Re: [PATCH v4] audio/pwaudio.c: Add Pipewire audio backend for QEMU

[Dorinda Bassey <dbassey@redhat.com>]

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>
> Can you make it a bit silent? Probably all AUD_log() & printf*( should
> be replaced by trace.

Yes it can be silent. It just made sense to me to have the output.

Playback works for me, however recording is failing. I haven't
> investigated, is it working for you?
>
I did check this and it seems playback for other audio backends are not
working either.
If I may ask what device options are you using for recording? I suspect
incorrect parameter configuration for recording on my end.

SPDX identifier instead would be neat
>
noted thanks.

Why that much?

 Because a larger ringbuffer size would help prevent buffer overruns which
could trigger seg faults. Also a larger size provides more flexibility for
handling data of various sizes/rates.

It looks like you don't take latency into account. Isn't there a more
> direct/simple pw push API that takes care of internal buffering? The
> intermediate ringbuffer is not nice to deal with.
>
The ringbuffer implementation is an essential component of the audio
processing pipeline, Pipewire minimizes latency by buffering audio data in
the ringbuffer. It is also designed to be thread-safe such that multiple
threads can access the buffer simultaneously without causing data
corruption or other issues.

Since it is not linux specific, no need to put it there. (filed
> alignment is kind of pointless to me)
>
 Agreed

Where is this latency value coming from?
>
Ah, I missed that. It's supposed to be 15000

Should be 8.0
>
Yes, Thanks.

On Sun, Feb 26, 2023 at 2:32 PM Marc-André Lureau <
marcandre.lureau@gmail.com> wrote:

> Hi
>
> On Fri, Feb 17, 2023 at 9:08 PM Dorinda Bassey <dbassey@redhat.com> wrote:
> >
> > This commit adds a new audiodev backend to allow QEMU to use Pipewire as
> > both an audio sink and source. This backend is available on most systems
> >
> > Add Pipewire entry points for QEMU Pipewire audio backend
> > Add wrappers for QEMU Pipewire audio backend in qpw_pcm_ops()
> > qpw_write function returns the current state of the stream to pwaudio
> > and Writes some data to the server for playback streams using pipewire
> > spa_ringbuffer implementation.
> > qpw_read function returns the current state of the stream to pwaudio and
> > reads some data from the server for capture streams using pipewire
> > spa_ringbuffer implementation. These functions qpw_write and qpw_read
> > are called during playback and capture.
> > Added some functions that convert pw audio formats to QEMU audio format
> > and vice versa which would be needed in the pipewire audio sink and
> > source functions qpw_init_in() & qpw_init_out().
> > These methods that implement playback and recording will create streams
> > for playback and capture that will start processing and will result in
> > the on_process callbacks to be called.
> > Built a connection to the Pipewire sound system server in the
> > qpw_audio_init() method.
> >
> > Signed-off-by: Dorinda Bassey <dbassey@redhat.com>
>
> pipewire: Initialize PW context
> stream state: "connecting"
> stream state: "connecting"
> stream state: "paused"
> node id: 125
> stream state: "paused"
> node id: 142
> stream state: "streaming"
> stream state: "streaming"
>
> Can you make it a bit silent? Probably all AUD_log() & printf*( should
> be replaced by trace.
>
> Playback works for me, however recording is failing. I haven't
> investigated, is it working for you?
>
> > ---
> > fix typo
> > raise version dependency
> >
> >  audio/audio.c                 |   3 +
> >  audio/audio_template.h        |   4 +
> >  audio/meson.build             |   1 +
> >  audio/pwaudio.c               | 827 ++++++++++++++++++++++++++++++++++
> >  meson.build                   |   8 +
> >  meson_options.txt             |   4 +-
> >  qapi/audio.json               |  45 ++
> >  qemu-options.hx               |  17 +
> >  scripts/meson-buildoptions.sh |   8 +-
> >  9 files changed, 914 insertions(+), 3 deletions(-)
> >  create mode 100644 audio/pwaudio.c
> >
> > diff --git a/audio/audio.c b/audio/audio.c
> > index 4290309d18..aa55e41ad8 100644
> > --- a/audio/audio.c
> > +++ b/audio/audio.c
> > @@ -2069,6 +2069,9 @@ void audio_create_pdos(Audiodev *dev)
> >  #ifdef CONFIG_AUDIO_PA
> >          CASE(PA, pa, Pa);
> >  #endif
> > +#ifdef CONFIG_AUDIO_PIPEWIRE
> > +        CASE(PIPEWIRE, pipewire, Pipewire);
> > +#endif
> >  #ifdef CONFIG_AUDIO_SDL
> >          CASE(SDL, sdl, Sdl);
> >  #endif
> > diff --git a/audio/audio_template.h b/audio/audio_template.h
> > index 42b4712acb..0f02afb921 100644
> > --- a/audio/audio_template.h
> > +++ b/audio/audio_template.h
> > @@ -355,6 +355,10 @@ AudiodevPerDirectionOptions *glue(audio_get_pdo_,
> TYPE)(Audiodev *dev)
> >      case AUDIODEV_DRIVER_PA:
> >          return qapi_AudiodevPaPerDirectionOptions_base(dev->u.pa.TYPE);
> >  #endif
> > +#ifdef CONFIG_AUDIO_PIPEWIRE
> > +    case AUDIODEV_DRIVER_PIPEWIRE:
> > +        return
> qapi_AudiodevPipewirePerDirectionOptions_base(dev->u.pipewire.TYPE);
> > +#endif
> >  #ifdef CONFIG_AUDIO_SDL
> >      case AUDIODEV_DRIVER_SDL:
> >          return
> qapi_AudiodevSdlPerDirectionOptions_base(dev->u.sdl.TYPE);
> > diff --git a/audio/meson.build b/audio/meson.build
> > index 0722224ba9..65a49c1a10 100644
> > --- a/audio/meson.build
> > +++ b/audio/meson.build
> > @@ -19,6 +19,7 @@ foreach m : [
> >    ['sdl', sdl, files('sdlaudio.c')],
> >    ['jack', jack, files('jackaudio.c')],
> >    ['sndio', sndio, files('sndioaudio.c')],
> > +  ['pipewire', pipewire, files('pwaudio.c')],
> >    ['spice', spice, files('spiceaudio.c')]
> >  ]
> >    if m[1].found()
> > diff --git a/audio/pwaudio.c b/audio/pwaudio.c
> > new file mode 100644
> > index 0000000000..aa22f40a80
> > --- /dev/null
> > +++ b/audio/pwaudio.c
> > @@ -0,0 +1,827 @@
> > +/*
> > + * QEMU Pipewire audio driver
> > + *
> > + * Copyright (c) 2023 Red Hat Inc.
> > + *
> > + * Author: Dorinda Bassey       <dbassey@redhat.com>
> > + *
> > + * Permission is hereby granted, free of charge, to any person
> obtaining a copy
> > + * of this software and associated documentation files (the
> "Software"), to deal
> > + * in the Software without restriction, including without limitation
> the rights
> > + * to use, copy, modify, merge, publish, distribute, sublicense, and/or
> sell
> > + * copies of the Software, and to permit persons to whom the Software is
> > + * furnished to do so, subject to the following conditions:
> > + *
> > + * The above copyright notice and this permission notice shall be
> included in
> > + * all copies or substantial portions of the Software.
> > + *
> > + * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND,
> EXPRESS OR
> > + * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF
> MERCHANTABILITY,
> > + * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT
> SHALL
> > + * THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR
> OTHER
> > + * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE,
> ARISING FROM,
> > + * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER
> DEALINGS IN
> > + * THE SOFTWARE.
>
> SPDX identifier instead would be neat
>
> > + */
> > +
> > +#include "qemu/osdep.h"
> > +#include "qemu/module.h"
> > +#include "audio.h"
> > +#include <errno.h>
> > +#include <spa/param/audio/format-utils.h>
> > +#include <spa/utils/ringbuffer.h>
> > +#include <spa/utils/result.h>
> > +
> > +#include <pipewire/pipewire.h>
> > +
> > +#define AUDIO_CAP "pipewire"
> > +#define RINGBUFFER_SIZE    (1u << 22)
>
> Why that much?
>
> > +#define RINGBUFFER_MASK    (RINGBUFFER_SIZE - 1)
> > +#define BUFFER_SAMPLES    512
> > +
> > +#include "audio_int.h"
> > +
> > +enum {
> > +    MODE_SINK,
> > +    MODE_SOURCE
> > +};
> > +
> > +typedef struct pwaudio {
> > +    Audiodev *dev;
> > +    struct pw_thread_loop *thread_loop;
> > +    struct pw_context *context;
> > +
> > +    struct pw_core *core;
> > +    struct spa_hook core_listener;
> > +    int seq;
> > +} pwaudio;
> > +
> > +typedef struct PWVoice {
> > +    pwaudio *g;
> > +    bool enabled;
> > +    struct pw_stream *stream;
> > +    struct spa_hook stream_listener;
> > +    struct spa_audio_info_raw info;
> > +    uint32_t frame_size;
> > +    struct spa_ringbuffer ring;
> > +    uint8_t buffer[RINGBUFFER_SIZE];
> > +
> > +    uint32_t mode;
> > +    struct pw_properties *props;
> > +} PWVoice;
> > +
> > +typedef struct PWVoiceOut {
> > +    HWVoiceOut hw;
> > +    PWVoice v;
> > +} PWVoiceOut;
> > +
> > +typedef struct PWVoiceIn {
> > +    HWVoiceIn hw;
> > +    PWVoice v;
> > +} PWVoiceIn;
> > +
> > +static void
> > +stream_destroy(void *data)
> > +{
> > +    PWVoice *v = (PWVoice *) data;
> > +    spa_hook_remove(&v->stream_listener);
> > +    v->stream = NULL;
> > +}
> > +
> > +/* output data processing function to read stuffs from the buffer */
> > +static void
> > +playback_on_process(void *data)
> > +{
> > +    PWVoice *v = (PWVoice *) data;
> > +    void *p;
> > +    struct pw_buffer *b;
> > +    struct spa_buffer *buf;
> > +    uint32_t n_frames, req, index, n_bytes;
> > +    int32_t avail;
> > +
> > +    if (!v->stream) {
> > +        return;
> > +    }
> > +
> > +    /* obtain a buffer to read from */
> > +    b = pw_stream_dequeue_buffer(v->stream);
> > +    if (b == NULL) {
> > +        pw_log_warn("out of buffers: %m");
> > +        return;
> > +    }
> > +
> > +    buf = b->buffer;
> > +    p = buf->datas[0].data;
> > +    if (p == NULL) {
> > +        return;
> > +    }
> > +    req = b->requested * v->frame_size;
> > +    if (req == 0) {
> > +        req = 4096 * v->frame_size;
> > +    }
> > +    n_frames = SPA_MIN(req, buf->datas[0].maxsize);
> > +    n_bytes = n_frames * v->frame_size;
> > +
> > +    /* get no of available bytes to read data from buffer */
> > +
> > +    avail = spa_ringbuffer_get_read_index(&v->ring, &index);
> > +
> > +    if (!v->enabled) {
> > +        avail = 0;
> > +    }
> > +
> > +    if (avail == 0) {
> > +        memset(p, 0, n_bytes);
> > +    } else {
> > +        if (avail < (int32_t) n_bytes) {
> > +            n_bytes = avail;
> > +        }
> > +
> > +        spa_ringbuffer_read_data(&v->ring,
> > +                                    v->buffer, RINGBUFFER_SIZE,
> > +                                    index & RINGBUFFER_MASK, p,
> n_bytes);
> > +
> > +        index += n_bytes;
> > +        spa_ringbuffer_read_update(&v->ring, index);
> > +    }
> > +
> > +    buf->datas[0].chunk->offset = 0;
> > +    buf->datas[0].chunk->stride = v->frame_size;
> > +    buf->datas[0].chunk->size = n_bytes;
> > +
> > +    /* queue the buffer for playback */
> > +    pw_stream_queue_buffer(v->stream, b);
> > +}
> > +
> > +/* output data processing function to generate stuffs in the buffer */
> > +static void
> > +capture_on_process(void *data)
> > +{
> > +    PWVoice *v = (PWVoice *) data;
> > +    void *p;
> > +    struct pw_buffer *b;
> > +    struct spa_buffer *buf;
> > +    int32_t filled;
> > +    uint32_t index, offs, n_bytes;
> > +
> > +    if (!v->stream) {
> > +        return;
> > +    }
> > +
> > +    /* obtain a buffer */
> > +    b = pw_stream_dequeue_buffer(v->stream);
> > +    if (b == NULL) {
> > +        pw_log_warn("out of buffers: %m");
> > +        return;
> > +    }
> > +
> > +    /* Write data into buffer */
> > +    buf = b->buffer;
> > +    p = buf->datas[0].data;
> > +    if (p == NULL) {
> > +        return;
> > +    }
> > +    offs = SPA_MIN(buf->datas[0].chunk->offset, buf->datas[0].maxsize);
> > +    n_bytes = SPA_MIN(buf->datas[0].chunk->size, buf->datas[0].maxsize
> - offs);
> > +
> > +    filled = spa_ringbuffer_get_write_index(&v->ring, &index);
> > +
> > +    if (!v->enabled) {
> > +        n_bytes = 0;
> > +    }
> > +
> > +    if (filled < 0) {
> > +        pw_log_warn("%p: underrun write:%u filled:%d", p, index,
> filled);
> > +    } else {
> > +        if ((uint32_t) filled + n_bytes > RINGBUFFER_SIZE) {
> > +            pw_log_warn("%p: overrun write:%u filled:%d + size:%u >
> max:%u",
> > +            p, index, filled, n_bytes, RINGBUFFER_SIZE);
> > +        }
> > +    }
> > +    spa_ringbuffer_write_data(&v->ring,
> > +                                v->buffer, RINGBUFFER_SIZE,
> > +                                index & RINGBUFFER_MASK,
> > +                                SPA_PTROFF(p, offs, void), n_bytes);
> > +    index += n_bytes;
> > +    spa_ringbuffer_write_update(&v->ring, index);
> > +
> > +    /* queue the buffer for playback */
> > +    pw_stream_queue_buffer(v->stream, b);
> > +}
> > +
> > +static void
> > +on_stream_state_changed(void *_data, enum pw_stream_state old,
> > +                        enum pw_stream_state state, const char *error)
> > +{
> > +    PWVoice *v = (PWVoice *) _data;
> > +
> > +    printf("stream state: \"%s\"\n", pw_stream_state_as_string(state));
> > +
> > +    switch (state) {
> > +    case PW_STREAM_STATE_ERROR:
> > +    case PW_STREAM_STATE_UNCONNECTED:
> > +        {
> > +            break;
> > +        }
> > +    case PW_STREAM_STATE_PAUSED:
> > +        printf("node id: %d\n", pw_stream_get_node_id(v->stream));
> > +        break;
> > +    case PW_STREAM_STATE_CONNECTING:
> > +    case PW_STREAM_STATE_STREAMING:
> > +        break;
> > +    }
> > +}
> > +
> > +static const struct pw_stream_events capture_stream_events = {
> > +    PW_VERSION_STREAM_EVENTS,
> > +    .destroy = stream_destroy,
> > +    .state_changed = on_stream_state_changed,
> > +    .process = capture_on_process
> > +};
> > +
> > +static const struct pw_stream_events playback_stream_events = {
> > +    PW_VERSION_STREAM_EVENTS,
> > +    .destroy = stream_destroy,
> > +    .state_changed = on_stream_state_changed,
> > +    .process = playback_on_process
> > +};
> > +
> > +static size_t
> > +qpw_read(HWVoiceIn *hw, void *data, size_t len)
> > +{
> > +    PWVoiceIn *pw = (PWVoiceIn *) hw;
> > +    PWVoice *v = &pw->v;
> > +    pwaudio *c = v->g;
> > +    const char *error = NULL;
> > +    size_t l;
> > +    int32_t avail;
> > +    uint32_t index;
> > +
> > +    pw_thread_loop_lock(c->thread_loop);
> > +    if (pw_stream_get_state(v->stream, &error) !=
> PW_STREAM_STATE_STREAMING) {
> > +        /* wait for stream to become ready */
> > +        l = 0;
> > +        goto done_unlock;
> > +    }
> > +    /* get no of available bytes to read data from buffer */
> > +    avail = spa_ringbuffer_get_read_index(&v->ring, &index);
> > +
> > +    if (avail < (int32_t) len) {
> > +        len = avail;
> > +    }
> > +
> > +    spa_ringbuffer_read_data(&v->ring,
> > +                             v->buffer, RINGBUFFER_SIZE,
> > +                             index & RINGBUFFER_MASK, data, len);
> > +    index += len;
> > +    spa_ringbuffer_read_update(&v->ring, index);
> > +    l = len;
> > +
> > +done_unlock:
> > +    pw_thread_loop_unlock(c->thread_loop);
> > +    return l;
> > +}
> > +
> > +static size_t
> > +qpw_write(HWVoiceOut *hw, void *data, size_t len)
> > +{
> > +    PWVoiceOut *pw = (PWVoiceOut *) hw;
> > +    PWVoice *v = &pw->v;
> > +    pwaudio *c = v->g;
> > +    const char *error = NULL;
> > +    const int periods = 3;
> > +    size_t l;
> > +    int32_t filled, avail;
> > +    uint32_t index;
> > +
> > +    pw_thread_loop_lock(c->thread_loop);
> > +    if (pw_stream_get_state(v->stream, &error) !=
> PW_STREAM_STATE_STREAMING) {
> > +        /* wait for stream to become ready */
> > +        l = 0;
> > +        goto done_unlock;
> > +    }
> > +    filled = spa_ringbuffer_get_write_index(&v->ring, &index);
> > +
> > +    avail = BUFFER_SAMPLES * v->frame_size * periods - filled;
> > +
> > +    pw_log_debug("%u %u %u %zu", filled, avail, index, len);
> > +
> > +    if (len > avail) {
> > +        len = avail;
> > +    }
> > +
> > +    if (filled < 0) {
> > +        pw_log_warn("%p: underrun write:%u filled:%d", pw, index,
> filled);
> > +    } else {
> > +        if ((uint32_t) filled + len > RINGBUFFER_SIZE) {
> > +            pw_log_warn("%p: overrun write:%u filled:%d + size:%zu >
> max:%u",
> > +            pw, index, filled, len, RINGBUFFER_SIZE);
> > +        }
> > +    }
> > +
> > +    spa_ringbuffer_write_data(&v->ring,
> > +                                v->buffer, RINGBUFFER_SIZE,
> > +                                index & RINGBUFFER_MASK, data, len);
>
> It looks like you don't take latency into account. Isn't there a more
> direct/simple pw push API that takes care of internal buffering? The
> intermediate ringbuffer is not nice to deal with.
>
> > +    index += len;
> > +    spa_ringbuffer_write_update(&v->ring, index);
> > +    l = len;
> > +
> > +done_unlock:
> > +    pw_thread_loop_unlock(c->thread_loop);
> > +    return l;
> > +}
> > +
> > +static int
> > +audfmt_to_pw(AudioFormat fmt, int endianness)
> > +{
> > +    int format;
> > +
> > +    switch (fmt) {
> > +    case AUDIO_FORMAT_S8:
> > +        format = SPA_AUDIO_FORMAT_S8;
> > +        break;
> > +    case AUDIO_FORMAT_U8:
> > +        format = SPA_AUDIO_FORMAT_U8;
> > +        break;
> > +    case AUDIO_FORMAT_S16:
> > +        format = endianness ? SPA_AUDIO_FORMAT_S16_BE :
> SPA_AUDIO_FORMAT_S16_LE;
> > +        break;
> > +    case AUDIO_FORMAT_U16:
> > +        format = endianness ? SPA_AUDIO_FORMAT_U16_BE :
> SPA_AUDIO_FORMAT_U16_LE;
> > +        break;
> > +    case AUDIO_FORMAT_S32:
> > +        format = endianness ? SPA_AUDIO_FORMAT_S32_BE :
> SPA_AUDIO_FORMAT_S32_LE;
> > +        break;
> > +    case AUDIO_FORMAT_U32:
> > +        format = endianness ? SPA_AUDIO_FORMAT_U32_BE :
> SPA_AUDIO_FORMAT_U32_LE;
> > +        break;
> > +    case AUDIO_FORMAT_F32:
> > +        format = endianness ? SPA_AUDIO_FORMAT_F32_BE :
> SPA_AUDIO_FORMAT_F32_LE;
> > +        break;
> > +    default:
> > +        dolog("Internal logic error: Bad audio format %d\n", fmt);
> > +        format = SPA_AUDIO_FORMAT_U8;
> > +        break;
> > +    }
> > +    return format;
> > +}
> > +
> > +static AudioFormat
> > +pw_to_audfmt(enum spa_audio_format fmt, int *endianness,
> > +             uint32_t *frame_size)
> > +{
> > +    switch (fmt) {
> > +    case SPA_AUDIO_FORMAT_S8:
> > +        *frame_size = 1;
> > +        return AUDIO_FORMAT_S8;
> > +    case SPA_AUDIO_FORMAT_U8:
> > +        *frame_size = 1;
> > +        return AUDIO_FORMAT_U8;
> > +    case SPA_AUDIO_FORMAT_S16_BE:
> > +        *frame_size = 2;
> > +        *endianness = 1;
> > +        return AUDIO_FORMAT_S16;
> > +    case SPA_AUDIO_FORMAT_S16_LE:
> > +        *frame_size = 2;
> > +        *endianness = 0;
> > +        return AUDIO_FORMAT_S16;
> > +    case SPA_AUDIO_FORMAT_U16_BE:
> > +        *frame_size = 2;
> > +        *endianness = 1;
> > +        return AUDIO_FORMAT_U16;
> > +    case SPA_AUDIO_FORMAT_U16_LE:
> > +        *frame_size = 2;
> > +        *endianness = 0;
> > +        return AUDIO_FORMAT_U16;
> > +    case SPA_AUDIO_FORMAT_S32_BE:
> > +        *frame_size = 4;
> > +        *endianness = 1;
> > +        return AUDIO_FORMAT_S32;
> > +    case SPA_AUDIO_FORMAT_S32_LE:
> > +        *frame_size = 4;
> > +        *endianness = 0;
> > +        return AUDIO_FORMAT_S32;
> > +    case SPA_AUDIO_FORMAT_U32_BE:
> > +        *frame_size = 4;
> > +        *endianness = 1;
> > +        return AUDIO_FORMAT_U32;
> > +    case SPA_AUDIO_FORMAT_U32_LE:
> > +        *frame_size = 4;
> > +        *endianness = 0;
> > +        return AUDIO_FORMAT_U32;
> > +    case SPA_AUDIO_FORMAT_F32_BE:
> > +        *frame_size = 4;
> > +        *endianness = 1;
> > +        return AUDIO_FORMAT_F32;
> > +    case SPA_AUDIO_FORMAT_F32_LE:
> > +        *frame_size = 4;
> > +        *endianness = 0;
> > +        return AUDIO_FORMAT_F32;
> > +    default:
> > +        *frame_size = 1;
> > +        dolog("Internal logic error: Bad spa_audio_format %d\n", fmt);
> > +        return AUDIO_FORMAT_U8;
> > +    }
> > +}
> > +
> > +static int
> > +create_stream(pwaudio *c, PWVoice *v, const char *name)
> > +{
> > +    int res;
> > +    uint32_t n_params;
> > +    const struct spa_pod *params[2];
> > +    uint8_t buffer[1024];
> > +    struct spa_pod_builder b;
> > +
> > +    v->stream = pw_stream_new(c->core, name, NULL);
> > +
> > +    if (v->stream == NULL) {
> > +        res = -errno;
> > +        goto error;
> > +    }
> > +
> > +    if (v->mode == MODE_SOURCE) {
> > +        pw_stream_add_listener(v->stream,
> > +                            &v->stream_listener,
> &capture_stream_events, v);
> > +    } else {
> > +        pw_stream_add_listener(v->stream,
> > +                            &v->stream_listener,
> &playback_stream_events, v);
> > +    }
> > +
> > +    n_params = 0;
> > +    spa_pod_builder_init(&b, buffer, sizeof(buffer));
> > +    params[n_params++] = spa_format_audio_raw_build(&b,
> > +                            SPA_PARAM_EnumFormat,
> > +                            &v->info);
> > +
> > +    /* connect the stream to a sink or source */
> > +    res = pw_stream_connect(v->stream,
> > +                            v->mode ==
> > +                            MODE_SOURCE ? PW_DIRECTION_INPUT :
> > +                            PW_DIRECTION_OUTPUT, PW_ID_ANY,
> > +                            PW_STREAM_FLAG_AUTOCONNECT |
> > +                            PW_STREAM_FLAG_MAP_BUFFERS |
> > +                            PW_STREAM_FLAG_RT_PROCESS, params,
> n_params);
> > +    if (res < 0) {
> > +        goto error;
> > +    }
> > +
> > +    return 0;
> > +error:
> > +    return res;
> > +}
> > +
> > +static void
> > +pw_destroy(pwaudio *c)
> > +{
> > +    if (c->thread_loop) {
> > +        pw_thread_loop_stop(c->thread_loop);
> > +    }
> > +    if (c->core) {
> > +        pw_core_disconnect(c->core);
> > +    }
> > +
> > +    g_free(c);
> > +}
> > +
> > +static int
> > +qpw_stream_new(pwaudio *c, PWVoice *v, const char *name)
> > +{
> > +    int r;
> > +
> > +    pw_thread_loop_lock(c->thread_loop);
> > +
> > +    switch (v->info.channels) {
> > +    case 8:
> > +        v->info.position[0] = SPA_AUDIO_CHANNEL_FL;
> > +        v->info.position[1] = SPA_AUDIO_CHANNEL_FR;
> > +        v->info.position[2] = SPA_AUDIO_CHANNEL_FC;
> > +        v->info.position[3] = SPA_AUDIO_CHANNEL_LFE;
> > +        v->info.position[4] = SPA_AUDIO_CHANNEL_RL;
> > +        v->info.position[5] = SPA_AUDIO_CHANNEL_RR;
> > +        v->info.position[6] = SPA_AUDIO_CHANNEL_SL;
> > +        v->info.position[7] = SPA_AUDIO_CHANNEL_SR;
> > +        break;
> > +    case 6:
> > +        v->info.position[0] = SPA_AUDIO_CHANNEL_FL;
> > +        v->info.position[1] = SPA_AUDIO_CHANNEL_FR;
> > +        v->info.position[2] = SPA_AUDIO_CHANNEL_FC;
> > +        v->info.position[3] = SPA_AUDIO_CHANNEL_LFE;
> > +        v->info.position[4] = SPA_AUDIO_CHANNEL_RL;
> > +        v->info.position[5] = SPA_AUDIO_CHANNEL_RR;
> > +        break;
> > +    case 5:
> > +        v->info.position[0] = SPA_AUDIO_CHANNEL_FL;
> > +        v->info.position[1] = SPA_AUDIO_CHANNEL_FR;
> > +        v->info.position[2] = SPA_AUDIO_CHANNEL_FC;
> > +        v->info.position[3] = SPA_AUDIO_CHANNEL_LFE;
> > +        v->info.position[4] = SPA_AUDIO_CHANNEL_RC;
> > +        break;
> > +    case 4:
> > +        v->info.position[0] = SPA_AUDIO_CHANNEL_FL;
> > +        v->info.position[1] = SPA_AUDIO_CHANNEL_FR;
> > +        v->info.position[2] = SPA_AUDIO_CHANNEL_FC;
> > +        v->info.position[3] = SPA_AUDIO_CHANNEL_RC;
> > +        break;
> > +    case 3:
> > +        v->info.position[0] = SPA_AUDIO_CHANNEL_FL;
> > +        v->info.position[1] = SPA_AUDIO_CHANNEL_FR;
> > +        v->info.position[2] = SPA_AUDIO_CHANNEL_LFE;
> > +        break;
> > +    case 2:
> > +        v->info.position[0] = SPA_AUDIO_CHANNEL_FL;
> > +        v->info.position[1] = SPA_AUDIO_CHANNEL_FR;
> > +        break;
> > +    case 1:
> > +        v->info.position[0] = SPA_AUDIO_CHANNEL_MONO;
> > +        break;
> > +    default:
> > +        for (size_t i = 0; i < v->info.channels; i++) {
> > +            v->info.position[i] = SPA_AUDIO_CHANNEL_UNKNOWN;
> > +        }
> > +        break;
> > +    }
> > +
> > +    /* create a new unconnected pwstream */
> > +    r = create_stream(c, v, name);
> > +    if (r < 0) {
> > +        goto error;
> > +    }
> > +
> > +    pw_thread_loop_unlock(c->thread_loop);
> > +    return r;
> > +
> > +error:
> > +    AUD_log(AUDIO_CAP, "Failed to create stream.");
> > +    pw_thread_loop_unlock(c->thread_loop);
> > +    pw_destroy(c);
> > +    return -1;
> > +}
> > +
> > +static int
> > +qpw_init_out(HWVoiceOut *hw, struct audsettings *as, void *drv_opaque)
> > +{
> > +    PWVoiceOut *pw = (PWVoiceOut *) hw;
> > +    PWVoice *v = &pw->v;
> > +    struct audsettings obt_as = *as;
> > +    pwaudio *c = v->g = drv_opaque;
> > +    AudiodevPipewireOptions *popts = &c->dev->u.pipewire;
> > +    AudiodevPipewirePerDirectionOptions *ppdo = popts->out;
> > +    int r;
> > +    v->enabled = false;
> > +
> > +    v->mode = MODE_SINK;
> > +
> > +    pw_thread_loop_lock(c->thread_loop);
> > +
> > +    v->info.format = audfmt_to_pw(as->fmt, as->endianness);
> > +    v->info.channels = as->nchannels;
> > +    v->info.rate = as->freq;
> > +
> > +    obt_as.fmt =
> > +        pw_to_audfmt(v->info.format, &obt_as.endianness,
> &v->frame_size);
> > +    v->frame_size *= as->nchannels;
> > +
> > +    /* call the function that creates a new stream for playback */
> > +    r = qpw_stream_new(c, v, ppdo->stream_name ? : c->dev->id);
> > +    if (r < 0) {
> > +        pw_log_error("qpw_stream_new for playback failed\n ");
> > +        goto fail;
> > +    }
> > +
> > +    /* report the audio format we support */
> > +    audio_pcm_init_info(&hw->info, &obt_as);
> > +
> > +    /* report the buffer size to qemu */
> > +    hw->samples = BUFFER_SAMPLES;
> > +
> > +    pw_thread_loop_unlock(c->thread_loop);
> > +    return 0;
> > +fail:
> > +    pw_thread_loop_unlock(c->thread_loop);
> > +    return -1;
> > +}
> > +
> > +static int
> > +qpw_init_in(HWVoiceIn *hw, struct audsettings *as, void *drv_opaque)
> > +{
> > +    PWVoiceIn *pw = (PWVoiceIn *) hw;
> > +    PWVoice *v = &pw->v;
> > +    struct audsettings obt_as = *as;
> > +    pwaudio *c = v->g = drv_opaque;
> > +    AudiodevPipewireOptions *popts = &c->dev->u.pipewire;
> > +    AudiodevPipewirePerDirectionOptions *ppdo = popts->in;
> > +    int r;
> > +    v->enabled = false;
> > +
> > +    v->mode = MODE_SOURCE;
> > +    pw_thread_loop_lock(c->thread_loop);
> > +
> > +    v->info.format = audfmt_to_pw(as->fmt, as->endianness);
> > +    v->info.channels = as->nchannels;
> > +    v->info.rate = as->freq;
> > +
> > +    obt_as.fmt =
> > +        pw_to_audfmt(v->info.format, &obt_as.endianness,
> &v->frame_size);
> > +    v->frame_size *= as->nchannels;
> > +
> > +    /* call the function that creates a new stream for recording */
> > +    r = qpw_stream_new(c, v, ppdo->stream_name ? : c->dev->id);
> > +    if (r < 0) {
> > +        pw_log_error("qpw_stream_new for recording failed\n ");
> > +        goto fail;
> > +    }
> > +
> > +    /* report the audio format we support */
> > +    audio_pcm_init_info(&hw->info, &obt_as);
> > +
> > +    /* report the buffer size to qemu */
> > +    hw->samples = BUFFER_SAMPLES;
> > +
> > +    pw_thread_loop_unlock(c->thread_loop);
> > +    return 0;
> > +fail:
> > +    pw_thread_loop_unlock(c->thread_loop);
> > +    return -1;
> > +}
> > +
> > +static void
> > +qpw_fini_out(HWVoiceOut *hw)
> > +{
> > +    PWVoiceOut *pw = (PWVoiceOut *) hw;
> > +    PWVoice *v = &pw->v;
> > +
> > +    if (v->stream) {
> > +        pwaudio *c = v->g;
> > +        pw_thread_loop_lock(c->thread_loop);
> > +        pw_stream_destroy(v->stream);
> > +        v->stream = NULL;
> > +        pw_thread_loop_unlock(c->thread_loop);
> > +    }
> > +}
> > +
> > +static void
> > +qpw_fini_in(HWVoiceIn *hw)
> > +{
> > +    PWVoiceIn *pw = (PWVoiceIn *) hw;
> > +    PWVoice *v = &pw->v;
> > +
> > +    if (v->stream) {
> > +        pwaudio *c = v->g;
> > +        pw_thread_loop_lock(c->thread_loop);
> > +        pw_stream_destroy(v->stream);
> > +        v->stream = NULL;
> > +        pw_thread_loop_unlock(c->thread_loop);
> > +    }
> > +}
> > +
> > +static void
> > +qpw_enable_out(HWVoiceOut *hw, bool enable)
> > +{
> > +    PWVoiceOut *po = (PWVoiceOut *) hw;
> > +    PWVoice *v = &po->v;
> > +    v->enabled = enable;
> > +}
> > +
> > +static void
> > +qpw_enable_in(HWVoiceIn *hw, bool enable)
> > +{
> > +    PWVoiceIn *pi = (PWVoiceIn *) hw;
> > +    PWVoice *v = &pi->v;
> > +    v->enabled = enable;
> > +}
> > +
> > +static void
> > +on_core_error(void *data, uint32_t id, int seq, int res, const char
> *message)
> > +{
> > +    pwaudio *pw = data;
> > +
> > +    pw_log_warn("error id:%u seq:%d res:%d (%s): %s",
> > +                id, seq, res, spa_strerror(res), message);
> > +
> > +    pw_thread_loop_signal(pw->thread_loop, FALSE);
> > +}
> > +
> > +static void
> > +on_core_done(void *data, uint32_t id, int seq)
> > +{
> > +    pwaudio *pw = data;
> > +    if (id == PW_ID_CORE) {
> > +        pw->seq = seq;
> > +        pw_thread_loop_signal(pw->thread_loop, FALSE);
> > +    }
> > +}
> > +
> > +static const struct pw_core_events core_events = {
> > +    PW_VERSION_CORE_EVENTS,
> > +    .done = on_core_done,
> > +    .error = on_core_error,
> > +};
> > +
> > +static void *
> > +qpw_audio_init(Audiodev *dev)
> > +{
> > +    pwaudio *pw;
> > +    pw = g_new0(pwaudio, 1);
> > +    pw_init(NULL, NULL);
> > +
> > +    AudiodevPipewireOptions *popts;
> > +    AUD_log(AUDIO_CAP, "Initialize PW context\n");
> > +    assert(dev->driver == AUDIODEV_DRIVER_PIPEWIRE);
> > +    popts = &dev->u.pipewire;
> > +
> > +    if (!popts->has_latency) {
> > +        popts->has_latency = true;
> > +        popts->latency = 15000;
> > +    }
> > +
> > +    pw->dev = dev;
> > +    pw->thread_loop = pw_thread_loop_new("Pipewire thread loop", NULL);
> > +    if (pw->thread_loop == NULL) {
> > +        goto fail;
> > +    }
> > +    pw->context =
> > +        pw_context_new(pw_thread_loop_get_loop(pw->thread_loop), NULL,
> 0);
> > +
> > +    if (pw_thread_loop_start(pw->thread_loop) < 0) {
> > +        goto fail;
> > +    }
> > +
> > +    pw_thread_loop_lock(pw->thread_loop);
> > +
> > +    pw->core = pw_context_connect(pw->context, NULL, 0);
> > +    if (pw->core == NULL) {
> > +        goto fail;
> > +    }
> > +
> > +    pw_core_add_listener(pw->core, &pw->core_listener, &core_events,
> pw);
> > +
> > +    pw_thread_loop_unlock(pw->thread_loop);
> > +
> > +    return pw;
> > +
> > +fail:
> > +    AUD_log(AUDIO_CAP, "Failed to initialize PW context");
> > +    pw_thread_loop_unlock(pw->thread_loop);
> > +    pw_context_destroy(pw->context);
> > +    pw_thread_loop_destroy(pw->thread_loop);
> > +    g_free(pw);
> > +    return NULL;
> > +}
> > +
> > +static void
> > +qpw_audio_fini(void *opaque)
> > +{
> > +    pwaudio *pw = opaque;
> > +
> > +    pw_thread_loop_stop(pw->thread_loop);
> > +
> > +    if (pw->core) {
> > +        spa_hook_remove(&pw->core_listener);
> > +        spa_zero(pw->core_listener);
> > +        pw_core_disconnect(pw->core);
> > +    }
> > +
> > +    if (pw->context) {
> > +        pw_context_destroy(pw->context);
> > +    }
> > +    pw_thread_loop_destroy(pw->thread_loop);
> > +
> > +    g_free(pw);
> > +}
> > +
> > +static struct audio_pcm_ops qpw_pcm_ops = {
> > +    .init_out = qpw_init_out,
> > +    .fini_out = qpw_fini_out,
> > +    .write = qpw_write,
> > +    .buffer_get_free = audio_generic_buffer_get_free,
> > +    .run_buffer_out = audio_generic_run_buffer_out,
> > +    .enable_out = qpw_enable_out,
> > +
> > +    .init_in = qpw_init_in,
> > +    .fini_in = qpw_fini_in,
> > +    .read = qpw_read,
> > +    .run_buffer_in = audio_generic_run_buffer_in,
> > +    .enable_in = qpw_enable_in
> > +};
> > +
> > +static struct audio_driver pw_audio_driver = {
> > +    .name = "pipewire",
> > +    .descr = "http://www.pipewire.org/",
> > +    .init = qpw_audio_init,
> > +    .fini = qpw_audio_fini,
> > +    .pcm_ops = &qpw_pcm_ops,
> > +    .can_be_default = 1,
> > +    .max_voices_out = INT_MAX,
> > +    .max_voices_in = INT_MAX,
> > +    .voice_size_out = sizeof(PWVoiceOut),
> > +    .voice_size_in = sizeof(PWVoiceIn),
> > +};
> > +
> > +static void
> > +register_audio_pw(void)
> > +{
> > +    audio_driver_register(&pw_audio_driver);
> > +}
> > +
> > +type_init(register_audio_pw);
> > diff --git a/meson.build b/meson.build
> > index a76c855312..eafe068044 100644
> > --- a/meson.build
> > +++ b/meson.build
> > @@ -734,6 +734,12 @@ if not get_option('jack').auto() or have_system
> >    jack = dependency('jack', required: get_option('jack'),
> >                      method: 'pkg-config', kwargs: static_kwargs)
> >  endif
> > +pipewire = not_found
> > +if not get_option('pipewire').auto() or (targetos == 'linux' and
> have_system)
> > +  pipewire = dependency('libpipewire-0.3', version: '>=0.3.60',
> > +                    required: get_option('pipewire'),
> > +                    method: 'pkg-config', kwargs: static_kwargs)
> > +endif
> >  sndio = not_found
> >  if not get_option('sndio').auto() or have_system
> >    sndio = dependency('sndio', required: get_option('sndio'),
> > @@ -1671,6 +1677,7 @@ if have_system
> >      'jack': jack.found(),
> >      'oss': oss.found(),
> >      'pa': pulse.found(),
> > +    'pipewire': pipewire.found(),
> >      'sdl': sdl.found(),
> >      'sndio': sndio.found(),
> >    }
> > @@ -3949,6 +3956,7 @@ endif
> >  if targetos == 'linux'
> >    summary_info += {'ALSA support':    alsa}
> >    summary_info += {'PulseAudio support': pulse}
> > +  summary_info += {'Pipewire support':   pipewire}
>
> Since it is not linux specific, no need to put it there. (filed
> alignment is kind of pointless to me)
>
> >  endif
> >  summary_info += {'JACK support':      jack}
> >  summary_info += {'brlapi support':    brlapi}
> > diff --git a/meson_options.txt b/meson_options.txt
> > index 7e5801db90..1b7847250d 100644
> > --- a/meson_options.txt
> > +++ b/meson_options.txt
> > @@ -21,7 +21,7 @@ option('tls_priority', type : 'string', value :
> 'NORMAL',
> >  option('default_devices', type : 'boolean', value : true,
> >         description: 'Include a default selection of devices in
> emulators')
> >  option('audio_drv_list', type: 'array', value: ['default'],
> > -       choices: ['alsa', 'coreaudio', 'default', 'dsound', 'jack',
> 'oss', 'pa', 'sdl', 'sndio'],
> > +       choices: ['alsa', 'coreaudio', 'default', 'dsound', 'jack',
> 'oss', 'pa', 'pipewire', 'sdl', 'sndio'],
> >         description: 'Set audio driver list')
> >  option('block_drv_rw_whitelist', type : 'string', value : '',
> >         description: 'set block driver read-write whitelist (by default
> affects only QEMU, not tools like qemu-img)')
> > @@ -255,6 +255,8 @@ option('oss', type: 'feature', value: 'auto',
> >         description: 'OSS sound support')
> >  option('pa', type: 'feature', value: 'auto',
> >         description: 'PulseAudio sound support')
> > +option('pipewire', type: 'feature', value: 'auto',
> > +       description: 'Pipewire sound support')
> >  option('sndio', type: 'feature', value: 'auto',
> >         description: 'sndio sound support')
> >
> > diff --git a/qapi/audio.json b/qapi/audio.json
> > index 4e54c00f51..b872e9f10d 100644
> > --- a/qapi/audio.json
> > +++ b/qapi/audio.json
> > @@ -324,6 +324,48 @@
> >      '*out':    'AudiodevPaPerDirectionOptions',
> >      '*server': 'str' } }
> >
> > +##
> > +# @AudiodevPipewirePerDirectionOptions:
> > +#
> > +# Options of the Pipewire backend that are used for both playback and
> > +# recording.
> > +#
> > +# @name: name of the sink/source to use
> > +#
> > +# @stream-name: name of the Pipewire stream created by qemu.  Can be
> > +#               used to identify the stream in Pipewire when you
> > +#               create multiple Pipewire devices or run multiple qemu
> > +#               instances (default: audiodev's id, since 7.1)
> > +#
> > +#
> > +# Since: 7.2
>
> Should be 8.0
>
> > +##
> > +{ 'struct': 'AudiodevPipewirePerDirectionOptions',
> > +  'base': 'AudiodevPerDirectionOptions',
> > +  'data': {
> > +    '*name': 'str',
> > +    '*stream-name': 'str' } }
> > +
> > +##
> > +# @AudiodevPipewireOptions:
> > +#
> > +# Options of the Pipewire audio backend.
> > +#
> > +# @in: options of the capture stream
> > +#
> > +# @out: options of the playback stream
> > +#
> > +# @latency: add latency to playback in microseconds
> > +#           (default 44100)
>
> Where is this latency value coming from?
>
> > +#
> > +# Since: 7.2
>
> 8.0
>
>
> > +##
> > +{ 'struct': 'AudiodevPipewireOptions',
> > +  'data': {
> > +    '*in':     'AudiodevPipewirePerDirectionOptions',
> > +    '*out':    'AudiodevPipewirePerDirectionOptions',
> > +    '*latency': 'uint32' } }
> > +
> >  ##
> >  # @AudiodevSdlPerDirectionOptions:
> >  #
> > @@ -416,6 +458,7 @@
> >              { 'name': 'jack', 'if': 'CONFIG_AUDIO_JACK' },
> >              { 'name': 'oss', 'if': 'CONFIG_AUDIO_OSS' },
> >              { 'name': 'pa', 'if': 'CONFIG_AUDIO_PA' },
> > +            { 'name': 'pipewire', 'if': 'CONFIG_AUDIO_PIPEWIRE' },
> >              { 'name': 'sdl', 'if': 'CONFIG_AUDIO_SDL' },
> >              { 'name': 'sndio', 'if': 'CONFIG_AUDIO_SNDIO' },
> >              { 'name': 'spice', 'if': 'CONFIG_SPICE' },
> > @@ -456,6 +499,8 @@
> >                     'if': 'CONFIG_AUDIO_OSS' },
> >      'pa':        { 'type': 'AudiodevPaOptions',
> >                     'if': 'CONFIG_AUDIO_PA' },
> > +    'pipewire':  { 'type': 'AudiodevPipewireOptions',
> > +                   'if': 'CONFIG_AUDIO_PIPEWIRE' },
> >      'sdl':       { 'type': 'AudiodevSdlOptions',
> >                     'if': 'CONFIG_AUDIO_SDL' },
> >      'sndio':     { 'type': 'AudiodevSndioOptions',
> > diff --git a/qemu-options.hx b/qemu-options.hx
> > index cafd8be8ed..95ed5e5c2d 100644
> > --- a/qemu-options.hx
> > +++ b/qemu-options.hx
> > @@ -779,6 +779,11 @@ DEF("audiodev", HAS_ARG, QEMU_OPTION_audiodev,
> >      "                in|out.name= source/sink device name\n"
> >      "                in|out.latency= desired latency in microseconds\n"
> >  #endif
> > +#ifdef CONFIG_AUDIO_PIPEWIRE
> > +    "-audiodev pipewire,id=id[,prop[=value][,...]]\n"
> > +    "                in|out.name= source/sink device name\n"
> > +    "                latency= desired latency in microseconds\n"
> > +#endif
> >  #ifdef CONFIG_AUDIO_SDL
> >      "-audiodev sdl,id=id[,prop[=value][,...]]\n"
> >      "                in|out.buffer-count= number of buffers\n"
> > @@ -942,6 +947,18 @@ SRST
> >          Desired latency in microseconds. The PulseAudio server will try
> >          to honor this value but actual latencies may be lower or higher.
> >
> > +``-audiodev pipewire,id=id[,prop[=value][,...]]``
> > +    Creates a backend using Pipewire. This backend is available on
> > +    most systems.
> > +
> > +    Pipewire specific options are:
> > +
> > +    ``latency=latency``
> > +        Add extra latency to playback in microseconds
> > +
> > +    ``in|out.name=sink``
> > +        Use the specified source/sink for recording/playback.
> > +
> >  ``-audiodev sdl,id=id[,prop[=value][,...]]``
> >      Creates a backend using SDL. This backend is available on most
> >      systems, but you should use your platform's native backend if
> > diff --git a/scripts/meson-buildoptions.sh
> b/scripts/meson-buildoptions.sh
> > index 939cc114dd..05c0090118 100644
> > --- a/scripts/meson-buildoptions.sh
> > +++ b/scripts/meson-buildoptions.sh
> > @@ -1,7 +1,8 @@
> >  # This file is generated by meson-buildoptions.py, do not edit!
> >  meson_options_help() {
> > -  printf "%s\n" '  --audio-drv-list=CHOICES Set audio driver list
> [default] (choices: alsa/co'
> > -  printf "%s\n" '
>  reaudio/default/dsound/jack/oss/pa/sdl/sndio)'
> > +  printf "%s\n" '  --audio-drv-list=CHOICES Set audio driver list
> [default] (choices: al'
> > +  printf "%s\n" '
>  sa/coreaudio/default/dsound/jack/oss/pa/'
> > +  printf "%s\n" '                           pipewire/sdl/sndio)'
> >    printf "%s\n" '  --block-drv-ro-whitelist=VALUE'
> >    printf "%s\n" '                           set block driver read-only
> whitelist (by default'
> >    printf "%s\n" '                           affects only QEMU, not
> tools like qemu-img)'
> > @@ -135,6 +136,7 @@ meson_options_help() {
> >    printf "%s\n" '  oss             OSS sound support'
> >    printf "%s\n" '  pa              PulseAudio sound support'
> >    printf "%s\n" '  parallels       parallels image format support'
> > +  printf "%s\n" '  pipewire        Pipewire sound support'
> >    printf "%s\n" '  png             PNG support with libpng'
> >    printf "%s\n" '  pvrdma          Enable PVRDMA support'
> >    printf "%s\n" '  qcow1           qcow1 image format support'
> > @@ -370,6 +372,8 @@ _meson_option_parse() {
> >      --disable-pa) printf "%s" -Dpa=disabled ;;
> >      --enable-parallels) printf "%s" -Dparallels=enabled ;;
> >      --disable-parallels) printf "%s" -Dparallels=disabled ;;
> > +    --enable-pipewire) printf "%s" -Dpipewire=enabled ;;
> > +    --disable-pipewire) printf "%s" -Dpipewire=disabled ;;
> >      --with-pkgversion=*) quote_sh "-Dpkgversion=$2" ;;
> >      --enable-png) printf "%s" -Dpng=enabled ;;
> >      --disable-png) printf "%s" -Dpng=disabled ;;
> > --
> > 2.39.1
> >
> >
>
>
> --
> Marc-André Lureau
>
>

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