[v10] audio/pwaudio.c: Add Pipewire audio backend for QEMU

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>
---
v10:
improve error handling
fix volume functions
add locks in enable_in out functions
cleanup in reverse order of intialization
add triggers for the sync method on the core object
add waiting function for pw_thread_loop_signal
improve trace

 audio/audio.c                 |   3 +
 audio/audio_template.h        |   4 +
 audio/meson.build             |   1 +
 audio/pwaudio.c               | 906 ++++++++++++++++++++++++++++++++++
 audio/trace-events            |   8 +
 meson.build                   |   8 +
 meson_options.txt             |   4 +-
 qapi/audio.json               |  44 ++
 qemu-options.hx               |  21 +
 scripts/meson-buildoptions.sh |   8 +-
 10 files changed, 1004 insertions(+), 3 deletions(-)
 create mode 100644 audio/pwaudio.c

Hi Volker, Marc.

I have spent a significant amount of time revising the patchset and I'm
eager to see them included in the project. I understand that reviewing the
patches can be a time-consuming process and I appreciate the effort you've
put into providing feedback and guiding me through the revision process.
However I would appreciate any information you can provide on the expected
timeline for merging the patches. Let me know if there's anything else I
can do to help move this process forward.

Regards,
Dorinda.

On Mon, Apr 3, 2023 at 10:21 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>
> ---
> v10:
> improve error handling
> fix volume functions
> add locks in enable_in out functions
> cleanup in reverse order of intialization
> add triggers for the sync method on the core object
> add waiting function for pw_thread_loop_signal
> improve trace
>
>  audio/audio.c                 |   3 +
>  audio/audio_template.h        |   4 +
>  audio/meson.build             |   1 +
>  audio/pwaudio.c               | 906 ++++++++++++++++++++++++++++++++++
>  audio/trace-events            |   8 +
>  meson.build                   |   8 +
>  meson_options.txt             |   4 +-
>  qapi/audio.json               |  44 ++
>  qemu-options.hx               |  21 +
>  scripts/meson-buildoptions.sh |   8 +-
>  10 files changed, 1004 insertions(+), 3 deletions(-)
>  create mode 100644 audio/pwaudio.c
>
> diff --git a/audio/audio.c b/audio/audio.c
> index 70b096713c..90c7c49d11 100644
> --- a/audio/audio.c
> +++ b/audio/audio.c
> @@ -2061,6 +2061,9 @@ 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 e42326c20d..dc0c74aa74 100644
> --- a/audio/audio_template.h
> +++ b/audio/audio_template.h
> @@ -362,6 +362,10 @@ 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..f9da86059f
> --- /dev/null
> +++ b/audio/pwaudio.c
> @@ -0,0 +1,906 @@
> +/*
> + * QEMU Pipewire audio driver
> + *
> + * Copyright (c) 2023 Red Hat Inc.
> + *
> + * Author: Dorinda Bassey       <dbassey@redhat.com>
> + *
> + * SPDX-License-Identifier: GPL-2.0-or-later
> + */
> +
> +#include "qemu/osdep.h"
> +#include "qemu/module.h"
> +#include "audio.h"
> +#include <errno.h>
> +#include "qemu/error-report.h"
> +#include <spa/param/audio/format-utils.h>
> +#include <spa/utils/ringbuffer.h>
> +#include <spa/utils/result.h>
> +#include <spa/param/props.h>
> +
> +#include <pipewire/pipewire.h>
> +#include "trace.h"
> +
> +#define AUDIO_CAP "pipewire"
> +#define RINGBUFFER_SIZE    (1u << 22)
> +#define RINGBUFFER_MASK    (RINGBUFFER_SIZE - 1)
> +
> +#include "audio_int.h"
> +
> +typedef struct pwvolume {
> +    uint32_t channels;
> +    float values[SPA_AUDIO_MAX_CHANNELS];
> +} pwvolume;
> +
> +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 last_seq, pending_seq, error;
> +} pwaudio;
> +
> +typedef struct PWVoice {
> +    pwaudio *g;
> +    struct pw_stream *stream;
> +    struct spa_hook stream_listener;
> +    struct spa_audio_info_raw info;
> +    uint32_t highwater_mark;
> +    uint32_t frame_size, req;
> +    struct spa_ringbuffer ring;
> +    uint8_t buffer[RINGBUFFER_SIZE];
> +
> +    struct pw_properties *props;
> +    pwvolume volume;
> +    bool muted;
> +} 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 req, index, n_bytes;
> +    int32_t avail;
> +
> +    assert(v->stream);
> +
> +    /* obtain a buffer to read from */
> +    b = pw_stream_dequeue_buffer(v->stream);
> +    if (b == NULL) {
> +        error_report("out of buffers: %s", strerror(errno));
> +        return;
> +    }
> +
> +    buf = b->buffer;
> +    p = buf->datas[0].data;
> +    if (p == NULL) {
> +        return;
> +    }
> +    /* calculate the total no of bytes to read data from buffer */
> +    req = b->requested * v->frame_size;
> +    if (req == 0) {
> +        req = v->req;
> +    }
> +    n_bytes = SPA_MIN(req, buf->datas[0].maxsize);
> +
> +    /* get no of available bytes to read data from buffer */
> +
> +    avail = spa_ringbuffer_get_read_index(&v->ring, &index);
> +
> +    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;
> +
> +    assert(v->stream);
> +
> +    /* obtain a buffer */
> +    b = pw_stream_dequeue_buffer(v->stream);
> +    if (b == NULL) {
> +        error_report("out of buffers: %s", strerror(errno));
> +        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 (filled < 0) {
> +        error_report("%p: underrun write:%u filled:%d", p, index, filled);
> +    } else {
> +        if ((uint32_t) filled + n_bytes > RINGBUFFER_SIZE) {
> +            error_report("%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;
> +
> +    trace_pw_state_changed(pw_stream_get_node_id(v->stream),
> +                           pw_stream_state_as_string(state));
> +
> +    switch (state) {
> +    case PW_STREAM_STATE_ERROR:
> +    case PW_STREAM_STATE_UNCONNECTED:
> +        break;
> +    case PW_STREAM_STATE_PAUSED:
> +    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);
> +
> +    trace_pw_read(avail, index, len);
> +
> +    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_buffer_get_free(HWVoiceOut *hw)
> +{
> +    PWVoiceOut *pw = (PWVoiceOut *)hw;
> +    PWVoice *v = &pw->v;
> +    pwaudio *c = v->g;
> +    const char *error = NULL;
> +    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 */
> +        avail = 0;
> +        goto done_unlock;
> +    }
> +
> +    filled = spa_ringbuffer_get_write_index(&v->ring, &index);
> +    avail = v->highwater_mark - filled;
> +
> +done_unlock:
> +    pw_thread_loop_unlock(c->thread_loop);
> +    return avail;
> +}
> +
> +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;
> +    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 */
> +        len = 0;
> +        goto done_unlock;
> +    }
> +    filled = spa_ringbuffer_get_write_index(&v->ring, &index);
> +    avail = v->highwater_mark - filled;
> +
> +    trace_pw_write(filled, avail, index, len);
> +
> +    if (len > avail) {
> +        len = avail;
> +    }
> +
> +    if (filled < 0) {
> +        error_report("%p: underrun write:%u filled:%d", pw, index,
> filled);
> +    } else {
> +        if ((uint32_t) filled + len > RINGBUFFER_SIZE) {
> +            error_report("%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);
> +    index += len;
> +    spa_ringbuffer_write_update(&v->ring, index);
> +
> +done_unlock:
> +    pw_thread_loop_unlock(c->thread_loop);
> +    return len;
> +}
> +
> +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 *stream_name,
> +              const char *name, enum spa_direction dir)
> +{
> +    int res;
> +    uint32_t n_params;
> +    const struct spa_pod *params[2];
> +    uint8_t buffer[1024];
> +    struct spa_pod_builder b;
> +    uint64_t buf_samples;
> +
> +    v->props = pw_properties_new(NULL, NULL);
> +
> +    /* 75% of the timer period for faster updates */
> +    buf_samples = (uint64_t)v->g->dev->timer_period * v->info.rate
> +                    * 3 / 4 / 1000000;
> +    trace_pw_timer(buf_samples);
> +    pw_properties_setf(v->props, PW_KEY_NODE_LATENCY, "%" PRIu64 "/%u",
> +                       buf_samples, v->info.rate);
> +
> +    if (name) {
> +        pw_properties_set(v->props, PW_KEY_TARGET_OBJECT, name);
> +    }
> +    v->stream = pw_stream_new(c->core, stream_name, v->props);
> +
> +    if (v->stream == NULL) {
> +        return -1;
> +    }
> +
> +    if (dir == SPA_DIRECTION_INPUT) {
> +        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,
> +                            dir ==
> +                            SPA_DIRECTION_INPUT ? PW_DIRECTION_INPUT :
> +                            PW_DIRECTION_OUTPUT, PW_ID_ANY,
> +                            PW_STREAM_FLAG_AUTOCONNECT |
> +                            PW_STREAM_FLAG_INACTIVE |
> +                            PW_STREAM_FLAG_MAP_BUFFERS |
> +                            PW_STREAM_FLAG_RT_PROCESS, params, n_params);
> +    if (res < 0) {
> +        pw_stream_destroy(v->stream);
> +        return -1;
> +    }
> +
> +    return 0;
> +}
> +
> +static int
> +qpw_stream_new(pwaudio *c, PWVoice *v, const char *stream_name,
> +               const char *name, enum spa_direction dir)
> +{
> +    int r;
> +
> +    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, stream_name, name, dir);
> +    if (r < 0) {
> +        AUD_log(AUDIO_CAP, "Failed to create stream.");
> +        return -1;
> +    }
> +
> +    return r;
> +}
> +
> +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;
> +
> +    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;
> +
> +    v->req = (uint64_t)c->dev->timer_period * v->info.rate
> +        * 1 / 2 / 1000000 * v->frame_size;
> +
> +    /* call the function that creates a new stream for playback */
> +    r = qpw_stream_new(c, v, ppdo->stream_name ? : c->dev->id,
> +                       ppdo->name, SPA_DIRECTION_OUTPUT);
> +    if (r < 0) {
> +        error_report("qpw_stream_new for playback failed");
> +        pw_thread_loop_unlock(c->thread_loop);
> +        return -1;
> +    }
> +
> +    /* report the audio format we support */
> +    audio_pcm_init_info(&hw->info, &obt_as);
> +
> +    /* report the buffer size to qemu */
> +    hw->samples = audio_buffer_frames(
> +        qapi_AudiodevPipewirePerDirectionOptions_base(ppdo), &obt_as,
> 46440);
> +    v->highwater_mark = MIN(RINGBUFFER_SIZE,
> +                            (ppdo->has_latency ? ppdo->latency : 46440)
> +                            * (uint64_t)v->info.rate / 1000000 *
> v->frame_size);
> +
> +    pw_thread_loop_unlock(c->thread_loop);
> +    return 0;
> +}
> +
> +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;
> +
> +    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,
> +                       ppdo->name, SPA_DIRECTION_INPUT);
> +    if (r < 0) {
> +        error_report("qpw_stream_new for recording failed");
> +        pw_thread_loop_unlock(c->thread_loop);
> +        return -1;
> +    }
> +
> +    /* report the audio format we support */
> +    audio_pcm_init_info(&hw->info, &obt_as);
> +
> +    /* report the buffer size to qemu */
> +    hw->samples = audio_buffer_frames(
> +        qapi_AudiodevPipewirePerDirectionOptions_base(ppdo), &obt_as,
> 46440);
> +
> +    pw_thread_loop_unlock(c->thread_loop);
> +    return 0;
> +}
> +
> +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;
> +    pwaudio *c = v->g;
> +    pw_thread_loop_lock(c->thread_loop);
> +    pw_stream_set_active(v->stream, enable);
> +    pw_thread_loop_unlock(c->thread_loop);
> +}
> +
> +static void
> +qpw_enable_in(HWVoiceIn *hw, bool enable)
> +{
> +    PWVoiceIn *pi = (PWVoiceIn *) hw;
> +    PWVoice *v = &pi->v;
> +    pwaudio *c = v->g;
> +    pw_thread_loop_lock(c->thread_loop);
> +    pw_stream_set_active(v->stream, enable);
> +    pw_thread_loop_unlock(c->thread_loop);
> +}
> +
> +static void
> +qpw_volume_out(HWVoiceOut *hw, Volume *vol)
> +{
> +    PWVoiceOut *pw = (PWVoiceOut *) hw;
> +    PWVoice *v = &pw->v;
> +    pwaudio *c = v->g;
> +    int i, ret;
> +
> +    pw_thread_loop_lock(c->thread_loop);
> +    v->volume.channels = vol->channels;
> +
> +    for (i = 0; i < vol->channels; ++i) {
> +        v->volume.values[i] = (float)vol->vol[i] / 255;
> +    }
> +
> +    ret = pw_stream_set_control(v->stream,
> +        SPA_PROP_channelVolumes, v->volume.channels, v->volume.values, 0);
> +    trace_pw_vol(ret == 0 ? "success" : "failed");
> +
> +    v->muted = vol->mute;
> +    float val = v->muted ? 1.f : 0.f;
> +    ret = pw_stream_set_control(v->stream, SPA_PROP_mute, 1, &val, 0);
> +    pw_thread_loop_unlock(c->thread_loop);
> +}
> +
> +static void
> +qpw_volume_in(HWVoiceIn *hw, Volume *vol)
> +{
> +    PWVoiceIn *pw = (PWVoiceIn *) hw;
> +    PWVoice *v = &pw->v;
> +    pwaudio *c = v->g;
> +    int i, ret;
> +
> +    pw_thread_loop_lock(c->thread_loop);
> +    v->volume.channels = vol->channels;
> +
> +    for (i = 0; i < vol->channels; ++i) {
> +        v->volume.values[i] = (float)vol->vol[i] / 255;
> +    }
> +
> +    ret = pw_stream_set_control(v->stream,
> +        SPA_PROP_channelVolumes, v->volume.channels, v->volume.values, 0);
> +    trace_pw_vol(ret == 0 ? "success" : "failed");
> +
> +    v->muted = vol->mute;
> +    float val = v->muted ? 1.f : 0.f;
> +    ret = pw_stream_set_control(v->stream, SPA_PROP_mute, 1, &val, 0);
> +    pw_thread_loop_unlock(c->thread_loop);
> +}
> +
> +static int wait_resync(pwaudio *pw)
> +{
> +    int res;
> +    pw->pending_seq = pw_core_sync(pw->core, PW_ID_CORE, pw->pending_seq);
> +
> +    while (true) {
> +        pw_thread_loop_wait(pw->thread_loop);
> +
> +        res = pw->error;
> +        if (res < 0) {
> +            pw->error = 0;
> +            return res;
> +        }
> +        if (pw->pending_seq == pw->last_seq) {
> +            break;
> +        }
> +    }
> +    return 0;
> +}
> +static void
> +on_core_error(void *data, uint32_t id, int seq, int res, const char
> *message)
> +{
> +    pwaudio *pw = data;
> +
> +    error_report("error id:%u seq:%d res:%d (%s): %s",
> +                id, seq, res, spa_strerror(res), message);
> +
> +    /* stop and exit the thread loop */
> +    pw_thread_loop_signal(pw->thread_loop, FALSE);
> +}
> +
> +static void
> +on_core_done(void *data, uint32_t id, int seq)
> +{
> +    pwaudio *pw = data;
> +    assert(id == PW_ID_CORE);
> +    pw->last_seq = seq;
> +    if (pw->pending_seq == seq) {
> +        /* stop and exit the thread loop */
> +        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)
> +{
> +    g_autofree pwaudio *pw = g_new0(pwaudio, 1);
> +    pw_init(NULL, NULL);
> +
> +    trace_pw_audio_init();
> +    assert(dev->driver == AUDIODEV_DRIVER_PIPEWIRE);
> +
> +    pw->dev = dev;
> +    pw->thread_loop = pw_thread_loop_new("Pipewire thread loop", NULL);
> +    if (pw->thread_loop == NULL) {
> +        error_report("Could not create Pipewire loop");
> +        goto fail;
> +    }
> +
> +    pw->context =
> +        pw_context_new(pw_thread_loop_get_loop(pw->thread_loop), NULL, 0);
> +    if (pw->context == NULL) {
> +        error_report("Could not create Pipewire context");
> +        goto fail;
> +    }
> +
> +    if (pw_thread_loop_start(pw->thread_loop) < 0) {
> +        error_report("Could not start Pipewire loop");
> +        goto fail;
> +    }
> +
> +    pw_thread_loop_lock(pw->thread_loop);
> +
> +    pw->core = pw_context_connect(pw->context, NULL, 0);
> +    if (pw->core == NULL) {
> +        pw_thread_loop_unlock(pw->thread_loop);
> +        goto fail;
> +    }
> +
> +    if (pw_core_add_listener(pw->core, &pw->core_listener,
> +                             &core_events, pw) < 0) {
> +        pw_thread_loop_unlock(pw->thread_loop);
> +        goto fail;
> +    }
> +    if (wait_resync(pw) < 0) {
> +        pw_thread_loop_unlock(pw->thread_loop);
> +    }
> +
> +    pw_thread_loop_unlock(pw->thread_loop);
> +
> +    return g_steal_pointer(&pw);
> +
> +fail:
> +    AUD_log(AUDIO_CAP, "Failed to initialize PW context");
> +    if (pw->thread_loop) {
> +        pw_thread_loop_stop(pw->thread_loop);
> +    }
> +    if (pw->context) {
> +        g_clear_pointer(&pw->context, pw_context_destroy);
> +    }
> +    if (pw->thread_loop) {
> +        g_clear_pointer(&pw->thread_loop, pw_thread_loop_destroy);
> +    }
> +    return NULL;
> +}
> +
> +static void
> +qpw_audio_fini(void *opaque)
> +{
> +    pwaudio *pw = opaque;
> +
> +    if (pw->thread_loop) {
> +        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 = qpw_buffer_get_free,
> +    .run_buffer_out = audio_generic_run_buffer_out,
> +    .enable_out = qpw_enable_out,
> +    .volume_out = qpw_volume_out,
> +    .volume_in = qpw_volume_in,
> +
> +    .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/audio/trace-events b/audio/trace-events
> index e1ab643add..d6c36139e5 100644
> --- a/audio/trace-events
> +++ b/audio/trace-events
> @@ -18,6 +18,14 @@ dbus_audio_register(const char *s, const char *dir)
> "sender = %s, dir = %s"
>  dbus_audio_put_buffer_out(size_t len) "len = %zu"
>  dbus_audio_read(size_t len) "len = %zu"
>
> +# pwaudio.c
> +pw_state_changed(int nodeid, const char *s) "node id: %d stream state: %s"
> +pw_read(int32_t avail, uint32_t index, size_t len) "avail=%d index=%u
> len=%zu"
> +pw_write(int32_t filled, int32_t avail, uint32_t index, size_t len)
> "filled=%d avail=%d index=%u len=%zu"
> +pw_vol(const char *ret) "set volume: %s"
> +pw_timer(uint64_t buf_samples) "buffer samples = %lu"
> +pw_audio_init(void) "Initialize Pipewire context"
> +
>  # audio.c
>  audio_timer_start(int interval) "interval %d ms"
>  audio_timer_stop(void) ""
> diff --git a/meson.build b/meson.build
> index 29f8644d6d..31bf280c0d 100644
> --- a/meson.build
> +++ b/meson.build
> @@ -730,6 +730,12 @@ 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'),
> @@ -1667,6 +1673,7 @@ if have_system
>      'jack': jack.found(),
>      'oss': oss.found(),
>      'pa': pulse.found(),
> +    'pipewire': pipewire.found(),
>      'sdl': sdl.found(),
>      'sndio': sndio.found(),
>    }
> @@ -3980,6 +3987,7 @@ if targetos == 'linux'
>    summary_info += {'ALSA support':    alsa}
>    summary_info += {'PulseAudio support': pulse}
>  endif
> +summary_info += {'Pipewire support':   pipewire}
>  summary_info += {'JACK support':      jack}
>  summary_info += {'brlapi support':    brlapi}
>  summary_info += {'vde support':       vde}
> diff --git a/meson_options.txt b/meson_options.txt
> index fc9447d267..9ae1ec7f47 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..e03396a7bc 100644
> --- a/qapi/audio.json
> +++ b/qapi/audio.json
> @@ -324,6 +324,47 @@
>      '*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)
> +#
> +# @latency: latency you want Pipewire to achieve in microseconds
> +#           (default 46000)
> +#
> +# Since: 8.1
> +##
> +{ 'struct': 'AudiodevPipewirePerDirectionOptions',
> +  'base': 'AudiodevPerDirectionOptions',
> +  'data': {
> +    '*name': 'str',
> +    '*stream-name': 'str',
> +    '*latency': 'uint32' } }
> +
> +##
> +# @AudiodevPipewireOptions:
> +#
> +# Options of the Pipewire audio backend.
> +#
> +# @in: options of the capture stream
> +#
> +# @out: options of the playback stream
> +#
> +# Since: 8.1
> +##
> +{ 'struct': 'AudiodevPipewireOptions',
> +  'data': {
> +    '*in':     'AudiodevPipewirePerDirectionOptions',
> +    '*out':    'AudiodevPipewirePerDirectionOptions' } }
> +
>  ##
>  # @AudiodevSdlPerDirectionOptions:
>  #
> @@ -416,6 +457,7 @@
>              { '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 +498,8 @@
>                     '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 59bdf67a2c..2d908717bd 100644
> --- a/qemu-options.hx
> +++ b/qemu-options.hx
> @@ -779,6 +779,12 @@ 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"
> +    "                in|out.stream-name= name of pipewire stream\n"
> +    "                in|out.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 +948,21 @@ 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:
> +
> +    ``in|out.latency=usecs``
> +        Desired latency in microseconds.
> +
> +    ``in|out.name=sink``
> +        Use the specified source/sink for recording/playback.
> +
> +    ``in|out.stream-name``
> +        Specify the name of pipewire stream.
> +
>  ``-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 009fab1515..ba1057b62c 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)'
> @@ -136,6 +137,7 @@ 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
>
>

Hi Dorinda

On Tue, Apr 4, 2023 at 12:36 AM Dorinda Bassey <dbassey@redhat.com> wrote:
>
> Hi Volker, Marc.
>
> I have spent a significant amount of time revising the patchset and I'm eager to see them included in the project. I understand that reviewing the patches can be a time-consuming process and I appreciate the effort you've put into providing feedback and guiding me through the revision process. However I would appreciate any information you can provide on the expected timeline for merging the patches. Let me know if there's anything else I can do to help move this process forward.
>

QEMU is in freeze, until next week or the week after
(https://wiki.qemu.org/Planning/8.0)

It is expected that newly introduced features or code take many
iterations before they are accepted. For the pipewire audio backend,
none of us is an expert afaik, so we are also learning about it, and
trying to make it better than the existing one if possible. Once all
comments are addressed and the patch has a few "Reviewed-by", the
maintainers can queue it. Then we can address further issues as
users/developpers start finding them. In any case, the pipewire
backend will not be "released" before QEMU 8.1 (in about 4 months,
depending on planning)

thanks

Hi Dorinda,

> 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>
> ---
> v10:
> improve error handling
> fix volume functions
> add locks in enable_in out functions
> cleanup in reverse order of intialization
> add triggers for the sync method on the core object
> add waiting function for pw_thread_loop_signal
> improve trace
>
>   audio/audio.c                 |   3 +
>   audio/audio_template.h        |   4 +
>   audio/meson.build             |   1 +
>   audio/pwaudio.c               | 906 ++++++++++++++++++++++++++++++++++
>   audio/trace-events            |   8 +
>   meson.build                   |   8 +
>   meson_options.txt             |   4 +-
>   qapi/audio.json               |  44 ++
>   qemu-options.hx               |  21 +
>   scripts/meson-buildoptions.sh |   8 +-
>   10 files changed, 1004 insertions(+), 3 deletions(-)
>   create mode 100644 audio/pwaudio.c
>
> diff --git a/audio/audio.c b/audio/audio.c
> index 70b096713c..90c7c49d11 100644
> --- a/audio/audio.c
> +++ b/audio/audio.c
> @@ -2061,6 +2061,9 @@ 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 e42326c20d..dc0c74aa74 100644
> --- a/audio/audio_template.h
> +++ b/audio/audio_template.h
> @@ -362,6 +362,10 @@ 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..f9da86059f
> --- /dev/null
> +++ b/audio/pwaudio.c
> @@ -0,0 +1,906 @@
> +/*
> + * QEMU Pipewire audio driver
> + *
> + * Copyright (c) 2023 Red Hat Inc.
> + *
> + * Author: Dorinda Bassey       <dbassey@redhat.com>
> + *
> + * SPDX-License-Identifier: GPL-2.0-or-later
> + */
> +
> +#include "qemu/osdep.h"
> +#include "qemu/module.h"
> +#include "audio.h"
> +#include <errno.h>
> +#include "qemu/error-report.h"
> +#include <spa/param/audio/format-utils.h>
> +#include <spa/utils/ringbuffer.h>
> +#include <spa/utils/result.h>
> +#include <spa/param/props.h>
> +
> +#include <pipewire/pipewire.h>
> +#include "trace.h"
> +
> +#define AUDIO_CAP "pipewire"
> +#define RINGBUFFER_SIZE    (1u << 22)
> +#define RINGBUFFER_MASK    (RINGBUFFER_SIZE - 1)
> +
> +#include "audio_int.h"
> +
> +typedef struct pwvolume {
> +    uint32_t channels;
> +    float values[SPA_AUDIO_MAX_CHANNELS];
> +} pwvolume;
> +
> +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 last_seq, pending_seq, error;
> +} pwaudio;
> +
> +typedef struct PWVoice {
> +    pwaudio *g;
> +    struct pw_stream *stream;
> +    struct spa_hook stream_listener;
> +    struct spa_audio_info_raw info;
> +    uint32_t highwater_mark;
> +    uint32_t frame_size, req;
> +    struct spa_ringbuffer ring;
> +    uint8_t buffer[RINGBUFFER_SIZE];
> +
> +    struct pw_properties *props;
> +    pwvolume volume;
> +    bool muted;
> +} 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 req, index, n_bytes;
> +    int32_t avail;
> +
> +    assert(v->stream);
> +
> +    /* obtain a buffer to read from */
> +    b = pw_stream_dequeue_buffer(v->stream);
> +    if (b == NULL) {
> +        error_report("out of buffers: %s", strerror(errno));
> +        return;
> +    }
> +
> +    buf = b->buffer;
> +    p = buf->datas[0].data;
> +    if (p == NULL) {
> +        return;
> +    }
> +    /* calculate the total no of bytes to read data from buffer */
> +    req = b->requested * v->frame_size;
> +    if (req == 0) {
> +        req = v->req;
> +    }
> +    n_bytes = SPA_MIN(req, buf->datas[0].maxsize);
> +
> +    /* get no of available bytes to read data from buffer */
> +
> +    avail = spa_ringbuffer_get_read_index(&v->ring, &index);

As I wrote in another email, the code should handle buffer underflows. 
The code in v7 is correct if you replace memset(p, 0, n_bytes) with 
audio_pcm_info_clear_buf().

> +
> +    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;
> +
> +    assert(v->stream);
> +
> +    /* obtain a buffer */
> +    b = pw_stream_dequeue_buffer(v->stream);
> +    if (b == NULL) {
> +        error_report("out of buffers: %s", strerror(errno));
> +        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 (filled < 0) {
> +        error_report("%p: underrun write:%u filled:%d", p, index, filled);
> +    } else {
> +        if ((uint32_t) filled + n_bytes > RINGBUFFER_SIZE) {
> +            error_report("%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;
> +
> +    trace_pw_state_changed(pw_stream_get_node_id(v->stream),
> +                           pw_stream_state_as_string(state));
> +
> +    switch (state) {
> +    case PW_STREAM_STATE_ERROR:
> +    case PW_STREAM_STATE_UNCONNECTED:
> +        break;
> +    case PW_STREAM_STATE_PAUSED:
> +    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);
> +
> +    trace_pw_read(avail, index, len);
> +
> +    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_buffer_get_free(HWVoiceOut *hw)
> +{
> +    PWVoiceOut *pw = (PWVoiceOut *)hw;
> +    PWVoice *v = &pw->v;
> +    pwaudio *c = v->g;
> +    const char *error = NULL;
> +    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 */
> +        avail = 0;
> +        goto done_unlock;
> +    }
> +
> +    filled = spa_ringbuffer_get_write_index(&v->ring, &index);
> +    avail = v->highwater_mark - filled;
> +
> +done_unlock:
> +    pw_thread_loop_unlock(c->thread_loop);
> +    return avail;
> +}
> +
> +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;
> +    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 */
> +        len = 0;
> +        goto done_unlock;
> +    }
> +    filled = spa_ringbuffer_get_write_index(&v->ring, &index);
> +    avail = v->highwater_mark - filled;
> +
> +    trace_pw_write(filled, avail, index, len);
> +
> +    if (len > avail) {
> +        len = avail;
> +    }
> +
> +    if (filled < 0) {
> +        error_report("%p: underrun write:%u filled:%d", pw, index, filled);
> +    } else {
> +        if ((uint32_t) filled + len > RINGBUFFER_SIZE) {
> +            error_report("%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);
> +    index += len;
> +    spa_ringbuffer_write_update(&v->ring, index);
> +
> +done_unlock:
> +    pw_thread_loop_unlock(c->thread_loop);
> +    return len;
> +}
> +
> +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)

The parameter name frame_size is not correct. The function 
pw_to_audiofmt() returns the sample 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 *stream_name,
> +              const char *name, enum spa_direction dir)
> +{
> +    int res;
> +    uint32_t n_params;
> +    const struct spa_pod *params[2];
> +    uint8_t buffer[1024];
> +    struct spa_pod_builder b;
> +    uint64_t buf_samples;
> +
> +    v->props = pw_properties_new(NULL, NULL);

Why do you use v->props instead of a local variable props? 
pw_stream_new() takes the ownership of the properties and after the call 
the properties pointer is no longer useful.

With best regards,
Volker

> +
> +    /* 75% of the timer period for faster updates */
> +    buf_samples = (uint64_t)v->g->dev->timer_period * v->info.rate
> +                    * 3 / 4 / 1000000;
> +    trace_pw_timer(buf_samples);
> +    pw_properties_setf(v->props, PW_KEY_NODE_LATENCY, "%" PRIu64 "/%u",
> +                       buf_samples, v->info.rate);
> +
> +    if (name) {
> +        pw_properties_set(v->props, PW_KEY_TARGET_OBJECT, name);
> +    }
> +    v->stream = pw_stream_new(c->core, stream_name, v->props);
> +
> +    if (v->stream == NULL) {
> +        return -1;
> +    }
> +
> +    if (dir == SPA_DIRECTION_INPUT) {
> +        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,
> +                            dir ==
> +                            SPA_DIRECTION_INPUT ? PW_DIRECTION_INPUT :
> +                            PW_DIRECTION_OUTPUT, PW_ID_ANY,
> +                            PW_STREAM_FLAG_AUTOCONNECT |
> +                            PW_STREAM_FLAG_INACTIVE |
> +                            PW_STREAM_FLAG_MAP_BUFFERS |
> +                            PW_STREAM_FLAG_RT_PROCESS, params, n_params);
> +    if (res < 0) {
> +        pw_stream_destroy(v->stream);
> +        return -1;
> +    }
> +
> +    return 0;
> +}
> +
>

Hi Dorinda,

> 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>
> ---
> v10:
> improve error handling
> fix volume functions
> add locks in enable_in out functions
> cleanup in reverse order of intialization
> add triggers for the sync method on the core object
> add waiting function for pw_thread_loop_signal
> improve trace
>
>   audio/audio.c                 |   3 +
>   audio/audio_template.h        |   4 +
>   audio/meson.build             |   1 +
>   audio/pwaudio.c               | 906 ++++++++++++++++++++++++++++++++++
>   audio/trace-events            |   8 +
>   meson.build                   |   8 +
>   meson_options.txt             |   4 +-
>   qapi/audio.json               |  44 ++
>   qemu-options.hx               |  21 +
>   scripts/meson-buildoptions.sh |   8 +-
>   10 files changed, 1004 insertions(+), 3 deletions(-)
>   create mode 100644 audio/pwaudio.c

> diff --git a/audio/trace-events b/audio/trace-events
> index e1ab643add..d6c36139e5 100644
> --- a/audio/trace-events
> +++ b/audio/trace-events
> @@ -18,6 +18,14 @@ dbus_audio_register(const char *s, const char *dir) "sender = %s, dir = %s"
>   dbus_audio_put_buffer_out(size_t len) "len = %zu"
>   dbus_audio_read(size_t len) "len = %zu"
>   
> +# pwaudio.c
> +pw_state_changed(int nodeid, const char *s) "node id: %d stream state: %s"
> +pw_read(int32_t avail, uint32_t index, size_t len) "avail=%d index=%u len=%zu"
> +pw_write(int32_t filled, int32_t avail, uint32_t index, size_t len) "filled=%d avail=%d index=%u len=%zu"
> +pw_vol(const char *ret) "set volume: %s"
> +pw_timer(uint64_t buf_samples) "buffer samples = %lu"

The compilation fails on Windows. It's "%" PRIu64 instead of "%lu". Btw. 
I think it would be better to trace the 'quantum' instead of 'buffer 
samples'.

FAILED: libqemuutil.a.p/meson-generated_.._trace_trace-audio.c.obj
"cc" "-m64" "-mcx16" "-Ilibqemuutil.a.p" "-I." "-I../qemu" "-Iqapi" 
"-Itrace" "-Iui" "-Iui/shader" 
"-IC:/usr/msys64/mingw64/include/glib-2.0" 
"-IC:/usr/msys64/mingw64/lib/glib-2.0/include" 
"-IC:/usr/msys64/mingw64/include/pixman-1" "-fdiagnostics-color=auto" 
"-Wall" "-Winvalid-pch" "-Werror" "-std=gnu11" "-O2" "-g" "-iquote" "." 
"-iquote" "C:/usr/msys64/home/ruemelin/git/qemu" "-iquote" 
"C:/usr/msys64/home/ruemelin/git/qemu/include" "-iquote" 
"C:/usr/msys64/home/ruemelin/git/qemu/tcg/i386" "-U_FORTIFY_SOURCE" 
"-D_FORTIFY_SOURCE=2" "-fno-pie" "-no-pie" "-D_GNU_SOURCE" 
"-D_FILE_OFFSET_BITS=64" "-D_LARGEFILE_SOURCE" "-fno-strict-aliasing" 
"-fno-common" "-fwrapv" "-Wundef" "-Wwrite-strings" 
"-Wmissing-prototypes" "-Wstrict-prototypes" "-Wredundant-decls" 
"-Wold-style-declaration" "-Wold-style-definition" "-Wtype-limits" 
"-Wformat-security" "-Wformat-y2k" "-Winit-self" "-Wignored-qualifiers" 
"-Wempty-body" "-Wnested-externs" "-Wendif-labels" 
"-Wexpansion-to-defined" "-Wimplicit-fallthrough=2" 
"-Wmissing-format-attribute" "-Wno-missing-include-dirs" 
"-Wno-shift-negative-value" "-Wno-psabi" "-fstack-protector-strong" 
"-pthread" -MD -MQ 
libqemuutil.a.p/meson-generated_.._trace_trace-audio.c.obj -MF 
"libqemuutil.a.p/meson-generated_.._trace_trace-audio.c.obj.d" -o 
libqemuutil.a.p/meson-generated_.._trace_trace-audio.c.obj "-c" 
trace/trace-audio.c
In file included from trace/trace-audio.c:5:
C:/usr/msys64/home/ruemelin/git/qemu/audio/trace-events: In function 
'_nocheck__trace_pw_timer':
C:/usr/msys64/home/ruemelin/git/qemu/audio/trace-events:26:22: error: 
format '%lu' expects argument of type 'long unsigned int', but argument 
5 has type 'uint64_t' {aka 'long long unsigned int'} [-Werror=format=]
    26 | pw_timer(uint64_t buf_samples) "buffer samples = %lu"
       |                      ^~~~~~~~~~~~~~~~~~~~~~~~
......
    29 | # audio.c
       |
       |                        |
       |                        uint64_t {aka long long unsigned int}
C:/usr/msys64/home/ruemelin/git/qemu/audio/trace-events:26:22: error: 
format '%lu' expects argument of type 'long unsigned int', but argument 
2 has type 'uint64_t' {aka 'long long unsigned int'} [-Werror=format=]
    26 | pw_timer(uint64_t buf_samples) "buffer samples = %lu"
       |                      ^~~~~~~~~~~
|                                                               |
| uint64_t {aka long long unsigned int}
cc1.exe: all warnings being treated as errors

With best regards,
Volker

> +pw_audio_init(void) "Initialize Pipewire context"
> +
>   # audio.c
>   audio_timer_start(int interval) "interval %d ms"
>   audio_timer_stop(void) ""
>