio_write(), audio_pint(), and audio_rint() invoke a filter pipeline. - ioctl() for FIONREAD, AUDIO_WSEEK, AUDIO_GETIOFFS, AUDIO_GETOOFFS, and audio_prinfo::{seek,samples} for AUDIO_GETINFO handle values for a buffer nearest to userland. * add `struct device *' parameter to ac97_attach() * all of audio HW drivers follow audio_hw_if and ac97 changes
2005-01-11 01:01:36 +03:00
/* #define MULAW_DEBUG */
#ifdef MULAW_DEBUG
# define DPRINTF(x) printf x
#else
# define DPRINTF(x)
#endif
Audio changes: - Change the way attach and open works to allow multiple audio devices. - Split the mulaw.c file into two to avoid dragging in mulaw convertsion when they are not needed. Add 16 bit alaw/mulaw tables. - Change the way audio properties are gotten. - Recognize more versions os SoundBlaster.
1997-08-01 02:33:08 +04:00
/*
Respect the law: Use A-law and mu-law as spellings as far as easily possible. Inspired by Igor Sobrado in PR 19680.
2003-04-06 22:20:07 +04:00
* This table converts a (8 bit) mu-law value to a 16 bit value.
fix typo in comment
2000-08-18 09:55:31 +04:00
* The 16 bits are represented as an array of two bytes for easier access
Audio changes: - Change the way attach and open works to allow multiple audio devices. - Split the mulaw.c file into two to avoid dragging in mulaw convertsion when they are not needed. Add 16 bit alaw/mulaw tables. - Change the way audio properties are gotten. - Recognize more versions os SoundBlaster.
1997-08-01 02:33:08 +04:00
* to the individual bytes.
*/
merge kent-audio1 branch, which introduces audio filter pipeline to the MI audio framework Summary of changes: * struct audio_params - remove sw_code, factor, factor_denom, hw_sample_rate, hw_encoding ,hw_precision, and hw_channels. Conversion information is conveyed by stream_filter_list_t. - change the type of sample_rate: u_long -> u_int - add `validbits,' which represents the valid data size in precision bits. It is required in order to distinguish 24/32bit from 24/24bit or 32/32bit. * audio_hw_if - add two parameters to set_params() stream_filter_list_t *pfil, stream_filter_list *rfil A HW driver should set filter recipes for requested formats - constify audio_params parameters of trigger_output() and trigger_input(). They represent audio formats for the hardware. - make open() and close() optional - add int (AUMODE_PLAY or AUMODE_RECORD) and audio_params_t parameters to round_blocksize() * sw_code is replaced with stream_filter_t. stream_filer_t converts audio data in an input buffer and writes into another output buffer unlike sw_code, which converts data in single buffer. converters in dev/auconv.c, dev/mulaw.c, dev/aurateconv.c, dev/tc/bba.c, dev/ic/msm6258.c, and arch/arm/iomd/vidcaudio.c are reimplemented as stream_filter_t * MI audio - audiosetinfo() builds filter pipelines from stream_filter_list_t filled by audio_hw_if::set_params() - audiosetinfo() returns with EINVAL if mmapped and set_params() requests filters - audio_write(), audio_pint(), and audio_rint() invoke a filter pipeline. - ioctl() for FIONREAD, AUDIO_WSEEK, AUDIO_GETIOFFS, AUDIO_GETOOFFS, and audio_prinfo::{seek,samples} for AUDIO_GETINFO handle values for a buffer nearest to userland. * add `struct device *' parameter to ac97_attach() * all of audio HW drivers follow audio_hw_if and ac97 changes
2005-01-11 01:01:36 +03:00
static const uint8_t mulawtolin16[256][2] = {
Audio changes: - Change the way attach and open works to allow multiple audio devices. - Split the mulaw.c file into two to avoid dragging in mulaw convertsion when they are not needed. Add 16 bit alaw/mulaw tables. - Change the way audio properties are gotten. - Recognize more versions os SoundBlaster.
1997-08-01 02:33:08 +04:00
{0x02,0x84}, {0x06,0x84}, {0x0a,0x84}, {0x0e,0x84},
{0x12,0x84}, {0x16,0x84}, {0x1a,0x84}, {0x1e,0x84},
{0x22,0x84}, {0x26,0x84}, {0x2a,0x84}, {0x2e,0x84},
{0x32,0x84}, {0x36,0x84}, {0x3a,0x84}, {0x3e,0x84},
{0x41,0x84}, {0x43,0x84}, {0x45,0x84}, {0x47,0x84},
{0x49,0x84}, {0x4b,0x84}, {0x4d,0x84}, {0x4f,0x84},
{0x51,0x84}, {0x53,0x84}, {0x55,0x84}, {0x57,0x84},
{0x59,0x84}, {0x5b,0x84}, {0x5d,0x84}, {0x5f,0x84},
{0x61,0x04}, {0x62,0x04}, {0x63,0x04}, {0x64,0x04},
{0x65,0x04}, {0x66,0x04}, {0x67,0x04}, {0x68,0x04},
{0x69,0x04}, {0x6a,0x04}, {0x6b,0x04}, {0x6c,0x04},
{0x6d,0x04}, {0x6e,0x04}, {0x6f,0x04}, {0x70,0x04},
{0x70,0xc4}, {0x71,0x44}, {0x71,0xc4}, {0x72,0x44},
{0x72,0xc4}, {0x73,0x44}, {0x73,0xc4}, {0x74,0x44},
{0x74,0xc4}, {0x75,0x44}, {0x75,0xc4}, {0x76,0x44},
{0x76,0xc4}, {0x77,0x44}, {0x77,0xc4}, {0x78,0x44},
{0x78,0xa4}, {0x78,0xe4}, {0x79,0x24}, {0x79,0x64},
{0x79,0xa4}, {0x79,0xe4}, {0x7a,0x24}, {0x7a,0x64},
{0x7a,0xa4}, {0x7a,0xe4}, {0x7b,0x24}, {0x7b,0x64},
{0x7b,0xa4}, {0x7b,0xe4}, {0x7c,0x24}, {0x7c,0x64},
{0x7c,0x94}, {0x7c,0xb4}, {0x7c,0xd4}, {0x7c,0xf4},
{0x7d,0x14}, {0x7d,0x34}, {0x7d,0x54}, {0x7d,0x74},
{0x7d,0x94}, {0x7d,0xb4}, {0x7d,0xd4}, {0x7d,0xf4},
{0x7e,0x14}, {0x7e,0x34}, {0x7e,0x54}, {0x7e,0x74},
{0x7e,0x8c}, {0x7e,0x9c}, {0x7e,0xac}, {0x7e,0xbc},
{0x7e,0xcc}, {0x7e,0xdc}, {0x7e,0xec}, {0x7e,0xfc},
{0x7f,0x0c}, {0x7f,0x1c}, {0x7f,0x2c}, {0x7f,0x3c},
{0x7f,0x4c}, {0x7f,0x5c}, {0x7f,0x6c}, {0x7f,0x7c},
{0x7f,0x88}, {0x7f,0x90}, {0x7f,0x98}, {0x7f,0xa0},
{0x7f,0xa8}, {0x7f,0xb0}, {0x7f,0xb8}, {0x7f,0xc0},
{0x7f,0xc8}, {0x7f,0xd0}, {0x7f,0xd8}, {0x7f,0xe0},
{0x7f,0xe8}, {0x7f,0xf0}, {0x7f,0xf8}, {0x80,0x00},
{0xfd,0x7c}, {0xf9,0x7c}, {0xf5,0x7c}, {0xf1,0x7c},
{0xed,0x7c}, {0xe9,0x7c}, {0xe5,0x7c}, {0xe1,0x7c},
{0xdd,0x7c}, {0xd9,0x7c}, {0xd5,0x7c}, {0xd1,0x7c},
{0xcd,0x7c}, {0xc9,0x7c}, {0xc5,0x7c}, {0xc1,0x7c},
{0xbe,0x7c}, {0xbc,0x7c}, {0xba,0x7c}, {0xb8,0x7c},
{0xb6,0x7c}, {0xb4,0x7c}, {0xb2,0x7c}, {0xb0,0x7c},
{0xae,0x7c}, {0xac,0x7c}, {0xaa,0x7c}, {0xa8,0x7c},
{0xa6,0x7c}, {0xa4,0x7c}, {0xa2,0x7c}, {0xa0,0x7c},
{0x9e,0xfc}, {0x9d,0xfc}, {0x9c,0xfc}, {0x9b,0xfc},
{0x9a,0xfc}, {0x99,0xfc}, {0x98,0xfc}, {0x97,0xfc},
{0x96,0xfc}, {0x95,0xfc}, {0x94,0xfc}, {0x93,0xfc},
{0x92,0xfc}, {0x91,0xfc}, {0x90,0xfc}, {0x8f,0xfc},
{0x8f,0x3c}, {0x8e,0xbc}, {0x8e,0x3c}, {0x8d,0xbc},
{0x8d,0x3c}, {0x8c,0xbc}, {0x8c,0x3c}, {0x8b,0xbc},
{0x8b,0x3c}, {0x8a,0xbc}, {0x8a,0x3c}, {0x89,0xbc},
{0x89,0x3c}, {0x88,0xbc}, {0x88,0x3c}, {0x87,0xbc},
{0x87,0x5c}, {0x87,0x1c}, {0x86,0xdc}, {0x86,0x9c},
{0x86,0x5c}, {0x86,0x1c}, {0x85,0xdc}, {0x85,0x9c},
{0x85,0x5c}, {0x85,0x1c}, {0x84,0xdc}, {0x84,0x9c},
{0x84,0x5c}, {0x84,0x1c}, {0x83,0xdc}, {0x83,0x9c},
{0x83,0x6c}, {0x83,0x4c}, {0x83,0x2c}, {0x83,0x0c},
{0x82,0xec}, {0x82,0xcc}, {0x82,0xac}, {0x82,0x8c},
{0x82,0x6c}, {0x82,0x4c}, {0x82,0x2c}, {0x82,0x0c},
{0x81,0xec}, {0x81,0xcc}, {0x81,0xac}, {0x81,0x8c},
{0x81,0x74}, {0x81,0x64}, {0x81,0x54}, {0x81,0x44},
{0x81,0x34}, {0x81,0x24}, {0x81,0x14}, {0x81,0x04},
{0x80,0xf4}, {0x80,0xe4}, {0x80,0xd4}, {0x80,0xc4},
{0x80,0xb4}, {0x80,0xa4}, {0x80,0x94}, {0x80,0x84},
{0x80,0x78}, {0x80,0x70}, {0x80,0x68}, {0x80,0x60},
{0x80,0x58}, {0x80,0x50}, {0x80,0x48}, {0x80,0x40},
{0x80,0x38}, {0x80,0x30}, {0x80,0x28}, {0x80,0x20},
{0x80,0x18}, {0x80,0x10}, {0x80,0x08}, {0x80,0x00},
GUS audio driver from John Kohl.
1995-07-19 23:58:09 +04:00
};
merge kent-audio1 branch, which introduces audio filter pipeline to the MI audio framework Summary of changes: * struct audio_params - remove sw_code, factor, factor_denom, hw_sample_rate, hw_encoding ,hw_precision, and hw_channels. Conversion information is conveyed by stream_filter_list_t. - change the type of sample_rate: u_long -> u_int - add `validbits,' which represents the valid data size in precision bits. It is required in order to distinguish 24/32bit from 24/24bit or 32/32bit. * audio_hw_if - add two parameters to set_params() stream_filter_list_t *pfil, stream_filter_list *rfil A HW driver should set filter recipes for requested formats - constify audio_params parameters of trigger_output() and trigger_input(). They represent audio formats for the hardware. - make open() and close() optional - add int (AUMODE_PLAY or AUMODE_RECORD) and audio_params_t parameters to round_blocksize() * sw_code is replaced with stream_filter_t. stream_filer_t converts audio data in an input buffer and writes into another output buffer unlike sw_code, which converts data in single buffer. converters in dev/auconv.c, dev/mulaw.c, dev/aurateconv.c, dev/tc/bba.c, dev/ic/msm6258.c, and arch/arm/iomd/vidcaudio.c are reimplemented as stream_filter_t * MI audio - audiosetinfo() builds filter pipelines from stream_filter_list_t filled by audio_hw_if::set_params() - audiosetinfo() returns with EINVAL if mmapped and set_params() requests filters - audio_write(), audio_pint(), and audio_rint() invoke a filter pipeline. - ioctl() for FIONREAD, AUDIO_WSEEK, AUDIO_GETIOFFS, AUDIO_GETOOFFS, and audio_prinfo::{seek,samples} for AUDIO_GETINFO handle values for a buffer nearest to userland. * add `struct device *' parameter to ac97_attach() * all of audio HW drivers follow audio_hw_if and ac97 changes
2005-01-11 01:01:36 +03:00
static const uint8_t lintomulaw[256] = {
Audio changes: - Change the way attach and open works to allow multiple audio devices. - Split the mulaw.c file into two to avoid dragging in mulaw convertsion when they are not needed. Add 16 bit alaw/mulaw tables. - Change the way audio properties are gotten. - Recognize more versions os SoundBlaster.
1997-08-01 02:33:08 +04:00
0x00, 0x00, 0x00, 0x00, 0x00, 0x01, 0x01, 0x01,
0x01, 0x02, 0x02, 0x02, 0x02, 0x03, 0x03, 0x03,
0x03, 0x04, 0x04, 0x04, 0x04, 0x05, 0x05, 0x05,
0x05, 0x06, 0x06, 0x06, 0x06, 0x07, 0x07, 0x07,
0x07, 0x08, 0x08, 0x08, 0x08, 0x09, 0x09, 0x09,
0x09, 0x0a, 0x0a, 0x0a, 0x0a, 0x0b, 0x0b, 0x0b,
0x0b, 0x0c, 0x0c, 0x0c, 0x0c, 0x0d, 0x0d, 0x0d,
0x0d, 0x0e, 0x0e, 0x0e, 0x0e, 0x0f, 0x0f, 0x0f,
0x0f, 0x10, 0x10, 0x11, 0x11, 0x12, 0x12, 0x13,
0x13, 0x14, 0x14, 0x15, 0x15, 0x16, 0x16, 0x17,
0x17, 0x18, 0x18, 0x19, 0x19, 0x1a, 0x1a, 0x1b,
0x1b, 0x1c, 0x1c, 0x1d, 0x1d, 0x1e, 0x1e, 0x1f,
0x1f, 0x20, 0x21, 0x22, 0x23, 0x24, 0x25, 0x26,
0x27, 0x28, 0x29, 0x2a, 0x2b, 0x2c, 0x2d, 0x2e,
0x2f, 0x30, 0x32, 0x34, 0x36, 0x38, 0x3a, 0x3c,
0x3e, 0x41, 0x45, 0x49, 0x4d, 0x53, 0x5b, 0x67,
0xff, 0xe7, 0xdb, 0xd3, 0xcd, 0xc9, 0xc5, 0xc1,
0xbe, 0xbc, 0xba, 0xb8, 0xb6, 0xb4, 0xb2, 0xb0,
0xaf, 0xae, 0xad, 0xac, 0xab, 0xaa, 0xa9, 0xa8,
0xa7, 0xa6, 0xa5, 0xa4, 0xa3, 0xa2, 0xa1, 0xa0,
0x9f, 0x9f, 0x9e, 0x9e, 0x9d, 0x9d, 0x9c, 0x9c,
0x9b, 0x9b, 0x9a, 0x9a, 0x99, 0x99, 0x98, 0x98,
0x97, 0x97, 0x96, 0x96, 0x95, 0x95, 0x94, 0x94,
0x93, 0x93, 0x92, 0x92, 0x91, 0x91, 0x90, 0x90,
0x8f, 0x8f, 0x8f, 0x8f, 0x8e, 0x8e, 0x8e, 0x8e,
0x8d, 0x8d, 0x8d, 0x8d, 0x8c, 0x8c, 0x8c, 0x8c,
0x8b, 0x8b, 0x8b, 0x8b, 0x8a, 0x8a, 0x8a, 0x8a,
0x89, 0x89, 0x89, 0x89, 0x88, 0x88, 0x88, 0x88,
0x87, 0x87, 0x87, 0x87, 0x86, 0x86, 0x86, 0x86,
0x85, 0x85, 0x85, 0x85, 0x84, 0x84, 0x84, 0x84,
0x83, 0x83, 0x83, 0x83, 0x82, 0x82, 0x82, 0x82,
0x81, 0x81, 0x81, 0x81, 0x80, 0x80, 0x80, 0x80,
GUS audio driver from John Kohl.
1995-07-19 23:58:09 +04:00
};
merge kent-audio1 branch, which introduces audio filter pipeline to the MI audio framework Summary of changes: * struct audio_params - remove sw_code, factor, factor_denom, hw_sample_rate, hw_encoding ,hw_precision, and hw_channels. Conversion information is conveyed by stream_filter_list_t. - change the type of sample_rate: u_long -> u_int - add `validbits,' which represents the valid data size in precision bits. It is required in order to distinguish 24/32bit from 24/24bit or 32/32bit. * audio_hw_if - add two parameters to set_params() stream_filter_list_t *pfil, stream_filter_list *rfil A HW driver should set filter recipes for requested formats - constify audio_params parameters of trigger_output() and trigger_input(). They represent audio formats for the hardware. - make open() and close() optional - add int (AUMODE_PLAY or AUMODE_RECORD) and audio_params_t parameters to round_blocksize() * sw_code is replaced with stream_filter_t. stream_filer_t converts audio data in an input buffer and writes into another output buffer unlike sw_code, which converts data in single buffer. converters in dev/auconv.c, dev/mulaw.c, dev/aurateconv.c, dev/tc/bba.c, dev/ic/msm6258.c, and arch/arm/iomd/vidcaudio.c are reimplemented as stream_filter_t * MI audio - audiosetinfo() builds filter pipelines from stream_filter_list_t filled by audio_hw_if::set_params() - audiosetinfo() returns with EINVAL if mmapped and set_params() requests filters - audio_write(), audio_pint(), and audio_rint() invoke a filter pipeline. - ioctl() for FIONREAD, AUDIO_WSEEK, AUDIO_GETIOFFS, AUDIO_GETOOFFS, and audio_prinfo::{seek,samples} for AUDIO_GETINFO handle values for a buffer nearest to userland. * add `struct device *' parameter to ac97_attach() * all of audio HW drivers follow audio_hw_if and ac97 changes
2005-01-11 01:01:36 +03:00
static const uint8_t alawtolin16[256][2] = {
Audio changes: - Change the way attach and open works to allow multiple audio devices. - Split the mulaw.c file into two to avoid dragging in mulaw convertsion when they are not needed. Add 16 bit alaw/mulaw tables. - Change the way audio properties are gotten. - Recognize more versions os SoundBlaster.
1997-08-01 02:33:08 +04:00
{0x6a,0x80}, {0x6b,0x80}, {0x68,0x80}, {0x69,0x80},
{0x6e,0x80}, {0x6f,0x80}, {0x6c,0x80}, {0x6d,0x80},
{0x62,0x80}, {0x63,0x80}, {0x60,0x80}, {0x61,0x80},
{0x66,0x80}, {0x67,0x80}, {0x64,0x80}, {0x65,0x80},
{0x75,0x40}, {0x75,0xc0}, {0x74,0x40}, {0x74,0xc0},
{0x77,0x40}, {0x77,0xc0}, {0x76,0x40}, {0x76,0xc0},
{0x71,0x40}, {0x71,0xc0}, {0x70,0x40}, {0x70,0xc0},
{0x73,0x40}, {0x73,0xc0}, {0x72,0x40}, {0x72,0xc0},
{0x2a,0x00}, {0x2e,0x00}, {0x22,0x00}, {0x26,0x00},
{0x3a,0x00}, {0x3e,0x00}, {0x32,0x00}, {0x36,0x00},
{0x0a,0x00}, {0x0e,0x00}, {0x02,0x00}, {0x06,0x00},
{0x1a,0x00}, {0x1e,0x00}, {0x12,0x00}, {0x16,0x00},
{0x55,0x00}, {0x57,0x00}, {0x51,0x00}, {0x53,0x00},
{0x5d,0x00}, {0x5f,0x00}, {0x59,0x00}, {0x5b,0x00},
{0x45,0x00}, {0x47,0x00}, {0x41,0x00}, {0x43,0x00},
{0x4d,0x00}, {0x4f,0x00}, {0x49,0x00}, {0x4b,0x00},
{0x7e,0xa8}, {0x7e,0xb8}, {0x7e,0x88}, {0x7e,0x98},
{0x7e,0xe8}, {0x7e,0xf8}, {0x7e,0xc8}, {0x7e,0xd8},
{0x7e,0x28}, {0x7e,0x38}, {0x7e,0x08}, {0x7e,0x18},
{0x7e,0x68}, {0x7e,0x78}, {0x7e,0x48}, {0x7e,0x58},
{0x7f,0xa8}, {0x7f,0xb8}, {0x7f,0x88}, {0x7f,0x98},
{0x7f,0xe8}, {0x7f,0xf8}, {0x7f,0xc8}, {0x7f,0xd8},
{0x7f,0x28}, {0x7f,0x38}, {0x7f,0x08}, {0x7f,0x18},
{0x7f,0x68}, {0x7f,0x78}, {0x7f,0x48}, {0x7f,0x58},
{0x7a,0xa0}, {0x7a,0xe0}, {0x7a,0x20}, {0x7a,0x60},
{0x7b,0xa0}, {0x7b,0xe0}, {0x7b,0x20}, {0x7b,0x60},
{0x78,0xa0}, {0x78,0xe0}, {0x78,0x20}, {0x78,0x60},
{0x79,0xa0}, {0x79,0xe0}, {0x79,0x20}, {0x79,0x60},
{0x7d,0x50}, {0x7d,0x70}, {0x7d,0x10}, {0x7d,0x30},
{0x7d,0xd0}, {0x7d,0xf0}, {0x7d,0x90}, {0x7d,0xb0},
{0x7c,0x50}, {0x7c,0x70}, {0x7c,0x10}, {0x7c,0x30},
{0x7c,0xd0}, {0x7c,0xf0}, {0x7c,0x90}, {0x7c,0xb0},
{0x95,0x80}, {0x94,0x80}, {0x97,0x80}, {0x96,0x80},
{0x91,0x80}, {0x90,0x80}, {0x93,0x80}, {0x92,0x80},
{0x9d,0x80}, {0x9c,0x80}, {0x9f,0x80}, {0x9e,0x80},
{0x99,0x80}, {0x98,0x80}, {0x9b,0x80}, {0x9a,0x80},
{0x8a,0xc0}, {0x8a,0x40}, {0x8b,0xc0}, {0x8b,0x40},
{0x88,0xc0}, {0x88,0x40}, {0x89,0xc0}, {0x89,0x40},
{0x8e,0xc0}, {0x8e,0x40}, {0x8f,0xc0}, {0x8f,0x40},
{0x8c,0xc0}, {0x8c,0x40}, {0x8d,0xc0}, {0x8d,0x40},
{0xd6,0x00}, {0xd2,0x00}, {0xde,0x00}, {0xda,0x00},
{0xc6,0x00}, {0xc2,0x00}, {0xce,0x00}, {0xca,0x00},
{0xf6,0x00}, {0xf2,0x00}, {0xfe,0x00}, {0xfa,0x00},
{0xe6,0x00}, {0xe2,0x00}, {0xee,0x00}, {0xea,0x00},
{0xab,0x00}, {0xa9,0x00}, {0xaf,0x00}, {0xad,0x00},
{0xa3,0x00}, {0xa1,0x00}, {0xa7,0x00}, {0xa5,0x00},
{0xbb,0x00}, {0xb9,0x00}, {0xbf,0x00}, {0xbd,0x00},
{0xb3,0x00}, {0xb1,0x00}, {0xb7,0x00}, {0xb5,0x00},
{0x81,0x58}, {0x81,0x48}, {0x81,0x78}, {0x81,0x68},
{0x81,0x18}, {0x81,0x08}, {0x81,0x38}, {0x81,0x28},
{0x81,0xd8}, {0x81,0xc8}, {0x81,0xf8}, {0x81,0xe8},
{0x81,0x98}, {0x81,0x88}, {0x81,0xb8}, {0x81,0xa8},
{0x80,0x58}, {0x80,0x48}, {0x80,0x78}, {0x80,0x68},
{0x80,0x18}, {0x80,0x08}, {0x80,0x38}, {0x80,0x28},
{0x80,0xd8}, {0x80,0xc8}, {0x80,0xf8}, {0x80,0xe8},
{0x80,0x98}, {0x80,0x88}, {0x80,0xb8}, {0x80,0xa8},
{0x85,0x60}, {0x85,0x20}, {0x85,0xe0}, {0x85,0xa0},
{0x84,0x60}, {0x84,0x20}, {0x84,0xe0}, {0x84,0xa0},
{0x87,0x60}, {0x87,0x20}, {0x87,0xe0}, {0x87,0xa0},
{0x86,0x60}, {0x86,0x20}, {0x86,0xe0}, {0x86,0xa0},
{0x82,0xb0}, {0x82,0x90}, {0x82,0xf0}, {0x82,0xd0},
{0x82,0x30}, {0x82,0x10}, {0x82,0x70}, {0x82,0x50},
{0x83,0xb0}, {0x83,0x90}, {0x83,0xf0}, {0x83,0xd0},
{0x83,0x30}, {0x83,0x10}, {0x83,0x70}, {0x83,0x50},
Add Alaw encoding emulation.
1997-05-28 04:07:46 +04:00
};
merge kent-audio1 branch, which introduces audio filter pipeline to the MI audio framework Summary of changes: * struct audio_params - remove sw_code, factor, factor_denom, hw_sample_rate, hw_encoding ,hw_precision, and hw_channels. Conversion information is conveyed by stream_filter_list_t. - change the type of sample_rate: u_long -> u_int - add `validbits,' which represents the valid data size in precision bits. It is required in order to distinguish 24/32bit from 24/24bit or 32/32bit. * audio_hw_if - add two parameters to set_params() stream_filter_list_t *pfil, stream_filter_list *rfil A HW driver should set filter recipes for requested formats - constify audio_params parameters of trigger_output() and trigger_input(). They represent audio formats for the hardware. - make open() and close() optional - add int (AUMODE_PLAY or AUMODE_RECORD) and audio_params_t parameters to round_blocksize() * sw_code is replaced with stream_filter_t. stream_filer_t converts audio data in an input buffer and writes into another output buffer unlike sw_code, which converts data in single buffer. converters in dev/auconv.c, dev/mulaw.c, dev/aurateconv.c, dev/tc/bba.c, dev/ic/msm6258.c, and arch/arm/iomd/vidcaudio.c are reimplemented as stream_filter_t * MI audio - audiosetinfo() builds filter pipelines from stream_filter_list_t filled by audio_hw_if::set_params() - audiosetinfo() returns with EINVAL if mmapped and set_params() requests filters - audio_write(), audio_pint(), and audio_rint() invoke a filter pipeline. - ioctl() for FIONREAD, AUDIO_WSEEK, AUDIO_GETIOFFS, AUDIO_GETOOFFS, and audio_prinfo::{seek,samples} for AUDIO_GETINFO handle values for a buffer nearest to userland. * add `struct device *' parameter to ac97_attach() * all of audio HW drivers follow audio_hw_if and ac97 changes
2005-01-11 01:01:36 +03:00
static const uint8_t lintoalaw[256] = {
Audio changes: - Change the way attach and open works to allow multiple audio devices. - Split the mulaw.c file into two to avoid dragging in mulaw convertsion when they are not needed. Add 16 bit alaw/mulaw tables. - Change the way audio properties are gotten. - Recognize more versions os SoundBlaster.
1997-08-01 02:33:08 +04:00
0x2a, 0x2a, 0x2a, 0x2a, 0x2b, 0x2b, 0x2b, 0x2b,
0x28, 0x28, 0x28, 0x28, 0x29, 0x29, 0x29, 0x29,
0x2e, 0x2e, 0x2e, 0x2e, 0x2f, 0x2f, 0x2f, 0x2f,
0x2c, 0x2c, 0x2c, 0x2c, 0x2d, 0x2d, 0x2d, 0x2d,
0x22, 0x22, 0x22, 0x22, 0x23, 0x23, 0x23, 0x23,
0x20, 0x20, 0x20, 0x20, 0x21, 0x21, 0x21, 0x21,
0x26, 0x26, 0x26, 0x26, 0x27, 0x27, 0x27, 0x27,
0x24, 0x24, 0x24, 0x24, 0x25, 0x25, 0x25, 0x25,
0x3a, 0x3a, 0x3b, 0x3b, 0x38, 0x38, 0x39, 0x39,
0x3e, 0x3e, 0x3f, 0x3f, 0x3c, 0x3c, 0x3d, 0x3d,
0x32, 0x32, 0x33, 0x33, 0x30, 0x30, 0x31, 0x31,
0x36, 0x36, 0x37, 0x37, 0x34, 0x34, 0x35, 0x35,
0x0a, 0x0b, 0x08, 0x09, 0x0e, 0x0f, 0x0c, 0x0d,
0x02, 0x03, 0x00, 0x01, 0x06, 0x07, 0x04, 0x05,
0x1a, 0x18, 0x1e, 0x1c, 0x12, 0x10, 0x16, 0x14,
0x6a, 0x6e, 0x62, 0x66, 0x7a, 0x72, 0x4a, 0x5a,
0xd5, 0xc5, 0xf5, 0xfd, 0xe5, 0xe1, 0xed, 0xe9,
0x95, 0x97, 0x91, 0x93, 0x9d, 0x9f, 0x99, 0x9b,
0x85, 0x84, 0x87, 0x86, 0x81, 0x80, 0x83, 0x82,
0x8d, 0x8c, 0x8f, 0x8e, 0x89, 0x88, 0x8b, 0x8a,
0xb5, 0xb5, 0xb4, 0xb4, 0xb7, 0xb7, 0xb6, 0xb6,
0xb1, 0xb1, 0xb0, 0xb0, 0xb3, 0xb3, 0xb2, 0xb2,
0xbd, 0xbd, 0xbc, 0xbc, 0xbf, 0xbf, 0xbe, 0xbe,
0xb9, 0xb9, 0xb8, 0xb8, 0xbb, 0xbb, 0xba, 0xba,
0xa5, 0xa5, 0xa5, 0xa5, 0xa4, 0xa4, 0xa4, 0xa4,
0xa7, 0xa7, 0xa7, 0xa7, 0xa6, 0xa6, 0xa6, 0xa6,
0xa1, 0xa1, 0xa1, 0xa1, 0xa0, 0xa0, 0xa0, 0xa0,
0xa3, 0xa3, 0xa3, 0xa3, 0xa2, 0xa2, 0xa2, 0xa2,
0xad, 0xad, 0xad, 0xad, 0xac, 0xac, 0xac, 0xac,
0xaf, 0xaf, 0xaf, 0xaf, 0xae, 0xae, 0xae, 0xae,
0xa9, 0xa9, 0xa9, 0xa9, 0xa8, 0xa8, 0xa8, 0xa8,
0xab, 0xab, 0xab, 0xab, 0xaa, 0xaa, 0xaa, 0xaa,
Add Alaw encoding emulation.
1997-05-28 04:07:46 +04:00
};
merge kent-audio1 branch, which introduces audio filter pipeline to the MI audio framework Summary of changes: * struct audio_params - remove sw_code, factor, factor_denom, hw_sample_rate, hw_encoding ,hw_precision, and hw_channels. Conversion information is conveyed by stream_filter_list_t. - change the type of sample_rate: u_long -> u_int - add `validbits,' which represents the valid data size in precision bits. It is required in order to distinguish 24/32bit from 24/24bit or 32/32bit. * audio_hw_if - add two parameters to set_params() stream_filter_list_t *pfil, stream_filter_list *rfil A HW driver should set filter recipes for requested formats - constify audio_params parameters of trigger_output() and trigger_input(). They represent audio formats for the hardware. - make open() and close() optional - add int (AUMODE_PLAY or AUMODE_RECORD) and audio_params_t parameters to round_blocksize() * sw_code is replaced with stream_filter_t. stream_filer_t converts audio data in an input buffer and writes into another output buffer unlike sw_code, which converts data in single buffer. converters in dev/auconv.c, dev/mulaw.c, dev/aurateconv.c, dev/tc/bba.c, dev/ic/msm6258.c, and arch/arm/iomd/vidcaudio.c are reimplemented as stream_filter_t * MI audio - audiosetinfo() builds filter pipelines from stream_filter_list_t filled by audio_hw_if::set_params() - audiosetinfo() returns with EINVAL if mmapped and set_params() requests filters - audio_write(), audio_pint(), and audio_rint() invoke a filter pipeline. - ioctl() for FIONREAD, AUDIO_WSEEK, AUDIO_GETIOFFS, AUDIO_GETOOFFS, and audio_prinfo::{seek,samples} for AUDIO_GETINFO handle values for a buffer nearest to userland. * add `struct device *' parameter to ac97_attach() * all of audio HW drivers follow audio_hw_if and ac97 changes
2005-01-11 01:01:36 +03:00
#define DEFINE_FILTER(name) \
static int \
Merge jmcneill-audiomp3 branch, which is derived from ad-audiomp2. From the original ad-audiomp branch notes: Add MP locking to the audio drivers. Making the audio drivers MP safe is necessary before efforts can be made to make the VM system MP safe. The are two locks per device instance, an ISR lock and a character device lock. The ISR lock replaces calls to splaudio()/splx(), and will be held across calls to device methods which were called at splaudio() before (e.g. trigger_output). The character device lock is held across calls to nearly all of the methods, excluding some only used for initialization, e.g. get_locks. Welcome to 5.99.57.
2011-11-24 03:07:28 +04:00
name##_fetch_to(struct audio_softc *, stream_fetcher_t *, audio_stream_t *, int); \
merge kent-audio1 branch, which introduces audio filter pipeline to the MI audio framework Summary of changes: * struct audio_params - remove sw_code, factor, factor_denom, hw_sample_rate, hw_encoding ,hw_precision, and hw_channels. Conversion information is conveyed by stream_filter_list_t. - change the type of sample_rate: u_long -> u_int - add `validbits,' which represents the valid data size in precision bits. It is required in order to distinguish 24/32bit from 24/24bit or 32/32bit. * audio_hw_if - add two parameters to set_params() stream_filter_list_t *pfil, stream_filter_list *rfil A HW driver should set filter recipes for requested formats - constify audio_params parameters of trigger_output() and trigger_input(). They represent audio formats for the hardware. - make open() and close() optional - add int (AUMODE_PLAY or AUMODE_RECORD) and audio_params_t parameters to round_blocksize() * sw_code is replaced with stream_filter_t. stream_filer_t converts audio data in an input buffer and writes into another output buffer unlike sw_code, which converts data in single buffer. converters in dev/auconv.c, dev/mulaw.c, dev/aurateconv.c, dev/tc/bba.c, dev/ic/msm6258.c, and arch/arm/iomd/vidcaudio.c are reimplemented as stream_filter_t * MI audio - audiosetinfo() builds filter pipelines from stream_filter_list_t filled by audio_hw_if::set_params() - audiosetinfo() returns with EINVAL if mmapped and set_params() requests filters - audio_write(), audio_pint(), and audio_rint() invoke a filter pipeline. - ioctl() for FIONREAD, AUDIO_WSEEK, AUDIO_GETIOFFS, AUDIO_GETOOFFS, and audio_prinfo::{seek,samples} for AUDIO_GETINFO handle values for a buffer nearest to userland. * add `struct device *' parameter to ac97_attach() * all of audio HW drivers follow audio_hw_if and ac97 changes
2005-01-11 01:01:36 +03:00
stream_filter_t * \
__unused removal on arguments; approved by core.
2006-11-16 04:32:37 +03:00
name(struct audio_softc *sc, const audio_params_t *from, \
const audio_params_t *to) \
merge kent-audio1 branch, which introduces audio filter pipeline to the MI audio framework Summary of changes: * struct audio_params - remove sw_code, factor, factor_denom, hw_sample_rate, hw_encoding ,hw_precision, and hw_channels. Conversion information is conveyed by stream_filter_list_t. - change the type of sample_rate: u_long -> u_int - add `validbits,' which represents the valid data size in precision bits. It is required in order to distinguish 24/32bit from 24/24bit or 32/32bit. * audio_hw_if - add two parameters to set_params() stream_filter_list_t *pfil, stream_filter_list *rfil A HW driver should set filter recipes for requested formats - constify audio_params parameters of trigger_output() and trigger_input(). They represent audio formats for the hardware. - make open() and close() optional - add int (AUMODE_PLAY or AUMODE_RECORD) and audio_params_t parameters to round_blocksize() * sw_code is replaced with stream_filter_t. stream_filer_t converts audio data in an input buffer and writes into another output buffer unlike sw_code, which converts data in single buffer. converters in dev/auconv.c, dev/mulaw.c, dev/aurateconv.c, dev/tc/bba.c, dev/ic/msm6258.c, and arch/arm/iomd/vidcaudio.c are reimplemented as stream_filter_t * MI audio - audiosetinfo() builds filter pipelines from stream_filter_list_t filled by audio_hw_if::set_params() - audiosetinfo() returns with EINVAL if mmapped and set_params() requests filters - audio_write(), audio_pint(), and audio_rint() invoke a filter pipeline. - ioctl() for FIONREAD, AUDIO_WSEEK, AUDIO_GETIOFFS, AUDIO_GETOOFFS, and audio_prinfo::{seek,samples} for AUDIO_GETINFO handle values for a buffer nearest to userland. * add `struct device *' parameter to ac97_attach() * all of audio HW drivers follow audio_hw_if and ac97 changes
2005-01-11 01:01:36 +03:00
{ \
DPRINTF(("Construct '%s' filter.\n", __func__)); \
return auconv_nocontext_filter_factory(name##_fetch_to); \
} \
static int \
Merge jmcneill-audiomp3 branch, which is derived from ad-audiomp2. From the original ad-audiomp branch notes: Add MP locking to the audio drivers. Making the audio drivers MP safe is necessary before efforts can be made to make the VM system MP safe. The are two locks per device instance, an ISR lock and a character device lock. The ISR lock replaces calls to splaudio()/splx(), and will be held across calls to device methods which were called at splaudio() before (e.g. trigger_output). The character device lock is held across calls to nearly all of the methods, excluding some only used for initialization, e.g. get_locks. Welcome to 5.99.57.
2011-11-24 03:07:28 +04:00
name##_fetch_to(struct audio_softc *sc, stream_fetcher_t *self, \
audio_stream_t *dst, int max_used)
merge kent-audio1 branch, which introduces audio filter pipeline to the MI audio framework Summary of changes: * struct audio_params - remove sw_code, factor, factor_denom, hw_sample_rate, hw_encoding ,hw_precision, and hw_channels. Conversion information is conveyed by stream_filter_list_t. - change the type of sample_rate: u_long -> u_int - add `validbits,' which represents the valid data size in precision bits. It is required in order to distinguish 24/32bit from 24/24bit or 32/32bit. * audio_hw_if - add two parameters to set_params() stream_filter_list_t *pfil, stream_filter_list *rfil A HW driver should set filter recipes for requested formats - constify audio_params parameters of trigger_output() and trigger_input(). They represent audio formats for the hardware. - make open() and close() optional - add int (AUMODE_PLAY or AUMODE_RECORD) and audio_params_t parameters to round_blocksize() * sw_code is replaced with stream_filter_t. stream_filer_t converts audio data in an input buffer and writes into another output buffer unlike sw_code, which converts data in single buffer. converters in dev/auconv.c, dev/mulaw.c, dev/aurateconv.c, dev/tc/bba.c, dev/ic/msm6258.c, and arch/arm/iomd/vidcaudio.c are reimplemented as stream_filter_t * MI audio - audiosetinfo() builds filter pipelines from stream_filter_list_t filled by audio_hw_if::set_params() - audiosetinfo() returns with EINVAL if mmapped and set_params() requests filters - audio_write(), audio_pint(), and audio_rint() invoke a filter pipeline. - ioctl() for FIONREAD, AUDIO_WSEEK, AUDIO_GETIOFFS, AUDIO_GETOOFFS, and audio_prinfo::{seek,samples} for AUDIO_GETINFO handle values for a buffer nearest to userland. * add `struct device *' parameter to ac97_attach() * all of audio HW drivers follow audio_hw_if and ac97 changes
2005-01-11 01:01:36 +03:00
DEFINE_FILTER(mulaw_to_linear8)
GUS audio driver from John Kohl.
1995-07-19 23:58:09 +04:00
{
merge kent-audio1 branch, which introduces audio filter pipeline to the MI audio framework Summary of changes: * struct audio_params - remove sw_code, factor, factor_denom, hw_sample_rate, hw_encoding ,hw_precision, and hw_channels. Conversion information is conveyed by stream_filter_list_t. - change the type of sample_rate: u_long -> u_int - add `validbits,' which represents the valid data size in precision bits. It is required in order to distinguish 24/32bit from 24/24bit or 32/32bit. * audio_hw_if - add two parameters to set_params() stream_filter_list_t *pfil, stream_filter_list *rfil A HW driver should set filter recipes for requested formats - constify audio_params parameters of trigger_output() and trigger_input(). They represent audio formats for the hardware. - make open() and close() optional - add int (AUMODE_PLAY or AUMODE_RECORD) and audio_params_t parameters to round_blocksize() * sw_code is replaced with stream_filter_t. stream_filer_t converts audio data in an input buffer and writes into another output buffer unlike sw_code, which converts data in single buffer. converters in dev/auconv.c, dev/mulaw.c, dev/aurateconv.c, dev/tc/bba.c, dev/ic/msm6258.c, and arch/arm/iomd/vidcaudio.c are reimplemented as stream_filter_t * MI audio - audiosetinfo() builds filter pipelines from stream_filter_list_t filled by audio_hw_if::set_params() - audiosetinfo() returns with EINVAL if mmapped and set_params() requests filters - audio_write(), audio_pint(), and audio_rint() invoke a filter pipeline. - ioctl() for FIONREAD, AUDIO_WSEEK, AUDIO_GETIOFFS, AUDIO_GETOOFFS, and audio_prinfo::{seek,samples} for AUDIO_GETINFO handle values for a buffer nearest to userland. * add `struct device *' parameter to ac97_attach() * all of audio HW drivers follow audio_hw_if and ac97 changes
2005-01-11 01:01:36 +03:00
stream_filter_t *this;
int m, err;
this = (stream_filter_t *)self;
Merge jmcneill-audiomp3 branch, which is derived from ad-audiomp2. From the original ad-audiomp branch notes: Add MP locking to the audio drivers. Making the audio drivers MP safe is necessary before efforts can be made to make the VM system MP safe. The are two locks per device instance, an ISR lock and a character device lock. The ISR lock replaces calls to splaudio()/splx(), and will be held across calls to device methods which were called at splaudio() before (e.g. trigger_output). The character device lock is held across calls to nearly all of the methods, excluding some only used for initialization, e.g. get_locks. Welcome to 5.99.57.
2011-11-24 03:07:28 +04:00
if ((err = this->prev->fetch_to(sc, this->prev, this->src, max_used)))
merge kent-audio1 branch, which introduces audio filter pipeline to the MI audio framework Summary of changes: * struct audio_params - remove sw_code, factor, factor_denom, hw_sample_rate, hw_encoding ,hw_precision, and hw_channels. Conversion information is conveyed by stream_filter_list_t. - change the type of sample_rate: u_long -> u_int - add `validbits,' which represents the valid data size in precision bits. It is required in order to distinguish 24/32bit from 24/24bit or 32/32bit. * audio_hw_if - add two parameters to set_params() stream_filter_list_t *pfil, stream_filter_list *rfil A HW driver should set filter recipes for requested formats - constify audio_params parameters of trigger_output() and trigger_input(). They represent audio formats for the hardware. - make open() and close() optional - add int (AUMODE_PLAY or AUMODE_RECORD) and audio_params_t parameters to round_blocksize() * sw_code is replaced with stream_filter_t. stream_filer_t converts audio data in an input buffer and writes into another output buffer unlike sw_code, which converts data in single buffer. converters in dev/auconv.c, dev/mulaw.c, dev/aurateconv.c, dev/tc/bba.c, dev/ic/msm6258.c, and arch/arm/iomd/vidcaudio.c are reimplemented as stream_filter_t * MI audio - audiosetinfo() builds filter pipelines from stream_filter_list_t filled by audio_hw_if::set_params() - audiosetinfo() returns with EINVAL if mmapped and set_params() requests filters - audio_write(), audio_pint(), and audio_rint() invoke a filter pipeline. - ioctl() for FIONREAD, AUDIO_WSEEK, AUDIO_GETIOFFS, AUDIO_GETOOFFS, and audio_prinfo::{seek,samples} for AUDIO_GETINFO handle values for a buffer nearest to userland. * add `struct device *' parameter to ac97_attach() * all of audio HW drivers follow audio_hw_if and ac97 changes
2005-01-11 01:01:36 +03:00
return err;
m = dst->end - dst->start;
m = min(m, max_used);
if (dst->param.encoding == AUDIO_ENCODING_ULINEAR_LE) {
FILTER_LOOP_PROLOGUE(this->src, 1, dst, 1, m) {
*d = mulawtolin16[*s][0];
} FILTER_LOOP_EPILOGUE(this->src, dst);
} else {
FILTER_LOOP_PROLOGUE(this->src, 1, dst, 1, m) {
*d = mulawtolin16[*s][0] ^ 0x80;
} FILTER_LOOP_EPILOGUE(this->src, dst);
GUS audio driver from John Kohl.
1995-07-19 23:58:09 +04:00
}
merge kent-audio1 branch, which introduces audio filter pipeline to the MI audio framework Summary of changes: * struct audio_params - remove sw_code, factor, factor_denom, hw_sample_rate, hw_encoding ,hw_precision, and hw_channels. Conversion information is conveyed by stream_filter_list_t. - change the type of sample_rate: u_long -> u_int - add `validbits,' which represents the valid data size in precision bits. It is required in order to distinguish 24/32bit from 24/24bit or 32/32bit. * audio_hw_if - add two parameters to set_params() stream_filter_list_t *pfil, stream_filter_list *rfil A HW driver should set filter recipes for requested formats - constify audio_params parameters of trigger_output() and trigger_input(). They represent audio formats for the hardware. - make open() and close() optional - add int (AUMODE_PLAY or AUMODE_RECORD) and audio_params_t parameters to round_blocksize() * sw_code is replaced with stream_filter_t. stream_filer_t converts audio data in an input buffer and writes into another output buffer unlike sw_code, which converts data in single buffer. converters in dev/auconv.c, dev/mulaw.c, dev/aurateconv.c, dev/tc/bba.c, dev/ic/msm6258.c, and arch/arm/iomd/vidcaudio.c are reimplemented as stream_filter_t * MI audio - audiosetinfo() builds filter pipelines from stream_filter_list_t filled by audio_hw_if::set_params() - audiosetinfo() returns with EINVAL if mmapped and set_params() requests filters - audio_write(), audio_pint(), and audio_rint() invoke a filter pipeline. - ioctl() for FIONREAD, AUDIO_WSEEK, AUDIO_GETIOFFS, AUDIO_GETOOFFS, and audio_prinfo::{seek,samples} for AUDIO_GETINFO handle values for a buffer nearest to userland. * add `struct device *' parameter to ac97_attach() * all of audio HW drivers follow audio_hw_if and ac97 changes
2005-01-11 01:01:36 +03:00
return 0;
Change the interface between high and lowlevel audio drivers again: Set the encoding parameters slightly differently. Remove the SW encoding/decodinf functions from this interface and move them to the audio_parameter struct; this is both more efficient and flexible.
1997-05-10 02:16:27 +04:00
}
GUS audio driver from John Kohl.
1995-07-19 23:58:09 +04:00
merge kent-audio1 branch, which introduces audio filter pipeline to the MI audio framework Summary of changes: * struct audio_params - remove sw_code, factor, factor_denom, hw_sample_rate, hw_encoding ,hw_precision, and hw_channels. Conversion information is conveyed by stream_filter_list_t. - change the type of sample_rate: u_long -> u_int - add `validbits,' which represents the valid data size in precision bits. It is required in order to distinguish 24/32bit from 24/24bit or 32/32bit. * audio_hw_if - add two parameters to set_params() stream_filter_list_t *pfil, stream_filter_list *rfil A HW driver should set filter recipes for requested formats - constify audio_params parameters of trigger_output() and trigger_input(). They represent audio formats for the hardware. - make open() and close() optional - add int (AUMODE_PLAY or AUMODE_RECORD) and audio_params_t parameters to round_blocksize() * sw_code is replaced with stream_filter_t. stream_filer_t converts audio data in an input buffer and writes into another output buffer unlike sw_code, which converts data in single buffer. converters in dev/auconv.c, dev/mulaw.c, dev/aurateconv.c, dev/tc/bba.c, dev/ic/msm6258.c, and arch/arm/iomd/vidcaudio.c are reimplemented as stream_filter_t * MI audio - audiosetinfo() builds filter pipelines from stream_filter_list_t filled by audio_hw_if::set_params() - audiosetinfo() returns with EINVAL if mmapped and set_params() requests filters - audio_write(), audio_pint(), and audio_rint() invoke a filter pipeline. - ioctl() for FIONREAD, AUDIO_WSEEK, AUDIO_GETIOFFS, AUDIO_GETOOFFS, and audio_prinfo::{seek,samples} for AUDIO_GETINFO handle values for a buffer nearest to userland. * add `struct device *' parameter to ac97_attach() * all of audio HW drivers follow audio_hw_if and ac97 changes
2005-01-11 01:01:36 +03:00
DEFINE_FILTER(mulaw_to_linear16)
Add alaw/mulaw conversion to/from 8 bit signed linear.
1998-05-22 17:05:31 +04:00
{
merge kent-audio1 branch, which introduces audio filter pipeline to the MI audio framework Summary of changes: * struct audio_params - remove sw_code, factor, factor_denom, hw_sample_rate, hw_encoding ,hw_precision, and hw_channels. Conversion information is conveyed by stream_filter_list_t. - change the type of sample_rate: u_long -> u_int - add `validbits,' which represents the valid data size in precision bits. It is required in order to distinguish 24/32bit from 24/24bit or 32/32bit. * audio_hw_if - add two parameters to set_params() stream_filter_list_t *pfil, stream_filter_list *rfil A HW driver should set filter recipes for requested formats - constify audio_params parameters of trigger_output() and trigger_input(). They represent audio formats for the hardware. - make open() and close() optional - add int (AUMODE_PLAY or AUMODE_RECORD) and audio_params_t parameters to round_blocksize() * sw_code is replaced with stream_filter_t. stream_filer_t converts audio data in an input buffer and writes into another output buffer unlike sw_code, which converts data in single buffer. converters in dev/auconv.c, dev/mulaw.c, dev/aurateconv.c, dev/tc/bba.c, dev/ic/msm6258.c, and arch/arm/iomd/vidcaudio.c are reimplemented as stream_filter_t * MI audio - audiosetinfo() builds filter pipelines from stream_filter_list_t filled by audio_hw_if::set_params() - audiosetinfo() returns with EINVAL if mmapped and set_params() requests filters - audio_write(), audio_pint(), and audio_rint() invoke a filter pipeline. - ioctl() for FIONREAD, AUDIO_WSEEK, AUDIO_GETIOFFS, AUDIO_GETOOFFS, and audio_prinfo::{seek,samples} for AUDIO_GETINFO handle values for a buffer nearest to userland. * add `struct device *' parameter to ac97_attach() * all of audio HW drivers follow audio_hw_if and ac97 changes
2005-01-11 01:01:36 +03:00
stream_filter_t *this;
int m, err;
this = (stream_filter_t *)self;
max_used = (max_used + 1) & ~1; /* round up to even */
Merge jmcneill-audiomp3 branch, which is derived from ad-audiomp2. From the original ad-audiomp branch notes: Add MP locking to the audio drivers. Making the audio drivers MP safe is necessary before efforts can be made to make the VM system MP safe. The are two locks per device instance, an ISR lock and a character device lock. The ISR lock replaces calls to splaudio()/splx(), and will be held across calls to device methods which were called at splaudio() before (e.g. trigger_output). The character device lock is held across calls to nearly all of the methods, excluding some only used for initialization, e.g. get_locks. Welcome to 5.99.57.
2011-11-24 03:07:28 +04:00
if ((err = this->prev->fetch_to(sc, this->prev, this->src, max_used / 2)))
merge kent-audio1 branch, which introduces audio filter pipeline to the MI audio framework Summary of changes: * struct audio_params - remove sw_code, factor, factor_denom, hw_sample_rate, hw_encoding ,hw_precision, and hw_channels. Conversion information is conveyed by stream_filter_list_t. - change the type of sample_rate: u_long -> u_int - add `validbits,' which represents the valid data size in precision bits. It is required in order to distinguish 24/32bit from 24/24bit or 32/32bit. * audio_hw_if - add two parameters to set_params() stream_filter_list_t *pfil, stream_filter_list *rfil A HW driver should set filter recipes for requested formats - constify audio_params parameters of trigger_output() and trigger_input(). They represent audio formats for the hardware. - make open() and close() optional - add int (AUMODE_PLAY or AUMODE_RECORD) and audio_params_t parameters to round_blocksize() * sw_code is replaced with stream_filter_t. stream_filer_t converts audio data in an input buffer and writes into another output buffer unlike sw_code, which converts data in single buffer. converters in dev/auconv.c, dev/mulaw.c, dev/aurateconv.c, dev/tc/bba.c, dev/ic/msm6258.c, and arch/arm/iomd/vidcaudio.c are reimplemented as stream_filter_t * MI audio - audiosetinfo() builds filter pipelines from stream_filter_list_t filled by audio_hw_if::set_params() - audiosetinfo() returns with EINVAL if mmapped and set_params() requests filters - audio_write(), audio_pint(), and audio_rint() invoke a filter pipeline. - ioctl() for FIONREAD, AUDIO_WSEEK, AUDIO_GETIOFFS, AUDIO_GETOOFFS, and audio_prinfo::{seek,samples} for AUDIO_GETINFO handle values for a buffer nearest to userland. * add `struct device *' parameter to ac97_attach() * all of audio HW drivers follow audio_hw_if and ac97 changes
2005-01-11 01:01:36 +03:00
return err;
m = (dst->end - dst->start) & ~1;
m = min(m, max_used);
switch (dst->param.encoding) {
case AUDIO_ENCODING_ULINEAR_LE:
FILTER_LOOP_PROLOGUE(this->src, 1, dst, 2, m) {
d[0] = mulawtolin16[s[0]][1];
d[1] = mulawtolin16[s[0]][0];
} FILTER_LOOP_EPILOGUE(this->src, dst);
break;
case AUDIO_ENCODING_ULINEAR_BE:
FILTER_LOOP_PROLOGUE(this->src, 1, dst, 2, m) {
d[0] = mulawtolin16[s[0]][0];
d[1] = mulawtolin16[s[0]][1];
} FILTER_LOOP_EPILOGUE(this->src, dst);
break;
case AUDIO_ENCODING_SLINEAR_LE:
FILTER_LOOP_PROLOGUE(this->src, 1, dst, 2, m) {
d[0] = mulawtolin16[s[0]][1];
d[1] = mulawtolin16[s[0]][0] ^ 0x80;
} FILTER_LOOP_EPILOGUE(this->src, dst);
break;
case AUDIO_ENCODING_SLINEAR_BE:
FILTER_LOOP_PROLOGUE(this->src, 1, dst, 2, m) {
d[0] = mulawtolin16[s[0]][0] ^ 0x80;
d[1] = mulawtolin16[s[0]][1];
} FILTER_LOOP_EPILOGUE(this->src, dst);
break;
Add alaw/mulaw conversion to/from 8 bit signed linear.
1998-05-22 17:05:31 +04:00
}
merge kent-audio1 branch, which introduces audio filter pipeline to the MI audio framework Summary of changes: * struct audio_params - remove sw_code, factor, factor_denom, hw_sample_rate, hw_encoding ,hw_precision, and hw_channels. Conversion information is conveyed by stream_filter_list_t. - change the type of sample_rate: u_long -> u_int - add `validbits,' which represents the valid data size in precision bits. It is required in order to distinguish 24/32bit from 24/24bit or 32/32bit. * audio_hw_if - add two parameters to set_params() stream_filter_list_t *pfil, stream_filter_list *rfil A HW driver should set filter recipes for requested formats - constify audio_params parameters of trigger_output() and trigger_input(). They represent audio formats for the hardware. - make open() and close() optional - add int (AUMODE_PLAY or AUMODE_RECORD) and audio_params_t parameters to round_blocksize() * sw_code is replaced with stream_filter_t. stream_filer_t converts audio data in an input buffer and writes into another output buffer unlike sw_code, which converts data in single buffer. converters in dev/auconv.c, dev/mulaw.c, dev/aurateconv.c, dev/tc/bba.c, dev/ic/msm6258.c, and arch/arm/iomd/vidcaudio.c are reimplemented as stream_filter_t * MI audio - audiosetinfo() builds filter pipelines from stream_filter_list_t filled by audio_hw_if::set_params() - audiosetinfo() returns with EINVAL if mmapped and set_params() requests filters - audio_write(), audio_pint(), and audio_rint() invoke a filter pipeline. - ioctl() for FIONREAD, AUDIO_WSEEK, AUDIO_GETIOFFS, AUDIO_GETOOFFS, and audio_prinfo::{seek,samples} for AUDIO_GETINFO handle values for a buffer nearest to userland. * add `struct device *' parameter to ac97_attach() * all of audio HW drivers follow audio_hw_if and ac97 changes
2005-01-11 01:01:36 +03:00
return 0;
Add alaw/mulaw conversion to/from 8 bit signed linear.
1998-05-22 17:05:31 +04:00
}
merge kent-audio1 branch, which introduces audio filter pipeline to the MI audio framework Summary of changes: * struct audio_params - remove sw_code, factor, factor_denom, hw_sample_rate, hw_encoding ,hw_precision, and hw_channels. Conversion information is conveyed by stream_filter_list_t. - change the type of sample_rate: u_long -> u_int - add `validbits,' which represents the valid data size in precision bits. It is required in order to distinguish 24/32bit from 24/24bit or 32/32bit. * audio_hw_if - add two parameters to set_params() stream_filter_list_t *pfil, stream_filter_list *rfil A HW driver should set filter recipes for requested formats - constify audio_params parameters of trigger_output() and trigger_input(). They represent audio formats for the hardware. - make open() and close() optional - add int (AUMODE_PLAY or AUMODE_RECORD) and audio_params_t parameters to round_blocksize() * sw_code is replaced with stream_filter_t. stream_filer_t converts audio data in an input buffer and writes into another output buffer unlike sw_code, which converts data in single buffer. converters in dev/auconv.c, dev/mulaw.c, dev/aurateconv.c, dev/tc/bba.c, dev/ic/msm6258.c, and arch/arm/iomd/vidcaudio.c are reimplemented as stream_filter_t * MI audio - audiosetinfo() builds filter pipelines from stream_filter_list_t filled by audio_hw_if::set_params() - audiosetinfo() returns with EINVAL if mmapped and set_params() requests filters - audio_write(), audio_pint(), and audio_rint() invoke a filter pipeline. - ioctl() for FIONREAD, AUDIO_WSEEK, AUDIO_GETIOFFS, AUDIO_GETOOFFS, and audio_prinfo::{seek,samples} for AUDIO_GETINFO handle values for a buffer nearest to userland. * add `struct device *' parameter to ac97_attach() * all of audio HW drivers follow audio_hw_if and ac97 changes
2005-01-11 01:01:36 +03:00
DEFINE_FILTER(linear16_to_mulaw)
Make little endian and big endian version of all functions that deal with 16 bit samples. Let the drivers decide which ones to use.
1999-11-01 21:12:19 +03:00
{
merge kent-audio1 branch, which introduces audio filter pipeline to the MI audio framework Summary of changes: * struct audio_params - remove sw_code, factor, factor_denom, hw_sample_rate, hw_encoding ,hw_precision, and hw_channels. Conversion information is conveyed by stream_filter_list_t. - change the type of sample_rate: u_long -> u_int - add `validbits,' which represents the valid data size in precision bits. It is required in order to distinguish 24/32bit from 24/24bit or 32/32bit. * audio_hw_if - add two parameters to set_params() stream_filter_list_t *pfil, stream_filter_list *rfil A HW driver should set filter recipes for requested formats - constify audio_params parameters of trigger_output() and trigger_input(). They represent audio formats for the hardware. - make open() and close() optional - add int (AUMODE_PLAY or AUMODE_RECORD) and audio_params_t parameters to round_blocksize() * sw_code is replaced with stream_filter_t. stream_filer_t converts audio data in an input buffer and writes into another output buffer unlike sw_code, which converts data in single buffer. converters in dev/auconv.c, dev/mulaw.c, dev/aurateconv.c, dev/tc/bba.c, dev/ic/msm6258.c, and arch/arm/iomd/vidcaudio.c are reimplemented as stream_filter_t * MI audio - audiosetinfo() builds filter pipelines from stream_filter_list_t filled by audio_hw_if::set_params() - audiosetinfo() returns with EINVAL if mmapped and set_params() requests filters - audio_write(), audio_pint(), and audio_rint() invoke a filter pipeline. - ioctl() for FIONREAD, AUDIO_WSEEK, AUDIO_GETIOFFS, AUDIO_GETOOFFS, and audio_prinfo::{seek,samples} for AUDIO_GETINFO handle values for a buffer nearest to userland. * add `struct device *' parameter to ac97_attach() * all of audio HW drivers follow audio_hw_if and ac97 changes
2005-01-11 01:01:36 +03:00
stream_filter_t *this;
int m, err;
this = (stream_filter_t *)self;
Merge jmcneill-audiomp3 branch, which is derived from ad-audiomp2. From the original ad-audiomp branch notes: Add MP locking to the audio drivers. Making the audio drivers MP safe is necessary before efforts can be made to make the VM system MP safe. The are two locks per device instance, an ISR lock and a character device lock. The ISR lock replaces calls to splaudio()/splx(), and will be held across calls to device methods which were called at splaudio() before (e.g. trigger_output). The character device lock is held across calls to nearly all of the methods, excluding some only used for initialization, e.g. get_locks. Welcome to 5.99.57.
2011-11-24 03:07:28 +04:00
if ((err = this->prev->fetch_to(sc, this->prev, this->src, max_used * 2)))
merge kent-audio1 branch, which introduces audio filter pipeline to the MI audio framework Summary of changes: * struct audio_params - remove sw_code, factor, factor_denom, hw_sample_rate, hw_encoding ,hw_precision, and hw_channels. Conversion information is conveyed by stream_filter_list_t. - change the type of sample_rate: u_long -> u_int - add `validbits,' which represents the valid data size in precision bits. It is required in order to distinguish 24/32bit from 24/24bit or 32/32bit. * audio_hw_if - add two parameters to set_params() stream_filter_list_t *pfil, stream_filter_list *rfil A HW driver should set filter recipes for requested formats - constify audio_params parameters of trigger_output() and trigger_input(). They represent audio formats for the hardware. - make open() and close() optional - add int (AUMODE_PLAY or AUMODE_RECORD) and audio_params_t parameters to round_blocksize() * sw_code is replaced with stream_filter_t. stream_filer_t converts audio data in an input buffer and writes into another output buffer unlike sw_code, which converts data in single buffer. converters in dev/auconv.c, dev/mulaw.c, dev/aurateconv.c, dev/tc/bba.c, dev/ic/msm6258.c, and arch/arm/iomd/vidcaudio.c are reimplemented as stream_filter_t * MI audio - audiosetinfo() builds filter pipelines from stream_filter_list_t filled by audio_hw_if::set_params() - audiosetinfo() returns with EINVAL if mmapped and set_params() requests filters - audio_write(), audio_pint(), and audio_rint() invoke a filter pipeline. - ioctl() for FIONREAD, AUDIO_WSEEK, AUDIO_GETIOFFS, AUDIO_GETOOFFS, and audio_prinfo::{seek,samples} for AUDIO_GETINFO handle values for a buffer nearest to userland. * add `struct device *' parameter to ac97_attach() * all of audio HW drivers follow audio_hw_if and ac97 changes
2005-01-11 01:01:36 +03:00
return err;
m = dst->end - dst->start;
m = min(m, max_used);
switch (this->src->param.encoding) {
case AUDIO_ENCODING_SLINEAR_LE:
FILTER_LOOP_PROLOGUE(this->src, 2, dst, 1, m) {
d[0] = lintomulaw[s[1] ^ 0x80];
} FILTER_LOOP_EPILOGUE(this->src, dst);
break;
case AUDIO_ENCODING_SLINEAR_BE:
FILTER_LOOP_PROLOGUE(this->src, 2, dst, 1, m) {
d[0] = lintomulaw[s[0] ^ 0x80];
} FILTER_LOOP_EPILOGUE(this->src, dst);
break;
case AUDIO_ENCODING_ULINEAR_LE:
FILTER_LOOP_PROLOGUE(this->src, 2, dst, 1, m) {
d[0] = lintomulaw[s[1]];
} FILTER_LOOP_EPILOGUE(this->src, dst);
break;
case AUDIO_ENCODING_ULINEAR_BE:
FILTER_LOOP_PROLOGUE(this->src, 2, dst, 1, m) {
d[0] = lintomulaw[s[0]];
} FILTER_LOOP_EPILOGUE(this->src, dst);
break;
Make little endian and big endian version of all functions that deal with 16 bit samples. Let the drivers decide which ones to use.
1999-11-01 21:12:19 +03:00
}
merge kent-audio1 branch, which introduces audio filter pipeline to the MI audio framework Summary of changes: * struct audio_params - remove sw_code, factor, factor_denom, hw_sample_rate, hw_encoding ,hw_precision, and hw_channels. Conversion information is conveyed by stream_filter_list_t. - change the type of sample_rate: u_long -> u_int - add `validbits,' which represents the valid data size in precision bits. It is required in order to distinguish 24/32bit from 24/24bit or 32/32bit. * audio_hw_if - add two parameters to set_params() stream_filter_list_t *pfil, stream_filter_list *rfil A HW driver should set filter recipes for requested formats - constify audio_params parameters of trigger_output() and trigger_input(). They represent audio formats for the hardware. - make open() and close() optional - add int (AUMODE_PLAY or AUMODE_RECORD) and audio_params_t parameters to round_blocksize() * sw_code is replaced with stream_filter_t. stream_filer_t converts audio data in an input buffer and writes into another output buffer unlike sw_code, which converts data in single buffer. converters in dev/auconv.c, dev/mulaw.c, dev/aurateconv.c, dev/tc/bba.c, dev/ic/msm6258.c, and arch/arm/iomd/vidcaudio.c are reimplemented as stream_filter_t * MI audio - audiosetinfo() builds filter pipelines from stream_filter_list_t filled by audio_hw_if::set_params() - audiosetinfo() returns with EINVAL if mmapped and set_params() requests filters - audio_write(), audio_pint(), and audio_rint() invoke a filter pipeline. - ioctl() for FIONREAD, AUDIO_WSEEK, AUDIO_GETIOFFS, AUDIO_GETOOFFS, and audio_prinfo::{seek,samples} for AUDIO_GETINFO handle values for a buffer nearest to userland. * add `struct device *' parameter to ac97_attach() * all of audio HW drivers follow audio_hw_if and ac97 changes
2005-01-11 01:01:36 +03:00
return 0;
Make little endian and big endian version of all functions that deal with 16 bit samples. Let the drivers decide which ones to use.
1999-11-01 21:12:19 +03:00
}
merge kent-audio1 branch, which introduces audio filter pipeline to the MI audio framework Summary of changes: * struct audio_params - remove sw_code, factor, factor_denom, hw_sample_rate, hw_encoding ,hw_precision, and hw_channels. Conversion information is conveyed by stream_filter_list_t. - change the type of sample_rate: u_long -> u_int - add `validbits,' which represents the valid data size in precision bits. It is required in order to distinguish 24/32bit from 24/24bit or 32/32bit. * audio_hw_if - add two parameters to set_params() stream_filter_list_t *pfil, stream_filter_list *rfil A HW driver should set filter recipes for requested formats - constify audio_params parameters of trigger_output() and trigger_input(). They represent audio formats for the hardware. - make open() and close() optional - add int (AUMODE_PLAY or AUMODE_RECORD) and audio_params_t parameters to round_blocksize() * sw_code is replaced with stream_filter_t. stream_filer_t converts audio data in an input buffer and writes into another output buffer unlike sw_code, which converts data in single buffer. converters in dev/auconv.c, dev/mulaw.c, dev/aurateconv.c, dev/tc/bba.c, dev/ic/msm6258.c, and arch/arm/iomd/vidcaudio.c are reimplemented as stream_filter_t * MI audio - audiosetinfo() builds filter pipelines from stream_filter_list_t filled by audio_hw_if::set_params() - audiosetinfo() returns with EINVAL if mmapped and set_params() requests filters - audio_write(), audio_pint(), and audio_rint() invoke a filter pipeline. - ioctl() for FIONREAD, AUDIO_WSEEK, AUDIO_GETIOFFS, AUDIO_GETOOFFS, and audio_prinfo::{seek,samples} for AUDIO_GETINFO handle values for a buffer nearest to userland. * add `struct device *' parameter to ac97_attach() * all of audio HW drivers follow audio_hw_if and ac97 changes
2005-01-11 01:01:36 +03:00
DEFINE_FILTER(linear8_to_mulaw)
Make little endian and big endian version of all functions that deal with 16 bit samples. Let the drivers decide which ones to use.
1999-11-01 21:12:19 +03:00
{
merge kent-audio1 branch, which introduces audio filter pipeline to the MI audio framework Summary of changes: * struct audio_params - remove sw_code, factor, factor_denom, hw_sample_rate, hw_encoding ,hw_precision, and hw_channels. Conversion information is conveyed by stream_filter_list_t. - change the type of sample_rate: u_long -> u_int - add `validbits,' which represents the valid data size in precision bits. It is required in order to distinguish 24/32bit from 24/24bit or 32/32bit. * audio_hw_if - add two parameters to set_params() stream_filter_list_t *pfil, stream_filter_list *rfil A HW driver should set filter recipes for requested formats - constify audio_params parameters of trigger_output() and trigger_input(). They represent audio formats for the hardware. - make open() and close() optional - add int (AUMODE_PLAY or AUMODE_RECORD) and audio_params_t parameters to round_blocksize() * sw_code is replaced with stream_filter_t. stream_filer_t converts audio data in an input buffer and writes into another output buffer unlike sw_code, which converts data in single buffer. converters in dev/auconv.c, dev/mulaw.c, dev/aurateconv.c, dev/tc/bba.c, dev/ic/msm6258.c, and arch/arm/iomd/vidcaudio.c are reimplemented as stream_filter_t * MI audio - audiosetinfo() builds filter pipelines from stream_filter_list_t filled by audio_hw_if::set_params() - audiosetinfo() returns with EINVAL if mmapped and set_params() requests filters - audio_write(), audio_pint(), and audio_rint() invoke a filter pipeline. - ioctl() for FIONREAD, AUDIO_WSEEK, AUDIO_GETIOFFS, AUDIO_GETOOFFS, and audio_prinfo::{seek,samples} for AUDIO_GETINFO handle values for a buffer nearest to userland. * add `struct device *' parameter to ac97_attach() * all of audio HW drivers follow audio_hw_if and ac97 changes
2005-01-11 01:01:36 +03:00
stream_filter_t *this;
int m, err;
this = (stream_filter_t *)self;
Merge jmcneill-audiomp3 branch, which is derived from ad-audiomp2. From the original ad-audiomp branch notes: Add MP locking to the audio drivers. Making the audio drivers MP safe is necessary before efforts can be made to make the VM system MP safe. The are two locks per device instance, an ISR lock and a character device lock. The ISR lock replaces calls to splaudio()/splx(), and will be held across calls to device methods which were called at splaudio() before (e.g. trigger_output). The character device lock is held across calls to nearly all of the methods, excluding some only used for initialization, e.g. get_locks. Welcome to 5.99.57.
2011-11-24 03:07:28 +04:00
if ((err = this->prev->fetch_to(sc, this->prev, this->src, max_used)))
merge kent-audio1 branch, which introduces audio filter pipeline to the MI audio framework Summary of changes: * struct audio_params - remove sw_code, factor, factor_denom, hw_sample_rate, hw_encoding ,hw_precision, and hw_channels. Conversion information is conveyed by stream_filter_list_t. - change the type of sample_rate: u_long -> u_int - add `validbits,' which represents the valid data size in precision bits. It is required in order to distinguish 24/32bit from 24/24bit or 32/32bit. * audio_hw_if - add two parameters to set_params() stream_filter_list_t *pfil, stream_filter_list *rfil A HW driver should set filter recipes for requested formats - constify audio_params parameters of trigger_output() and trigger_input(). They represent audio formats for the hardware. - make open() and close() optional - add int (AUMODE_PLAY or AUMODE_RECORD) and audio_params_t parameters to round_blocksize() * sw_code is replaced with stream_filter_t. stream_filer_t converts audio data in an input buffer and writes into another output buffer unlike sw_code, which converts data in single buffer. converters in dev/auconv.c, dev/mulaw.c, dev/aurateconv.c, dev/tc/bba.c, dev/ic/msm6258.c, and arch/arm/iomd/vidcaudio.c are reimplemented as stream_filter_t * MI audio - audiosetinfo() builds filter pipelines from stream_filter_list_t filled by audio_hw_if::set_params() - audiosetinfo() returns with EINVAL if mmapped and set_params() requests filters - audio_write(), audio_pint(), and audio_rint() invoke a filter pipeline. - ioctl() for FIONREAD, AUDIO_WSEEK, AUDIO_GETIOFFS, AUDIO_GETOOFFS, and audio_prinfo::{seek,samples} for AUDIO_GETINFO handle values for a buffer nearest to userland. * add `struct device *' parameter to ac97_attach() * all of audio HW drivers follow audio_hw_if and ac97 changes
2005-01-11 01:01:36 +03:00
return err;
m = dst->end - dst->start;
m = min(m, max_used);
if (this->src->param.encoding == AUDIO_ENCODING_ULINEAR_LE) {
FILTER_LOOP_PROLOGUE(this->src, 1, dst, 1, m) {
*d = lintomulaw[*s];
} FILTER_LOOP_EPILOGUE(this->src, dst);
} else { /* SLINEAR_LE */
FILTER_LOOP_PROLOGUE(this->src, 1, dst, 1, m) {
*d = lintomulaw[*s ^ 0x80];
} FILTER_LOOP_EPILOGUE(this->src, dst);
Make little endian and big endian version of all functions that deal with 16 bit samples. Let the drivers decide which ones to use.
1999-11-01 21:12:19 +03:00
}
merge kent-audio1 branch, which introduces audio filter pipeline to the MI audio framework Summary of changes: * struct audio_params - remove sw_code, factor, factor_denom, hw_sample_rate, hw_encoding ,hw_precision, and hw_channels. Conversion information is conveyed by stream_filter_list_t. - change the type of sample_rate: u_long -> u_int - add `validbits,' which represents the valid data size in precision bits. It is required in order to distinguish 24/32bit from 24/24bit or 32/32bit. * audio_hw_if - add two parameters to set_params() stream_filter_list_t *pfil, stream_filter_list *rfil A HW driver should set filter recipes for requested formats - constify audio_params parameters of trigger_output() and trigger_input(). They represent audio formats for the hardware. - make open() and close() optional - add int (AUMODE_PLAY or AUMODE_RECORD) and audio_params_t parameters to round_blocksize() * sw_code is replaced with stream_filter_t. stream_filer_t converts audio data in an input buffer and writes into another output buffer unlike sw_code, which converts data in single buffer. converters in dev/auconv.c, dev/mulaw.c, dev/aurateconv.c, dev/tc/bba.c, dev/ic/msm6258.c, and arch/arm/iomd/vidcaudio.c are reimplemented as stream_filter_t * MI audio - audiosetinfo() builds filter pipelines from stream_filter_list_t filled by audio_hw_if::set_params() - audiosetinfo() returns with EINVAL if mmapped and set_params() requests filters - audio_write(), audio_pint(), and audio_rint() invoke a filter pipeline. - ioctl() for FIONREAD, AUDIO_WSEEK, AUDIO_GETIOFFS, AUDIO_GETOOFFS, and audio_prinfo::{seek,samples} for AUDIO_GETINFO handle values for a buffer nearest to userland. * add `struct device *' parameter to ac97_attach() * all of audio HW drivers follow audio_hw_if and ac97 changes
2005-01-11 01:01:36 +03:00
return 0;
Make little endian and big endian version of all functions that deal with 16 bit samples. Let the drivers decide which ones to use.
1999-11-01 21:12:19 +03:00
}
merge kent-audio1 branch, which introduces audio filter pipeline to the MI audio framework Summary of changes: * struct audio_params - remove sw_code, factor, factor_denom, hw_sample_rate, hw_encoding ,hw_precision, and hw_channels. Conversion information is conveyed by stream_filter_list_t. - change the type of sample_rate: u_long -> u_int - add `validbits,' which represents the valid data size in precision bits. It is required in order to distinguish 24/32bit from 24/24bit or 32/32bit. * audio_hw_if - add two parameters to set_params() stream_filter_list_t *pfil, stream_filter_list *rfil A HW driver should set filter recipes for requested formats - constify audio_params parameters of trigger_output() and trigger_input(). They represent audio formats for the hardware. - make open() and close() optional - add int (AUMODE_PLAY or AUMODE_RECORD) and audio_params_t parameters to round_blocksize() * sw_code is replaced with stream_filter_t. stream_filer_t converts audio data in an input buffer and writes into another output buffer unlike sw_code, which converts data in single buffer. converters in dev/auconv.c, dev/mulaw.c, dev/aurateconv.c, dev/tc/bba.c, dev/ic/msm6258.c, and arch/arm/iomd/vidcaudio.c are reimplemented as stream_filter_t * MI audio - audiosetinfo() builds filter pipelines from stream_filter_list_t filled by audio_hw_if::set_params() - audiosetinfo() returns with EINVAL if mmapped and set_params() requests filters - audio_write(), audio_pint(), and audio_rint() invoke a filter pipeline. - ioctl() for FIONREAD, AUDIO_WSEEK, AUDIO_GETIOFFS, AUDIO_GETOOFFS, and audio_prinfo::{seek,samples} for AUDIO_GETINFO handle values for a buffer nearest to userland. * add `struct device *' parameter to ac97_attach() * all of audio HW drivers follow audio_hw_if and ac97 changes
2005-01-11 01:01:36 +03:00
DEFINE_FILTER(alaw_to_linear8)
Change the interface between high and lowlevel audio drivers again: Set the encoding parameters slightly differently. Remove the SW encoding/decodinf functions from this interface and move them to the audio_parameter struct; this is both more efficient and flexible.
1997-05-10 02:16:27 +04:00
{
merge kent-audio1 branch, which introduces audio filter pipeline to the MI audio framework Summary of changes: * struct audio_params - remove sw_code, factor, factor_denom, hw_sample_rate, hw_encoding ,hw_precision, and hw_channels. Conversion information is conveyed by stream_filter_list_t. - change the type of sample_rate: u_long -> u_int - add `validbits,' which represents the valid data size in precision bits. It is required in order to distinguish 24/32bit from 24/24bit or 32/32bit. * audio_hw_if - add two parameters to set_params() stream_filter_list_t *pfil, stream_filter_list *rfil A HW driver should set filter recipes for requested formats - constify audio_params parameters of trigger_output() and trigger_input(). They represent audio formats for the hardware. - make open() and close() optional - add int (AUMODE_PLAY or AUMODE_RECORD) and audio_params_t parameters to round_blocksize() * sw_code is replaced with stream_filter_t. stream_filer_t converts audio data in an input buffer and writes into another output buffer unlike sw_code, which converts data in single buffer. converters in dev/auconv.c, dev/mulaw.c, dev/aurateconv.c, dev/tc/bba.c, dev/ic/msm6258.c, and arch/arm/iomd/vidcaudio.c are reimplemented as stream_filter_t * MI audio - audiosetinfo() builds filter pipelines from stream_filter_list_t filled by audio_hw_if::set_params() - audiosetinfo() returns with EINVAL if mmapped and set_params() requests filters - audio_write(), audio_pint(), and audio_rint() invoke a filter pipeline. - ioctl() for FIONREAD, AUDIO_WSEEK, AUDIO_GETIOFFS, AUDIO_GETOOFFS, and audio_prinfo::{seek,samples} for AUDIO_GETINFO handle values for a buffer nearest to userland. * add `struct device *' parameter to ac97_attach() * all of audio HW drivers follow audio_hw_if and ac97 changes
2005-01-11 01:01:36 +03:00
stream_filter_t *this;
int m, err;
this = (stream_filter_t *)self;
Merge jmcneill-audiomp3 branch, which is derived from ad-audiomp2. From the original ad-audiomp branch notes: Add MP locking to the audio drivers. Making the audio drivers MP safe is necessary before efforts can be made to make the VM system MP safe. The are two locks per device instance, an ISR lock and a character device lock. The ISR lock replaces calls to splaudio()/splx(), and will be held across calls to device methods which were called at splaudio() before (e.g. trigger_output). The character device lock is held across calls to nearly all of the methods, excluding some only used for initialization, e.g. get_locks. Welcome to 5.99.57.
2011-11-24 03:07:28 +04:00
if ((err = this->prev->fetch_to(sc, this->prev, this->src, max_used)))
merge kent-audio1 branch, which introduces audio filter pipeline to the MI audio framework Summary of changes: * struct audio_params - remove sw_code, factor, factor_denom, hw_sample_rate, hw_encoding ,hw_precision, and hw_channels. Conversion information is conveyed by stream_filter_list_t. - change the type of sample_rate: u_long -> u_int - add `validbits,' which represents the valid data size in precision bits. It is required in order to distinguish 24/32bit from 24/24bit or 32/32bit. * audio_hw_if - add two parameters to set_params() stream_filter_list_t *pfil, stream_filter_list *rfil A HW driver should set filter recipes for requested formats - constify audio_params parameters of trigger_output() and trigger_input(). They represent audio formats for the hardware. - make open() and close() optional - add int (AUMODE_PLAY or AUMODE_RECORD) and audio_params_t parameters to round_blocksize() * sw_code is replaced with stream_filter_t. stream_filer_t converts audio data in an input buffer and writes into another output buffer unlike sw_code, which converts data in single buffer. converters in dev/auconv.c, dev/mulaw.c, dev/aurateconv.c, dev/tc/bba.c, dev/ic/msm6258.c, and arch/arm/iomd/vidcaudio.c are reimplemented as stream_filter_t * MI audio - audiosetinfo() builds filter pipelines from stream_filter_list_t filled by audio_hw_if::set_params() - audiosetinfo() returns with EINVAL if mmapped and set_params() requests filters - audio_write(), audio_pint(), and audio_rint() invoke a filter pipeline. - ioctl() for FIONREAD, AUDIO_WSEEK, AUDIO_GETIOFFS, AUDIO_GETOOFFS, and audio_prinfo::{seek,samples} for AUDIO_GETINFO handle values for a buffer nearest to userland. * add `struct device *' parameter to ac97_attach() * all of audio HW drivers follow audio_hw_if and ac97 changes
2005-01-11 01:01:36 +03:00
return err;
m = dst->end - dst->start;
m = min(m, max_used);
if (dst->param.encoding == AUDIO_ENCODING_ULINEAR_LE) {
FILTER_LOOP_PROLOGUE(this->src, 1, dst, 1, m) {
*d = alawtolin16[*s][0];
} FILTER_LOOP_EPILOGUE(this->src, dst);
} else { /* SLINEAR */
FILTER_LOOP_PROLOGUE(this->src, 1, dst, 1, m) {
*d = alawtolin16[*s][0] ^ 0x80;
} FILTER_LOOP_EPILOGUE(this->src, dst);
Add Alaw encoding emulation.
1997-05-28 04:07:46 +04:00
}
merge kent-audio1 branch, which introduces audio filter pipeline to the MI audio framework Summary of changes: * struct audio_params - remove sw_code, factor, factor_denom, hw_sample_rate, hw_encoding ,hw_precision, and hw_channels. Conversion information is conveyed by stream_filter_list_t. - change the type of sample_rate: u_long -> u_int - add `validbits,' which represents the valid data size in precision bits. It is required in order to distinguish 24/32bit from 24/24bit or 32/32bit. * audio_hw_if - add two parameters to set_params() stream_filter_list_t *pfil, stream_filter_list *rfil A HW driver should set filter recipes for requested formats - constify audio_params parameters of trigger_output() and trigger_input(). They represent audio formats for the hardware. - make open() and close() optional - add int (AUMODE_PLAY or AUMODE_RECORD) and audio_params_t parameters to round_blocksize() * sw_code is replaced with stream_filter_t. stream_filer_t converts audio data in an input buffer and writes into another output buffer unlike sw_code, which converts data in single buffer. converters in dev/auconv.c, dev/mulaw.c, dev/aurateconv.c, dev/tc/bba.c, dev/ic/msm6258.c, and arch/arm/iomd/vidcaudio.c are reimplemented as stream_filter_t * MI audio - audiosetinfo() builds filter pipelines from stream_filter_list_t filled by audio_hw_if::set_params() - audiosetinfo() returns with EINVAL if mmapped and set_params() requests filters - audio_write(), audio_pint(), and audio_rint() invoke a filter pipeline. - ioctl() for FIONREAD, AUDIO_WSEEK, AUDIO_GETIOFFS, AUDIO_GETOOFFS, and audio_prinfo::{seek,samples} for AUDIO_GETINFO handle values for a buffer nearest to userland. * add `struct device *' parameter to ac97_attach() * all of audio HW drivers follow audio_hw_if and ac97 changes
2005-01-11 01:01:36 +03:00
return 0;
Add Alaw encoding emulation.
1997-05-28 04:07:46 +04:00
}
merge kent-audio1 branch, which introduces audio filter pipeline to the MI audio framework Summary of changes: * struct audio_params - remove sw_code, factor, factor_denom, hw_sample_rate, hw_encoding ,hw_precision, and hw_channels. Conversion information is conveyed by stream_filter_list_t. - change the type of sample_rate: u_long -> u_int - add `validbits,' which represents the valid data size in precision bits. It is required in order to distinguish 24/32bit from 24/24bit or 32/32bit. * audio_hw_if - add two parameters to set_params() stream_filter_list_t *pfil, stream_filter_list *rfil A HW driver should set filter recipes for requested formats - constify audio_params parameters of trigger_output() and trigger_input(). They represent audio formats for the hardware. - make open() and close() optional - add int (AUMODE_PLAY or AUMODE_RECORD) and audio_params_t parameters to round_blocksize() * sw_code is replaced with stream_filter_t. stream_filer_t converts audio data in an input buffer and writes into another output buffer unlike sw_code, which converts data in single buffer. converters in dev/auconv.c, dev/mulaw.c, dev/aurateconv.c, dev/tc/bba.c, dev/ic/msm6258.c, and arch/arm/iomd/vidcaudio.c are reimplemented as stream_filter_t * MI audio - audiosetinfo() builds filter pipelines from stream_filter_list_t filled by audio_hw_if::set_params() - audiosetinfo() returns with EINVAL if mmapped and set_params() requests filters - audio_write(), audio_pint(), and audio_rint() invoke a filter pipeline. - ioctl() for FIONREAD, AUDIO_WSEEK, AUDIO_GETIOFFS, AUDIO_GETOOFFS, and audio_prinfo::{seek,samples} for AUDIO_GETINFO handle values for a buffer nearest to userland. * add `struct device *' parameter to ac97_attach() * all of audio HW drivers follow audio_hw_if and ac97 changes
2005-01-11 01:01:36 +03:00
DEFINE_FILTER(alaw_to_linear16)
Add mulaw_to_slinear16 and alaw_to_slinear16.
1998-08-09 23:22:15 +04:00
{
merge kent-audio1 branch, which introduces audio filter pipeline to the MI audio framework Summary of changes: * struct audio_params - remove sw_code, factor, factor_denom, hw_sample_rate, hw_encoding ,hw_precision, and hw_channels. Conversion information is conveyed by stream_filter_list_t. - change the type of sample_rate: u_long -> u_int - add `validbits,' which represents the valid data size in precision bits. It is required in order to distinguish 24/32bit from 24/24bit or 32/32bit. * audio_hw_if - add two parameters to set_params() stream_filter_list_t *pfil, stream_filter_list *rfil A HW driver should set filter recipes for requested formats - constify audio_params parameters of trigger_output() and trigger_input(). They represent audio formats for the hardware. - make open() and close() optional - add int (AUMODE_PLAY or AUMODE_RECORD) and audio_params_t parameters to round_blocksize() * sw_code is replaced with stream_filter_t. stream_filer_t converts audio data in an input buffer and writes into another output buffer unlike sw_code, which converts data in single buffer. converters in dev/auconv.c, dev/mulaw.c, dev/aurateconv.c, dev/tc/bba.c, dev/ic/msm6258.c, and arch/arm/iomd/vidcaudio.c are reimplemented as stream_filter_t * MI audio - audiosetinfo() builds filter pipelines from stream_filter_list_t filled by audio_hw_if::set_params() - audiosetinfo() returns with EINVAL if mmapped and set_params() requests filters - audio_write(), audio_pint(), and audio_rint() invoke a filter pipeline. - ioctl() for FIONREAD, AUDIO_WSEEK, AUDIO_GETIOFFS, AUDIO_GETOOFFS, and audio_prinfo::{seek,samples} for AUDIO_GETINFO handle values for a buffer nearest to userland. * add `struct device *' parameter to ac97_attach() * all of audio HW drivers follow audio_hw_if and ac97 changes
2005-01-11 01:01:36 +03:00
stream_filter_t *this;
int m, err;
this = (stream_filter_t *)self;
max_used = (max_used + 1) & ~1; /* round up to even */
Merge jmcneill-audiomp3 branch, which is derived from ad-audiomp2. From the original ad-audiomp branch notes: Add MP locking to the audio drivers. Making the audio drivers MP safe is necessary before efforts can be made to make the VM system MP safe. The are two locks per device instance, an ISR lock and a character device lock. The ISR lock replaces calls to splaudio()/splx(), and will be held across calls to device methods which were called at splaudio() before (e.g. trigger_output). The character device lock is held across calls to nearly all of the methods, excluding some only used for initialization, e.g. get_locks. Welcome to 5.99.57.
2011-11-24 03:07:28 +04:00
if ((err = this->prev->fetch_to(sc, this->prev, this->src, max_used / 2)))
merge kent-audio1 branch, which introduces audio filter pipeline to the MI audio framework Summary of changes: * struct audio_params - remove sw_code, factor, factor_denom, hw_sample_rate, hw_encoding ,hw_precision, and hw_channels. Conversion information is conveyed by stream_filter_list_t. - change the type of sample_rate: u_long -> u_int - add `validbits,' which represents the valid data size in precision bits. It is required in order to distinguish 24/32bit from 24/24bit or 32/32bit. * audio_hw_if - add two parameters to set_params() stream_filter_list_t *pfil, stream_filter_list *rfil A HW driver should set filter recipes for requested formats - constify audio_params parameters of trigger_output() and trigger_input(). They represent audio formats for the hardware. - make open() and close() optional - add int (AUMODE_PLAY or AUMODE_RECORD) and audio_params_t parameters to round_blocksize() * sw_code is replaced with stream_filter_t. stream_filer_t converts audio data in an input buffer and writes into another output buffer unlike sw_code, which converts data in single buffer. converters in dev/auconv.c, dev/mulaw.c, dev/aurateconv.c, dev/tc/bba.c, dev/ic/msm6258.c, and arch/arm/iomd/vidcaudio.c are reimplemented as stream_filter_t * MI audio - audiosetinfo() builds filter pipelines from stream_filter_list_t filled by audio_hw_if::set_params() - audiosetinfo() returns with EINVAL if mmapped and set_params() requests filters - audio_write(), audio_pint(), and audio_rint() invoke a filter pipeline. - ioctl() for FIONREAD, AUDIO_WSEEK, AUDIO_GETIOFFS, AUDIO_GETOOFFS, and audio_prinfo::{seek,samples} for AUDIO_GETINFO handle values for a buffer nearest to userland. * add `struct device *' parameter to ac97_attach() * all of audio HW drivers follow audio_hw_if and ac97 changes
2005-01-11 01:01:36 +03:00
return err;
m = (dst->end - dst->start) & ~1;
m = min(m, max_used);
switch (dst->param.encoding) {
case AUDIO_ENCODING_ULINEAR_LE:
FILTER_LOOP_PROLOGUE(this->src, 1, dst, 2, m) {
d[0] = alawtolin16[s[0]][1];
d[1] = alawtolin16[s[0]][0];
} FILTER_LOOP_EPILOGUE(this->src, dst);
break;
case AUDIO_ENCODING_ULINEAR_BE:
FILTER_LOOP_PROLOGUE(this->src, 1, dst, 2, m) {
d[0] = alawtolin16[s[0]][0];
d[1] = alawtolin16[s[0]][1];
} FILTER_LOOP_EPILOGUE(this->src, dst);
break;
case AUDIO_ENCODING_SLINEAR_LE:
FILTER_LOOP_PROLOGUE(this->src, 1, dst, 2, m) {
d[0] = alawtolin16[s[0]][1];
d[1] = alawtolin16[s[0]][0] ^ 0x80;
} FILTER_LOOP_EPILOGUE(this->src, dst);
break;
case AUDIO_ENCODING_SLINEAR_BE:
FILTER_LOOP_PROLOGUE(this->src, 1, dst, 2, m) {
d[0] = alawtolin16[s[0]][0] ^ 0x80;
d[1] = alawtolin16[s[0]][1];
} FILTER_LOOP_EPILOGUE(this->src, dst);
break;
Add mulaw_to_slinear16 and alaw_to_slinear16.
1998-08-09 23:22:15 +04:00
}
merge kent-audio1 branch, which introduces audio filter pipeline to the MI audio framework Summary of changes: * struct audio_params - remove sw_code, factor, factor_denom, hw_sample_rate, hw_encoding ,hw_precision, and hw_channels. Conversion information is conveyed by stream_filter_list_t. - change the type of sample_rate: u_long -> u_int - add `validbits,' which represents the valid data size in precision bits. It is required in order to distinguish 24/32bit from 24/24bit or 32/32bit. * audio_hw_if - add two parameters to set_params() stream_filter_list_t *pfil, stream_filter_list *rfil A HW driver should set filter recipes for requested formats - constify audio_params parameters of trigger_output() and trigger_input(). They represent audio formats for the hardware. - make open() and close() optional - add int (AUMODE_PLAY or AUMODE_RECORD) and audio_params_t parameters to round_blocksize() * sw_code is replaced with stream_filter_t. stream_filer_t converts audio data in an input buffer and writes into another output buffer unlike sw_code, which converts data in single buffer. converters in dev/auconv.c, dev/mulaw.c, dev/aurateconv.c, dev/tc/bba.c, dev/ic/msm6258.c, and arch/arm/iomd/vidcaudio.c are reimplemented as stream_filter_t * MI audio - audiosetinfo() builds filter pipelines from stream_filter_list_t filled by audio_hw_if::set_params() - audiosetinfo() returns with EINVAL if mmapped and set_params() requests filters - audio_write(), audio_pint(), and audio_rint() invoke a filter pipeline. - ioctl() for FIONREAD, AUDIO_WSEEK, AUDIO_GETIOFFS, AUDIO_GETOOFFS, and audio_prinfo::{seek,samples} for AUDIO_GETINFO handle values for a buffer nearest to userland. * add `struct device *' parameter to ac97_attach() * all of audio HW drivers follow audio_hw_if and ac97 changes
2005-01-11 01:01:36 +03:00
return 0;
Add mulaw_to_slinear16 and alaw_to_slinear16.
1998-08-09 23:22:15 +04:00
}
merge kent-audio1 branch, which introduces audio filter pipeline to the MI audio framework Summary of changes: * struct audio_params - remove sw_code, factor, factor_denom, hw_sample_rate, hw_encoding ,hw_precision, and hw_channels. Conversion information is conveyed by stream_filter_list_t. - change the type of sample_rate: u_long -> u_int - add `validbits,' which represents the valid data size in precision bits. It is required in order to distinguish 24/32bit from 24/24bit or 32/32bit. * audio_hw_if - add two parameters to set_params() stream_filter_list_t *pfil, stream_filter_list *rfil A HW driver should set filter recipes for requested formats - constify audio_params parameters of trigger_output() and trigger_input(). They represent audio formats for the hardware. - make open() and close() optional - add int (AUMODE_PLAY or AUMODE_RECORD) and audio_params_t parameters to round_blocksize() * sw_code is replaced with stream_filter_t. stream_filer_t converts audio data in an input buffer and writes into another output buffer unlike sw_code, which converts data in single buffer. converters in dev/auconv.c, dev/mulaw.c, dev/aurateconv.c, dev/tc/bba.c, dev/ic/msm6258.c, and arch/arm/iomd/vidcaudio.c are reimplemented as stream_filter_t * MI audio - audiosetinfo() builds filter pipelines from stream_filter_list_t filled by audio_hw_if::set_params() - audiosetinfo() returns with EINVAL if mmapped and set_params() requests filters - audio_write(), audio_pint(), and audio_rint() invoke a filter pipeline. - ioctl() for FIONREAD, AUDIO_WSEEK, AUDIO_GETIOFFS, AUDIO_GETOOFFS, and audio_prinfo::{seek,samples} for AUDIO_GETINFO handle values for a buffer nearest to userland. * add `struct device *' parameter to ac97_attach() * all of audio HW drivers follow audio_hw_if and ac97 changes
2005-01-11 01:01:36 +03:00
DEFINE_FILTER(linear8_to_alaw)
Add slinear16_to_mulaw_le().
2002-02-10 09:27:06 +03:00
{
merge kent-audio1 branch, which introduces audio filter pipeline to the MI audio framework Summary of changes: * struct audio_params - remove sw_code, factor, factor_denom, hw_sample_rate, hw_encoding ,hw_precision, and hw_channels. Conversion information is conveyed by stream_filter_list_t. - change the type of sample_rate: u_long -> u_int - add `validbits,' which represents the valid data size in precision bits. It is required in order to distinguish 24/32bit from 24/24bit or 32/32bit. * audio_hw_if - add two parameters to set_params() stream_filter_list_t *pfil, stream_filter_list *rfil A HW driver should set filter recipes for requested formats - constify audio_params parameters of trigger_output() and trigger_input(). They represent audio formats for the hardware. - make open() and close() optional - add int (AUMODE_PLAY or AUMODE_RECORD) and audio_params_t parameters to round_blocksize() * sw_code is replaced with stream_filter_t. stream_filer_t converts audio data in an input buffer and writes into another output buffer unlike sw_code, which converts data in single buffer. converters in dev/auconv.c, dev/mulaw.c, dev/aurateconv.c, dev/tc/bba.c, dev/ic/msm6258.c, and arch/arm/iomd/vidcaudio.c are reimplemented as stream_filter_t * MI audio - audiosetinfo() builds filter pipelines from stream_filter_list_t filled by audio_hw_if::set_params() - audiosetinfo() returns with EINVAL if mmapped and set_params() requests filters - audio_write(), audio_pint(), and audio_rint() invoke a filter pipeline. - ioctl() for FIONREAD, AUDIO_WSEEK, AUDIO_GETIOFFS, AUDIO_GETOOFFS, and audio_prinfo::{seek,samples} for AUDIO_GETINFO handle values for a buffer nearest to userland. * add `struct device *' parameter to ac97_attach() * all of audio HW drivers follow audio_hw_if and ac97 changes
2005-01-11 01:01:36 +03:00
stream_filter_t *this;
int m, err;
this = (stream_filter_t *)self;
Merge jmcneill-audiomp3 branch, which is derived from ad-audiomp2. From the original ad-audiomp branch notes: Add MP locking to the audio drivers. Making the audio drivers MP safe is necessary before efforts can be made to make the VM system MP safe. The are two locks per device instance, an ISR lock and a character device lock. The ISR lock replaces calls to splaudio()/splx(), and will be held across calls to device methods which were called at splaudio() before (e.g. trigger_output). The character device lock is held across calls to nearly all of the methods, excluding some only used for initialization, e.g. get_locks. Welcome to 5.99.57.
2011-11-24 03:07:28 +04:00
if ((err = this->prev->fetch_to(sc, this->prev, this->src, max_used)))
merge kent-audio1 branch, which introduces audio filter pipeline to the MI audio framework Summary of changes: * struct audio_params - remove sw_code, factor, factor_denom, hw_sample_rate, hw_encoding ,hw_precision, and hw_channels. Conversion information is conveyed by stream_filter_list_t. - change the type of sample_rate: u_long -> u_int - add `validbits,' which represents the valid data size in precision bits. It is required in order to distinguish 24/32bit from 24/24bit or 32/32bit. * audio_hw_if - add two parameters to set_params() stream_filter_list_t *pfil, stream_filter_list *rfil A HW driver should set filter recipes for requested formats - constify audio_params parameters of trigger_output() and trigger_input(). They represent audio formats for the hardware. - make open() and close() optional - add int (AUMODE_PLAY or AUMODE_RECORD) and audio_params_t parameters to round_blocksize() * sw_code is replaced with stream_filter_t. stream_filer_t converts audio data in an input buffer and writes into another output buffer unlike sw_code, which converts data in single buffer. converters in dev/auconv.c, dev/mulaw.c, dev/aurateconv.c, dev/tc/bba.c, dev/ic/msm6258.c, and arch/arm/iomd/vidcaudio.c are reimplemented as stream_filter_t * MI audio - audiosetinfo() builds filter pipelines from stream_filter_list_t filled by audio_hw_if::set_params() - audiosetinfo() returns with EINVAL if mmapped and set_params() requests filters - audio_write(), audio_pint(), and audio_rint() invoke a filter pipeline. - ioctl() for FIONREAD, AUDIO_WSEEK, AUDIO_GETIOFFS, AUDIO_GETOOFFS, and audio_prinfo::{seek,samples} for AUDIO_GETINFO handle values for a buffer nearest to userland. * add `struct device *' parameter to ac97_attach() * all of audio HW drivers follow audio_hw_if and ac97 changes
2005-01-11 01:01:36 +03:00
return err;
m = dst->end - dst->start;
m = min(m, max_used);
if (this->src->param.encoding == AUDIO_ENCODING_ULINEAR_LE) {
FILTER_LOOP_PROLOGUE(this->src, 1, dst, 1, m) {
*d = lintoalaw[*s];
} FILTER_LOOP_EPILOGUE(this->src, dst);
} else { /* SLINEAR_LE */
FILTER_LOOP_PROLOGUE(this->src, 1, dst, 1, m) {
*d = lintoalaw[*s ^ 0x80];
} FILTER_LOOP_EPILOGUE(this->src, dst);
add the follwings: - ulinear16_to_mulaw_le() - ulinear16_to_mulaw_be() - slinear16_to_mulaw_be() - ulinear16_to_alaw_le() - ulinear16_to_alaw_be()
2004-11-05 19:31:14 +03:00
}
merge kent-audio1 branch, which introduces audio filter pipeline to the MI audio framework Summary of changes: * struct audio_params - remove sw_code, factor, factor_denom, hw_sample_rate, hw_encoding ,hw_precision, and hw_channels. Conversion information is conveyed by stream_filter_list_t. - change the type of sample_rate: u_long -> u_int - add `validbits,' which represents the valid data size in precision bits. It is required in order to distinguish 24/32bit from 24/24bit or 32/32bit. * audio_hw_if - add two parameters to set_params() stream_filter_list_t *pfil, stream_filter_list *rfil A HW driver should set filter recipes for requested formats - constify audio_params parameters of trigger_output() and trigger_input(). They represent audio formats for the hardware. - make open() and close() optional - add int (AUMODE_PLAY or AUMODE_RECORD) and audio_params_t parameters to round_blocksize() * sw_code is replaced with stream_filter_t. stream_filer_t converts audio data in an input buffer and writes into another output buffer unlike sw_code, which converts data in single buffer. converters in dev/auconv.c, dev/mulaw.c, dev/aurateconv.c, dev/tc/bba.c, dev/ic/msm6258.c, and arch/arm/iomd/vidcaudio.c are reimplemented as stream_filter_t * MI audio - audiosetinfo() builds filter pipelines from stream_filter_list_t filled by audio_hw_if::set_params() - audiosetinfo() returns with EINVAL if mmapped and set_params() requests filters - audio_write(), audio_pint(), and audio_rint() invoke a filter pipeline. - ioctl() for FIONREAD, AUDIO_WSEEK, AUDIO_GETIOFFS, AUDIO_GETOOFFS, and audio_prinfo::{seek,samples} for AUDIO_GETINFO handle values for a buffer nearest to userland. * add `struct device *' parameter to ac97_attach() * all of audio HW drivers follow audio_hw_if and ac97 changes
2005-01-11 01:01:36 +03:00
return 0;
add the follwings: - ulinear16_to_mulaw_le() - ulinear16_to_mulaw_be() - slinear16_to_mulaw_be() - ulinear16_to_alaw_le() - ulinear16_to_alaw_be()
2004-11-05 19:31:14 +03:00
}
merge kent-audio1 branch, which introduces audio filter pipeline to the MI audio framework Summary of changes: * struct audio_params - remove sw_code, factor, factor_denom, hw_sample_rate, hw_encoding ,hw_precision, and hw_channels. Conversion information is conveyed by stream_filter_list_t. - change the type of sample_rate: u_long -> u_int - add `validbits,' which represents the valid data size in precision bits. It is required in order to distinguish 24/32bit from 24/24bit or 32/32bit. * audio_hw_if - add two parameters to set_params() stream_filter_list_t *pfil, stream_filter_list *rfil A HW driver should set filter recipes for requested formats - constify audio_params parameters of trigger_output() and trigger_input(). They represent audio formats for the hardware. - make open() and close() optional - add int (AUMODE_PLAY or AUMODE_RECORD) and audio_params_t parameters to round_blocksize() * sw_code is replaced with stream_filter_t. stream_filer_t converts audio data in an input buffer and writes into another output buffer unlike sw_code, which converts data in single buffer. converters in dev/auconv.c, dev/mulaw.c, dev/aurateconv.c, dev/tc/bba.c, dev/ic/msm6258.c, and arch/arm/iomd/vidcaudio.c are reimplemented as stream_filter_t * MI audio - audiosetinfo() builds filter pipelines from stream_filter_list_t filled by audio_hw_if::set_params() - audiosetinfo() returns with EINVAL if mmapped and set_params() requests filters - audio_write(), audio_pint(), and audio_rint() invoke a filter pipeline. - ioctl() for FIONREAD, AUDIO_WSEEK, AUDIO_GETIOFFS, AUDIO_GETOOFFS, and audio_prinfo::{seek,samples} for AUDIO_GETINFO handle values for a buffer nearest to userland. * add `struct device *' parameter to ac97_attach() * all of audio HW drivers follow audio_hw_if and ac97 changes
2005-01-11 01:01:36 +03:00
DEFINE_FILTER(linear16_to_alaw)
Add Alaw encoding emulation.
1997-05-28 04:07:46 +04:00
{
merge kent-audio1 branch, which introduces audio filter pipeline to the MI audio framework Summary of changes: * struct audio_params - remove sw_code, factor, factor_denom, hw_sample_rate, hw_encoding ,hw_precision, and hw_channels. Conversion information is conveyed by stream_filter_list_t. - change the type of sample_rate: u_long -> u_int - add `validbits,' which represents the valid data size in precision bits. It is required in order to distinguish 24/32bit from 24/24bit or 32/32bit. * audio_hw_if - add two parameters to set_params() stream_filter_list_t *pfil, stream_filter_list *rfil A HW driver should set filter recipes for requested formats - constify audio_params parameters of trigger_output() and trigger_input(). They represent audio formats for the hardware. - make open() and close() optional - add int (AUMODE_PLAY or AUMODE_RECORD) and audio_params_t parameters to round_blocksize() * sw_code is replaced with stream_filter_t. stream_filer_t converts audio data in an input buffer and writes into another output buffer unlike sw_code, which converts data in single buffer. converters in dev/auconv.c, dev/mulaw.c, dev/aurateconv.c, dev/tc/bba.c, dev/ic/msm6258.c, and arch/arm/iomd/vidcaudio.c are reimplemented as stream_filter_t * MI audio - audiosetinfo() builds filter pipelines from stream_filter_list_t filled by audio_hw_if::set_params() - audiosetinfo() returns with EINVAL if mmapped and set_params() requests filters - audio_write(), audio_pint(), and audio_rint() invoke a filter pipeline. - ioctl() for FIONREAD, AUDIO_WSEEK, AUDIO_GETIOFFS, AUDIO_GETOOFFS, and audio_prinfo::{seek,samples} for AUDIO_GETINFO handle values for a buffer nearest to userland. * add `struct device *' parameter to ac97_attach() * all of audio HW drivers follow audio_hw_if and ac97 changes
2005-01-11 01:01:36 +03:00
stream_filter_t *this;
int m, err;
this = (stream_filter_t *)self;
Merge jmcneill-audiomp3 branch, which is derived from ad-audiomp2. From the original ad-audiomp branch notes: Add MP locking to the audio drivers. Making the audio drivers MP safe is necessary before efforts can be made to make the VM system MP safe. The are two locks per device instance, an ISR lock and a character device lock. The ISR lock replaces calls to splaudio()/splx(), and will be held across calls to device methods which were called at splaudio() before (e.g. trigger_output). The character device lock is held across calls to nearly all of the methods, excluding some only used for initialization, e.g. get_locks. Welcome to 5.99.57.
2011-11-24 03:07:28 +04:00
if ((err = this->prev->fetch_to(sc, this->prev, this->src, max_used * 2)))
merge kent-audio1 branch, which introduces audio filter pipeline to the MI audio framework Summary of changes: * struct audio_params - remove sw_code, factor, factor_denom, hw_sample_rate, hw_encoding ,hw_precision, and hw_channels. Conversion information is conveyed by stream_filter_list_t. - change the type of sample_rate: u_long -> u_int - add `validbits,' which represents the valid data size in precision bits. It is required in order to distinguish 24/32bit from 24/24bit or 32/32bit. * audio_hw_if - add two parameters to set_params() stream_filter_list_t *pfil, stream_filter_list *rfil A HW driver should set filter recipes for requested formats - constify audio_params parameters of trigger_output() and trigger_input(). They represent audio formats for the hardware. - make open() and close() optional - add int (AUMODE_PLAY or AUMODE_RECORD) and audio_params_t parameters to round_blocksize() * sw_code is replaced with stream_filter_t. stream_filer_t converts audio data in an input buffer and writes into another output buffer unlike sw_code, which converts data in single buffer. converters in dev/auconv.c, dev/mulaw.c, dev/aurateconv.c, dev/tc/bba.c, dev/ic/msm6258.c, and arch/arm/iomd/vidcaudio.c are reimplemented as stream_filter_t * MI audio - audiosetinfo() builds filter pipelines from stream_filter_list_t filled by audio_hw_if::set_params() - audiosetinfo() returns with EINVAL if mmapped and set_params() requests filters - audio_write(), audio_pint(), and audio_rint() invoke a filter pipeline. - ioctl() for FIONREAD, AUDIO_WSEEK, AUDIO_GETIOFFS, AUDIO_GETOOFFS, and audio_prinfo::{seek,samples} for AUDIO_GETINFO handle values for a buffer nearest to userland. * add `struct device *' parameter to ac97_attach() * all of audio HW drivers follow audio_hw_if and ac97 changes
2005-01-11 01:01:36 +03:00
return err;
m = dst->end - dst->start;
m = min(m, max_used);
switch (this->src->param.encoding) {
case AUDIO_ENCODING_SLINEAR_LE:
FILTER_LOOP_PROLOGUE(this->src, 2, dst, 1, m) {
d[0] = lintoalaw[s[1] ^ 0x80];
} FILTER_LOOP_EPILOGUE(this->src, dst);
break;
case AUDIO_ENCODING_SLINEAR_BE:
FILTER_LOOP_PROLOGUE(this->src, 2, dst, 1, m) {
d[0] = lintoalaw[s[0] ^ 0x80];
} FILTER_LOOP_EPILOGUE(this->src, dst);
break;
case AUDIO_ENCODING_ULINEAR_LE:
FILTER_LOOP_PROLOGUE(this->src, 2, dst, 1, m) {
d[0] = lintoalaw[s[1]];
} FILTER_LOOP_EPILOGUE(this->src, dst);
break;
case AUDIO_ENCODING_ULINEAR_BE:
FILTER_LOOP_PROLOGUE(this->src, 2, dst, 1, m) {
d[0] = lintoalaw[s[0]];
} FILTER_LOOP_EPILOGUE(this->src, dst);
break;
add the follwings: - ulinear16_to_mulaw_le() - ulinear16_to_mulaw_be() - slinear16_to_mulaw_be() - ulinear16_to_alaw_le() - ulinear16_to_alaw_be()
2004-11-05 19:31:14 +03:00
}
merge kent-audio1 branch, which introduces audio filter pipeline to the MI audio framework Summary of changes: * struct audio_params - remove sw_code, factor, factor_denom, hw_sample_rate, hw_encoding ,hw_precision, and hw_channels. Conversion information is conveyed by stream_filter_list_t. - change the type of sample_rate: u_long -> u_int - add `validbits,' which represents the valid data size in precision bits. It is required in order to distinguish 24/32bit from 24/24bit or 32/32bit. * audio_hw_if - add two parameters to set_params() stream_filter_list_t *pfil, stream_filter_list *rfil A HW driver should set filter recipes for requested formats - constify audio_params parameters of trigger_output() and trigger_input(). They represent audio formats for the hardware. - make open() and close() optional - add int (AUMODE_PLAY or AUMODE_RECORD) and audio_params_t parameters to round_blocksize() * sw_code is replaced with stream_filter_t. stream_filer_t converts audio data in an input buffer and writes into another output buffer unlike sw_code, which converts data in single buffer. converters in dev/auconv.c, dev/mulaw.c, dev/aurateconv.c, dev/tc/bba.c, dev/ic/msm6258.c, and arch/arm/iomd/vidcaudio.c are reimplemented as stream_filter_t * MI audio - audiosetinfo() builds filter pipelines from stream_filter_list_t filled by audio_hw_if::set_params() - audiosetinfo() returns with EINVAL if mmapped and set_params() requests filters - audio_write(), audio_pint(), and audio_rint() invoke a filter pipeline. - ioctl() for FIONREAD, AUDIO_WSEEK, AUDIO_GETIOFFS, AUDIO_GETOOFFS, and audio_prinfo::{seek,samples} for AUDIO_GETINFO handle values for a buffer nearest to userland. * add `struct device *' parameter to ac97_attach() * all of audio HW drivers follow audio_hw_if and ac97 changes
2005-01-11 01:01:36 +03:00
return 0;
add the follwings: - ulinear16_to_mulaw_le() - ulinear16_to_mulaw_be() - slinear16_to_mulaw_be() - ulinear16_to_alaw_le() - ulinear16_to_alaw_be()
2004-11-05 19:31:14 +03:00
}
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