2004-11-10 07:09:44 +08:00
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/*
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* QEMU Audio subsystem header
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2005-10-31 02:58:22 +08:00
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*
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* Copyright (c) 2003-2005 Vassili Karpov (malc)
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*
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2004-11-10 07:09:44 +08:00
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* Permission is hereby granted, free of charge, to any person obtaining a copy
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* of this software and associated documentation files (the "Software"), to deal
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* in the Software without restriction, including without limitation the rights
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* to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
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* copies of the Software, and to permit persons to whom the Software is
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* furnished to do so, subject to the following conditions:
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*
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* The above copyright notice and this permission notice shall be included in
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* all copies or substantial portions of the Software.
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*
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* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
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* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
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* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
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* THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
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* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
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* OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
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* THE SOFTWARE.
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*/
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2016-06-29 21:29:06 +08:00
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2004-11-10 07:09:44 +08:00
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#ifndef QEMU_AUDIO_INT_H
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#define QEMU_AUDIO_INT_H
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2018-03-01 18:05:46 +08:00
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#ifdef CONFIG_AUDIO_COREAUDIO
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2005-10-31 02:58:22 +08:00
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#define FLOAT_MIXENG
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/* #define RECIPROCAL */
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#endif
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#include "mixeng.h"
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2021-03-09 21:15:28 +08:00
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#ifdef CONFIG_GIO
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#include <gio/gio.h>
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#endif
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2005-10-31 02:58:22 +08:00
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struct audio_pcm_ops;
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struct audio_callback {
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void *opaque;
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2009-10-15 06:40:17 +08:00
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audio_callback_fn fn;
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2005-10-31 02:58:22 +08:00
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};
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2004-11-10 07:09:44 +08:00
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2005-10-31 02:58:22 +08:00
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struct audio_pcm_info {
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int bits;
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2020-02-03 03:38:07 +08:00
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bool is_signed;
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bool is_float;
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2005-10-31 02:58:22 +08:00
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int freq;
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int nchannels;
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2019-10-14 03:58:02 +08:00
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int bytes_per_frame;
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2005-10-31 02:58:22 +08:00
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int bytes_per_second;
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2006-07-05 05:47:22 +08:00
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int swap_endianness;
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2005-10-31 02:58:22 +08:00
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};
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2004-11-10 07:09:44 +08:00
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2019-08-19 07:06:46 +08:00
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typedef struct AudioState AudioState;
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2006-07-17 02:57:03 +08:00
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typedef struct SWVoiceCap SWVoiceCap;
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2019-09-20 05:24:20 +08:00
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typedef struct STSampleBuffer {
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size_t pos, size;
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st_sample samples[];
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} STSampleBuffer;
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2005-10-31 02:58:22 +08:00
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typedef struct HWVoiceOut {
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2019-08-19 07:06:46 +08:00
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AudioState *s;
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2004-11-10 07:09:44 +08:00
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int enabled;
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2009-09-12 06:28:45 +08:00
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int poll_mode;
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2004-11-10 07:09:44 +08:00
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int pending_disable;
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2005-10-31 02:58:22 +08:00
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struct audio_pcm_info info;
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2004-11-10 07:09:44 +08:00
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f_sample *clip;
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2005-10-31 02:58:22 +08:00
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uint64_t ts_helper;
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2004-11-10 07:09:44 +08:00
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2019-09-20 05:24:20 +08:00
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STSampleBuffer *mix_buf;
|
audio: api for mixeng code free backends
This will make it possible to skip mixeng with audio playback and
recording, allowing us to free ourselves from the limitations of the
current mixeng (stereo, int64 samples only). In this case, HW and SW
voices will be essentially the same, for every SW voice we will create
a HW voice, since we can no longer mix multiple voices together.
Some backends expect us to call a function when we have data ready
write()/read() style, while others provide a buffer and expects us to
directly write/read it, so for optimal performance audio_pcm_ops provide
methods for both cases. Previously backends asked mixeng for more data
in run_out/run_it, now instead mixeng or the frontends will call the
backends, so that's why two sets of functions required. audio.c
contains glue code between the two styles, so backends only ever have to
implement one style and frontends are free to call whichever is more
convenient for them.
Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
Message-id: 15a33c03a62228922d851f7324c52f73cb8d2414.1568927990.git.DirtY.iCE.hu@gmail.com
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
2019-09-20 05:24:09 +08:00
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void *buf_emul;
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size_t pos_emul, pending_emul, size_emul;
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2004-11-10 07:09:44 +08:00
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2019-08-19 07:06:58 +08:00
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size_t samples;
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2009-09-12 15:36:22 +08:00
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QLIST_HEAD (sw_out_listhead, SWVoiceOut) sw_head;
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QLIST_HEAD (sw_cap_listhead, SWVoiceCap) cap_head;
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2008-10-07 02:08:30 +08:00
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struct audio_pcm_ops *pcm_ops;
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2009-09-12 15:36:22 +08:00
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QLIST_ENTRY (HWVoiceOut) entries;
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2005-10-31 02:58:22 +08:00
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} HWVoiceOut;
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2004-11-10 07:09:44 +08:00
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2005-10-31 02:58:22 +08:00
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typedef struct HWVoiceIn {
|
2019-08-19 07:06:46 +08:00
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AudioState *s;
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2005-10-31 02:58:22 +08:00
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int enabled;
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2009-09-12 06:28:45 +08:00
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int poll_mode;
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2005-10-31 02:58:22 +08:00
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struct audio_pcm_info info;
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t_sample *conv;
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2004-11-12 00:55:09 +08:00
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2019-08-19 07:06:58 +08:00
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size_t total_samples_captured;
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2005-10-31 02:58:22 +08:00
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uint64_t ts_helper;
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2004-11-10 07:09:44 +08:00
|
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|
|
2019-09-20 05:24:20 +08:00
|
|
|
STSampleBuffer *conv_buf;
|
audio: api for mixeng code free backends
This will make it possible to skip mixeng with audio playback and
recording, allowing us to free ourselves from the limitations of the
current mixeng (stereo, int64 samples only). In this case, HW and SW
voices will be essentially the same, for every SW voice we will create
a HW voice, since we can no longer mix multiple voices together.
Some backends expect us to call a function when we have data ready
write()/read() style, while others provide a buffer and expects us to
directly write/read it, so for optimal performance audio_pcm_ops provide
methods for both cases. Previously backends asked mixeng for more data
in run_out/run_it, now instead mixeng or the frontends will call the
backends, so that's why two sets of functions required. audio.c
contains glue code between the two styles, so backends only ever have to
implement one style and frontends are free to call whichever is more
convenient for them.
Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
Message-id: 15a33c03a62228922d851f7324c52f73cb8d2414.1568927990.git.DirtY.iCE.hu@gmail.com
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
2019-09-20 05:24:09 +08:00
|
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void *buf_emul;
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size_t pos_emul, pending_emul, size_emul;
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2004-11-10 07:09:44 +08:00
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|
2019-08-19 07:06:58 +08:00
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size_t samples;
|
2009-09-12 15:36:22 +08:00
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QLIST_HEAD (sw_in_listhead, SWVoiceIn) sw_head;
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2008-10-07 02:08:30 +08:00
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struct audio_pcm_ops *pcm_ops;
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2009-09-12 15:36:22 +08:00
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QLIST_ENTRY (HWVoiceIn) entries;
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2005-10-31 02:58:22 +08:00
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} HWVoiceIn;
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2004-11-10 07:09:44 +08:00
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2005-10-31 02:58:22 +08:00
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struct SWVoiceOut {
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2009-05-14 07:11:35 +08:00
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QEMUSoundCard *card;
|
2019-08-19 07:06:46 +08:00
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AudioState *s;
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2005-10-31 02:58:22 +08:00
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struct audio_pcm_info info;
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2004-11-10 07:09:44 +08:00
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t_sample *conv;
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int64_t ratio;
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2008-12-04 06:48:44 +08:00
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struct st_sample *buf;
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2004-11-10 07:09:44 +08:00
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void *rate;
|
2019-08-19 07:06:58 +08:00
|
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size_t total_hw_samples_mixed;
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2005-10-31 02:58:22 +08:00
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int active;
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int empty;
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HWVoiceOut *hw;
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char *name;
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2008-12-04 06:48:44 +08:00
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struct mixeng_volume vol;
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2005-10-31 02:58:22 +08:00
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struct audio_callback callback;
|
2009-09-12 15:36:22 +08:00
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QLIST_ENTRY (SWVoiceOut) entries;
|
2005-10-31 02:58:22 +08:00
|
|
|
};
|
2004-11-10 07:09:44 +08:00
|
|
|
|
2005-10-31 02:58:22 +08:00
|
|
|
struct SWVoiceIn {
|
2009-05-14 07:11:35 +08:00
|
|
|
QEMUSoundCard *card;
|
2019-08-19 07:06:46 +08:00
|
|
|
AudioState *s;
|
2004-11-10 07:09:44 +08:00
|
|
|
int active;
|
2005-10-31 02:58:22 +08:00
|
|
|
struct audio_pcm_info info;
|
|
|
|
int64_t ratio;
|
|
|
|
void *rate;
|
2019-08-19 07:06:58 +08:00
|
|
|
size_t total_hw_samples_acquired;
|
2008-12-04 06:48:44 +08:00
|
|
|
struct st_sample *buf;
|
2005-10-31 02:58:22 +08:00
|
|
|
f_sample *clip;
|
|
|
|
HWVoiceIn *hw;
|
2004-11-10 07:09:44 +08:00
|
|
|
char *name;
|
2008-12-04 06:48:44 +08:00
|
|
|
struct mixeng_volume vol;
|
2005-10-31 02:58:22 +08:00
|
|
|
struct audio_callback callback;
|
2009-09-12 15:36:22 +08:00
|
|
|
QLIST_ENTRY (SWVoiceIn) entries;
|
2004-11-10 07:09:44 +08:00
|
|
|
};
|
|
|
|
|
2018-03-06 15:40:47 +08:00
|
|
|
typedef struct audio_driver audio_driver;
|
2005-11-06 02:55:28 +08:00
|
|
|
struct audio_driver {
|
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|
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const char *name;
|
|
|
|
const char *descr;
|
2019-03-09 06:34:15 +08:00
|
|
|
void *(*init) (Audiodev *);
|
2005-11-06 02:55:28 +08:00
|
|
|
void (*fini) (void *);
|
2021-03-09 21:15:28 +08:00
|
|
|
#ifdef CONFIG_GIO
|
|
|
|
void (*set_dbus_server) (AudioState *s, GDBusObjectManagerServer *manager);
|
|
|
|
#endif
|
2008-10-07 02:08:30 +08:00
|
|
|
struct audio_pcm_ops *pcm_ops;
|
2005-11-06 02:55:28 +08:00
|
|
|
int can_be_default;
|
|
|
|
int max_voices_out;
|
|
|
|
int max_voices_in;
|
|
|
|
int voice_size_out;
|
|
|
|
int voice_size_in;
|
2018-03-06 15:40:47 +08:00
|
|
|
QLIST_ENTRY(audio_driver) next;
|
2005-11-06 02:55:28 +08:00
|
|
|
};
|
|
|
|
|
2005-10-31 02:58:22 +08:00
|
|
|
struct audio_pcm_ops {
|
audio: api for mixeng code free backends
This will make it possible to skip mixeng with audio playback and
recording, allowing us to free ourselves from the limitations of the
current mixeng (stereo, int64 samples only). In this case, HW and SW
voices will be essentially the same, for every SW voice we will create
a HW voice, since we can no longer mix multiple voices together.
Some backends expect us to call a function when we have data ready
write()/read() style, while others provide a buffer and expects us to
directly write/read it, so for optimal performance audio_pcm_ops provide
methods for both cases. Previously backends asked mixeng for more data
in run_out/run_it, now instead mixeng or the frontends will call the
backends, so that's why two sets of functions required. audio.c
contains glue code between the two styles, so backends only ever have to
implement one style and frontends are free to call whichever is more
convenient for them.
Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
Message-id: 15a33c03a62228922d851f7324c52f73cb8d2414.1568927990.git.DirtY.iCE.hu@gmail.com
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
2019-09-20 05:24:09 +08:00
|
|
|
int (*init_out)(HWVoiceOut *hw, audsettings *as, void *drv_opaque);
|
|
|
|
void (*fini_out)(HWVoiceOut *hw);
|
|
|
|
size_t (*write) (HWVoiceOut *hw, void *buf, size_t size);
|
2020-01-23 15:49:39 +08:00
|
|
|
void (*run_buffer_out)(HWVoiceOut *hw);
|
2022-03-02 03:13:06 +08:00
|
|
|
/*
|
|
|
|
* Get the free output buffer size. This is an upper limit. The size
|
|
|
|
* returned by function get_buffer_out may be smaller.
|
|
|
|
*/
|
|
|
|
size_t (*buffer_get_free)(HWVoiceOut *hw);
|
audio: api for mixeng code free backends
This will make it possible to skip mixeng with audio playback and
recording, allowing us to free ourselves from the limitations of the
current mixeng (stereo, int64 samples only). In this case, HW and SW
voices will be essentially the same, for every SW voice we will create
a HW voice, since we can no longer mix multiple voices together.
Some backends expect us to call a function when we have data ready
write()/read() style, while others provide a buffer and expects us to
directly write/read it, so for optimal performance audio_pcm_ops provide
methods for both cases. Previously backends asked mixeng for more data
in run_out/run_it, now instead mixeng or the frontends will call the
backends, so that's why two sets of functions required. audio.c
contains glue code between the two styles, so backends only ever have to
implement one style and frontends are free to call whichever is more
convenient for them.
Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
Message-id: 15a33c03a62228922d851f7324c52f73cb8d2414.1568927990.git.DirtY.iCE.hu@gmail.com
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
2019-09-20 05:24:09 +08:00
|
|
|
/*
|
|
|
|
* get a buffer that after later can be passed to put_buffer_out; optional
|
|
|
|
* returns the buffer, and writes it's size to size (in bytes)
|
|
|
|
*/
|
|
|
|
void *(*get_buffer_out)(HWVoiceOut *hw, size_t *size);
|
|
|
|
/*
|
|
|
|
* put back the buffer returned by get_buffer_out; optional
|
|
|
|
* buf must be equal the pointer returned by get_buffer_out,
|
|
|
|
* size may be smaller
|
|
|
|
*/
|
|
|
|
size_t (*put_buffer_out)(HWVoiceOut *hw, void *buf, size_t size);
|
2019-09-20 05:24:22 +08:00
|
|
|
void (*enable_out)(HWVoiceOut *hw, bool enable);
|
2019-10-14 03:58:01 +08:00
|
|
|
void (*volume_out)(HWVoiceOut *hw, Volume *vol);
|
audio: api for mixeng code free backends
This will make it possible to skip mixeng with audio playback and
recording, allowing us to free ourselves from the limitations of the
current mixeng (stereo, int64 samples only). In this case, HW and SW
voices will be essentially the same, for every SW voice we will create
a HW voice, since we can no longer mix multiple voices together.
Some backends expect us to call a function when we have data ready
write()/read() style, while others provide a buffer and expects us to
directly write/read it, so for optimal performance audio_pcm_ops provide
methods for both cases. Previously backends asked mixeng for more data
in run_out/run_it, now instead mixeng or the frontends will call the
backends, so that's why two sets of functions required. audio.c
contains glue code between the two styles, so backends only ever have to
implement one style and frontends are free to call whichever is more
convenient for them.
Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
Message-id: 15a33c03a62228922d851f7324c52f73cb8d2414.1568927990.git.DirtY.iCE.hu@gmail.com
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
2019-09-20 05:24:09 +08:00
|
|
|
|
|
|
|
int (*init_in) (HWVoiceIn *hw, audsettings *as, void *drv_opaque);
|
|
|
|
void (*fini_in) (HWVoiceIn *hw);
|
|
|
|
size_t (*read) (HWVoiceIn *hw, void *buf, size_t size);
|
2021-01-10 18:02:24 +08:00
|
|
|
void (*run_buffer_in)(HWVoiceIn *hw);
|
audio: api for mixeng code free backends
This will make it possible to skip mixeng with audio playback and
recording, allowing us to free ourselves from the limitations of the
current mixeng (stereo, int64 samples only). In this case, HW and SW
voices will be essentially the same, for every SW voice we will create
a HW voice, since we can no longer mix multiple voices together.
Some backends expect us to call a function when we have data ready
write()/read() style, while others provide a buffer and expects us to
directly write/read it, so for optimal performance audio_pcm_ops provide
methods for both cases. Previously backends asked mixeng for more data
in run_out/run_it, now instead mixeng or the frontends will call the
backends, so that's why two sets of functions required. audio.c
contains glue code between the two styles, so backends only ever have to
implement one style and frontends are free to call whichever is more
convenient for them.
Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
Message-id: 15a33c03a62228922d851f7324c52f73cb8d2414.1568927990.git.DirtY.iCE.hu@gmail.com
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
2019-09-20 05:24:09 +08:00
|
|
|
void *(*get_buffer_in)(HWVoiceIn *hw, size_t *size);
|
|
|
|
void (*put_buffer_in)(HWVoiceIn *hw, void *buf, size_t size);
|
2019-09-20 05:24:22 +08:00
|
|
|
void (*enable_in)(HWVoiceIn *hw, bool enable);
|
2019-10-14 03:58:01 +08:00
|
|
|
void (*volume_in)(HWVoiceIn *hw, Volume *vol);
|
2004-11-10 07:09:44 +08:00
|
|
|
};
|
|
|
|
|
2021-01-10 18:02:24 +08:00
|
|
|
void audio_generic_run_buffer_in(HWVoiceIn *hw);
|
audio: api for mixeng code free backends
This will make it possible to skip mixeng with audio playback and
recording, allowing us to free ourselves from the limitations of the
current mixeng (stereo, int64 samples only). In this case, HW and SW
voices will be essentially the same, for every SW voice we will create
a HW voice, since we can no longer mix multiple voices together.
Some backends expect us to call a function when we have data ready
write()/read() style, while others provide a buffer and expects us to
directly write/read it, so for optimal performance audio_pcm_ops provide
methods for both cases. Previously backends asked mixeng for more data
in run_out/run_it, now instead mixeng or the frontends will call the
backends, so that's why two sets of functions required. audio.c
contains glue code between the two styles, so backends only ever have to
implement one style and frontends are free to call whichever is more
convenient for them.
Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
Message-id: 15a33c03a62228922d851f7324c52f73cb8d2414.1568927990.git.DirtY.iCE.hu@gmail.com
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
2019-09-20 05:24:09 +08:00
|
|
|
void *audio_generic_get_buffer_in(HWVoiceIn *hw, size_t *size);
|
|
|
|
void audio_generic_put_buffer_in(HWVoiceIn *hw, void *buf, size_t size);
|
2020-01-23 15:49:39 +08:00
|
|
|
void audio_generic_run_buffer_out(HWVoiceOut *hw);
|
2022-03-02 03:13:06 +08:00
|
|
|
size_t audio_generic_buffer_get_free(HWVoiceOut *hw);
|
audio: api for mixeng code free backends
This will make it possible to skip mixeng with audio playback and
recording, allowing us to free ourselves from the limitations of the
current mixeng (stereo, int64 samples only). In this case, HW and SW
voices will be essentially the same, for every SW voice we will create
a HW voice, since we can no longer mix multiple voices together.
Some backends expect us to call a function when we have data ready
write()/read() style, while others provide a buffer and expects us to
directly write/read it, so for optimal performance audio_pcm_ops provide
methods for both cases. Previously backends asked mixeng for more data
in run_out/run_it, now instead mixeng or the frontends will call the
backends, so that's why two sets of functions required. audio.c
contains glue code between the two styles, so backends only ever have to
implement one style and frontends are free to call whichever is more
convenient for them.
Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
Message-id: 15a33c03a62228922d851f7324c52f73cb8d2414.1568927990.git.DirtY.iCE.hu@gmail.com
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
2019-09-20 05:24:09 +08:00
|
|
|
void *audio_generic_get_buffer_out(HWVoiceOut *hw, size_t *size);
|
|
|
|
size_t audio_generic_put_buffer_out(HWVoiceOut *hw, void *buf, size_t size);
|
|
|
|
size_t audio_generic_write(HWVoiceOut *hw, void *buf, size_t size);
|
|
|
|
size_t audio_generic_read(HWVoiceIn *hw, void *buf, size_t size);
|
|
|
|
|
2006-07-05 00:51:32 +08:00
|
|
|
struct capture_callback {
|
|
|
|
struct audio_capture_ops ops;
|
|
|
|
void *opaque;
|
2009-09-12 15:36:22 +08:00
|
|
|
QLIST_ENTRY (capture_callback) entries;
|
2006-07-05 00:51:32 +08:00
|
|
|
};
|
|
|
|
|
2006-07-17 02:57:03 +08:00
|
|
|
struct CaptureVoiceOut {
|
2006-07-05 00:51:32 +08:00
|
|
|
HWVoiceOut hw;
|
|
|
|
void *buf;
|
2009-09-12 15:36:22 +08:00
|
|
|
QLIST_HEAD (cb_listhead, capture_callback) cb_head;
|
|
|
|
QLIST_ENTRY (CaptureVoiceOut) entries;
|
2006-07-17 02:57:03 +08:00
|
|
|
};
|
|
|
|
|
|
|
|
struct SWVoiceCap {
|
|
|
|
SWVoiceOut sw;
|
|
|
|
CaptureVoiceOut *cap;
|
2009-09-12 15:36:22 +08:00
|
|
|
QLIST_ENTRY (SWVoiceCap) entries;
|
2006-07-17 02:57:03 +08:00
|
|
|
};
|
2006-07-05 00:51:32 +08:00
|
|
|
|
2019-01-11 22:08:53 +08:00
|
|
|
typedef struct AudioState {
|
2005-11-06 02:55:28 +08:00
|
|
|
struct audio_driver *drv;
|
2019-03-09 06:34:15 +08:00
|
|
|
Audiodev *dev;
|
2005-11-06 02:55:28 +08:00
|
|
|
void *drv_opaque;
|
|
|
|
|
|
|
|
QEMUTimer *ts;
|
2009-09-12 15:36:22 +08:00
|
|
|
QLIST_HEAD (card_listhead, QEMUSoundCard) card_head;
|
|
|
|
QLIST_HEAD (hw_in_listhead, HWVoiceIn) hw_head_in;
|
|
|
|
QLIST_HEAD (hw_out_listhead, HWVoiceOut) hw_head_out;
|
|
|
|
QLIST_HEAD (cap_listhead, CaptureVoiceOut) cap_head;
|
2005-11-06 02:55:28 +08:00
|
|
|
int nb_hw_voices_out;
|
|
|
|
int nb_hw_voices_in;
|
2009-02-19 04:44:04 +08:00
|
|
|
int vm_running;
|
2019-03-09 06:34:15 +08:00
|
|
|
int64_t period_ticks;
|
2019-08-19 07:06:46 +08:00
|
|
|
|
|
|
|
bool timer_running;
|
|
|
|
uint64_t timer_last;
|
2019-08-19 07:06:47 +08:00
|
|
|
|
|
|
|
QTAILQ_ENTRY(AudioState) list;
|
2019-01-11 22:08:53 +08:00
|
|
|
} AudioState;
|
2005-11-06 02:55:28 +08:00
|
|
|
|
2011-01-05 08:05:47 +08:00
|
|
|
extern const struct mixeng_volume nominal_volume;
|
2005-11-06 02:55:28 +08:00
|
|
|
|
2019-03-09 06:34:15 +08:00
|
|
|
extern const char *audio_prio_list[];
|
|
|
|
|
2018-03-06 15:40:47 +08:00
|
|
|
void audio_driver_register(audio_driver *drv);
|
|
|
|
audio_driver *audio_driver_lookup(const char *name);
|
|
|
|
|
2008-12-04 06:48:44 +08:00
|
|
|
void audio_pcm_init_info (struct audio_pcm_info *info, struct audsettings *as);
|
2005-10-31 02:58:22 +08:00
|
|
|
void audio_pcm_info_clear_buf (struct audio_pcm_info *info, void *buf, int len);
|
|
|
|
|
2005-11-06 02:55:28 +08:00
|
|
|
int audio_bug (const char *funcname, int cond);
|
|
|
|
void *audio_calloc (const char *funcname, int nmemb, size_t size);
|
|
|
|
|
2019-08-19 07:06:55 +08:00
|
|
|
void audio_run(AudioState *s, const char *msg);
|
2009-09-12 06:28:45 +08:00
|
|
|
|
2021-05-18 03:46:03 +08:00
|
|
|
const char *audio_application_name(void);
|
|
|
|
|
2019-09-20 05:24:21 +08:00
|
|
|
typedef struct RateCtl {
|
|
|
|
int64_t start_ticks;
|
|
|
|
int64_t bytes_sent;
|
|
|
|
} RateCtl;
|
|
|
|
|
|
|
|
void audio_rate_start(RateCtl *rate);
|
|
|
|
size_t audio_rate_get_bytes(struct audio_pcm_info *info, RateCtl *rate,
|
|
|
|
size_t bytes_avail);
|
|
|
|
|
2019-08-19 07:06:58 +08:00
|
|
|
static inline size_t audio_ring_dist(size_t dst, size_t src, size_t len)
|
2005-10-31 02:58:22 +08:00
|
|
|
{
|
|
|
|
return (dst >= src) ? (dst - src) : (len - src + dst);
|
|
|
|
}
|
|
|
|
|
2022-03-02 03:12:57 +08:00
|
|
|
/**
|
|
|
|
* audio_ring_posb() - returns new position in ringbuffer in backward
|
|
|
|
* direction at given distance
|
|
|
|
*
|
|
|
|
* @pos: current position in ringbuffer
|
|
|
|
* @dist: distance in ringbuffer to walk in reverse direction
|
|
|
|
* @len: size of ringbuffer
|
|
|
|
*/
|
|
|
|
static inline size_t audio_ring_posb(size_t pos, size_t dist, size_t len)
|
|
|
|
{
|
|
|
|
return pos >= dist ? pos - dist : len - dist + pos;
|
|
|
|
}
|
|
|
|
|
2013-06-16 17:19:31 +08:00
|
|
|
#define dolog(fmt, ...) AUD_log(AUDIO_CAP, fmt, ## __VA_ARGS__)
|
2005-10-31 02:58:22 +08:00
|
|
|
|
|
|
|
#ifdef DEBUG
|
2013-06-16 17:19:31 +08:00
|
|
|
#define ldebug(fmt, ...) AUD_log(AUDIO_CAP, fmt, ## __VA_ARGS__)
|
2005-10-31 02:58:22 +08:00
|
|
|
#else
|
2013-06-16 17:19:31 +08:00
|
|
|
#define ldebug(fmt, ...) (void)0
|
2005-10-31 02:58:22 +08:00
|
|
|
#endif
|
|
|
|
|
|
|
|
#define AUDIO_STRINGIFY_(n) #n
|
|
|
|
#define AUDIO_STRINGIFY(n) AUDIO_STRINGIFY_(n)
|
|
|
|
|
2019-03-09 06:34:15 +08:00
|
|
|
typedef struct AudiodevListEntry {
|
|
|
|
Audiodev *dev;
|
|
|
|
QSIMPLEQ_ENTRY(AudiodevListEntry) next;
|
|
|
|
} AudiodevListEntry;
|
|
|
|
|
|
|
|
typedef QSIMPLEQ_HEAD(, AudiodevListEntry) AudiodevListHead;
|
|
|
|
AudiodevListHead audio_handle_legacy_opts(void);
|
|
|
|
|
|
|
|
void audio_free_audiodev_list(AudiodevListHead *head);
|
|
|
|
|
|
|
|
void audio_create_pdos(Audiodev *dev);
|
|
|
|
AudiodevPerDirectionOptions *audio_get_pdo_in(Audiodev *dev);
|
|
|
|
AudiodevPerDirectionOptions *audio_get_pdo_out(Audiodev *dev);
|
|
|
|
|
2016-06-29 21:29:06 +08:00
|
|
|
#endif /* QEMU_AUDIO_INT_H */
|