Re: [PATCH v7] audio/pwaudio.c: Add Pipewire audio backend for QEMU

On Wed, Mar 8, 2023 at 11:39 AM Marc-André Lureau <marcandre.lureau@gmail.com> wrote:

Hi

On Mon, Mar 6, 2023 at 9:11 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>
> ---
> v7:
> use qemu tracing tool
>
> audio/audio.c | 3 +
> audio/audio_template.h | 4 +
> audio/meson.build | 1 +
> audio/pwaudio.c | 814 ++++++++++++++++++++++++++++++++++
> audio/trace-events | 7 +
> meson.build | 8 +
> meson_options.txt | 4 +-
> qapi/audio.json | 45 ++
> qemu-options.hx | 17 +
> scripts/meson-buildoptions.sh | 8 +-
> 10 files changed, 908 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..d357761152
> --- /dev/null
> +++ b/audio/pwaudio.c
> @@ -0,0 +1,814 @@
> +/*
> + * 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 <pipewire/pipewire.h>
> +#include "trace.h"
> +
> +#define AUDIO_CAP "pipewire"
> +#define RINGBUFFER_SIZE (1u << 22)
> +#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;
> +}

That doesn't seem necessary. The doc doesn't say much, but none of the
examples or pw_stream_new_simple() implementation remove the hook on
destroy. I think you can remove that callback.

> +
> +/* 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;
> + }

Can this happen? If you get an event callback for a stream, it should be alive.

> +
> + /* 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;
> + }
> + req = b->requested * v->frame_size;
> + if (req == 0) {
> + req = 4096 * v->frame_size;

What is this for? worth a comment at least.

> + }
> + n_frames = SPA_MIN(req, buf->datas[0].maxsize);

req is in bytes already, maxsize as well.

> + n_bytes = n_frames * v->frame_size;

So here, you multiply by frame_size^2, looking wrong to me.

> +
> + /* get no of available bytes to read data from buffer */
> +
> + avail = spa_ringbuffer_get_read_index(&v->ring, &index);
> +
> + if (!v->enabled) {

See below for state/enabled change.

> + 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);

The ringbuffer maybe full, in which case playback may suffer a large
delay. I think we should maintain a small fixed-delay ringbuffer, or
find a different solution. See below

> +
> + 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) {
> + 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 (!v->enabled) {
> + n_bytes = 0;
> + }
> +
> + 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_state_as_string(state));
> +
> + switch (state) {
> + case PW_STREAM_STATE_ERROR:
> + case PW_STREAM_STATE_UNCONNECTED:
> + {
> + break;
> + }
> + case PW_STREAM_STATE_PAUSED:
> + trace_pw_node(pw_stream_get_node_id(v->stream));
> + break;
> + case PW_STREAM_STATE_CONNECTING:
> + case PW_STREAM_STATE_STREAMING:
> + break;
> + }
> +}

I suggest removing that callback, it doesn't seem useful either.

> +
> +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_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;

(hmm..)

> + 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;

Ok, here I think you tried to mimic jackaudio.c, by using a default of
3 * 512 frames ring/fifo (seems quite arbitrary but if it works...)

However, you don't actually maintain the ringbuffer size, it will keep
growing, just with smaller writes,? I think you could use this value
as the ringbuffer size. The potential additional delay would then be
3*512/samplerate (ex 32ms at 48khz).

> +
> + 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);
> + 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) {
> + 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:
> + pw_stream_destroy(v->stream);
> + return -1;
> +}
> +
> +static int
> +qpw_stream_new(pwaudio *c, PWVoice *v, const char *name)
> +{
> + 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, name);
> + if (r < 0) {
> + goto error;
> + }
> +
> + return r;
> +
> +error:
> + AUD_log(AUDIO_CAP, "Failed to create stream.");
> + 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) {
> + error_report("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) {
> + error_report("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;

Have you considered pw_stream_set_active() ?

> +}
> +
> +static void
> +qpw_enable_in(HWVoiceIn *hw, bool enable)
> +{
> + PWVoiceIn *pi = (PWVoiceIn *) hw;
> + PWVoice *v = &pi->v;
> + v->enabled = enable;

same

> +}
> +
> +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);
> +
> + 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;
> + trace_pw_audio_init("Initialize Pipewire context\n");
> + assert(dev->driver == AUDIODEV_DRIVER_PIPEWIRE);
> + popts = &dev->u.pipewire;
> +
> + if (!popts->has_latency) {
> + popts->has_latency = true;
> + popts->latency = 15000;
> + }

This option is not taken into account. It seems you should set the
stream PW_KEY_NODE_LATENCY, which must be a fraction as a string. You
can either convert the microseconds latency to a fraction
("delay/1000000"), or change the argument to use fractions instead
(although this would be different from other latency properties in
QEMU...)

> +
> + 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_loop;
> + }
> +
> + 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_loop;
> + }
> +
> + if (pw_thread_loop_start(pw->thread_loop) < 0) {
> + error_report("Could not start Pipewire loop");
> + goto fail_start;
> + }
> +
> + pw_thread_loop_lock(pw->thread_loop);
> +
> + pw->core = pw_context_connect(pw->context, NULL, 0);
> + if (pw->core == NULL) {
> + goto fail_conn;
> + }
> +
> + pw_core_add_listener(pw->core, &pw->core_listener, &core_events, pw);
> +
> + pw_thread_loop_unlock(pw->thread_loop);
> +
> + return pw;
> +
> +fail_loop:
> + pw_thread_loop_destroy(pw->thread_loop);
> + g_free(pw);
> + return NULL;
> +fail_start:
> + pw_context_destroy(pw->context);
> + g_free(pw);
> + return NULL;
> +fail_conn:
> + AUD_log(AUDIO_CAP, "Failed to initialize PW context");
> + pw_thread_loop_unlock(pw->thread_loop);
> + pw_thread_loop_stop(pw->thread_loop);
> + pw_core_disconnect(pw->core);
> + 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/audio/trace-events b/audio/trace-events
> index e1ab643add..2ecd8851e6 100644
> --- a/audio/trace-events
> +++ b/audio/trace-events
> @@ -18,6 +18,13 @@ 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(const char *s) "stream state: %s"
> +pw_node(int nodeid) "node id: %d"
> +pw_read(int32_t avail, uint32_t index, size_t len) "avail=%u index=%u len=%zu"
> +pw_write(int32_t filled, int32_t avail, uint32_t index, size_t len) "filled=%u avail=%u index=%u len=%zu"
> +pw_audio_init(const char *msg) "pipewire: %s"
> +
> # audio.c
> audio_timer_start(int interval) "interval %d ms"
> audio_timer_stop(void) ""
> diff --git a/meson.build b/meson.build
> index 6bcab8bf0d..51ec2931e1 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..9a0d7d9ece 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)

not 7.1, but you don't need to say "since .." here, as we assume it
was added with the struct then.

> +#
> +#
> +# Since: 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 15000)
> +#
> +# Since: 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 d42f60fb91..009d58bbf2 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 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
>
>

Volker, Wim, it would be nice if you could review/comment too!

thanks

--
Marc-André Lureau

From:	Dorinda Bassey
Subject:	Re: [PATCH v7] audio/pwaudio.c: Add Pipewire audio backend for QEMU
Date:	Thu, 9 Mar 2023 14:24:26 +0100