audio/audio.c | 3 + audio/audio_template.h | 4 + audio/meson.build | 1 + audio/pwaudio.c | 811 ++++++++++++++++++++++++++++++++++ meson.build | 8 + meson_options.txt | 4 +- qapi/audio.json | 45 ++ qemu-options.hx | 17 + scripts/meson-buildoptions.sh | 8 +- 9 files changed, 898 insertions(+), 3 deletions(-) create mode 100644 audio/pwaudio.c
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>
---
v5:
silence output to the console, use pw debug log
use SPDX identifier
fix typo
change version release
audio/audio.c | 3 +
audio/audio_template.h | 4 +
audio/meson.build | 1 +
audio/pwaudio.c | 811 ++++++++++++++++++++++++++++++++++
meson.build | 8 +
meson_options.txt | 4 +-
qapi/audio.json | 45 ++
qemu-options.hx | 17 +
scripts/meson-buildoptions.sh | 8 +-
9 files changed, 898 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..7448f0abd0
--- /dev/null
+++ b/audio/pwaudio.c
@@ -0,0 +1,811 @@
+/*
+ * 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 <spa/param/audio/format-utils.h>
+#include <spa/utils/ringbuffer.h>
+#include <spa/utils/result.h>
+
+#include <pipewire/pipewire.h>
+
+#define AUDIO_CAP "pipewire"
+#define RINGBUFFER_SIZE (1u << 22)
+#define RINGBUFFER_MASK (RINGBUFFER_SIZE - 1)
+#define BUFFER_SAMPLES 512
+
+#include "audio_int.h"
+
+enum {
+ MODE_SINK,
+ MODE_SOURCE
+};
+
+typedef struct pwaudio {
+ Audiodev *dev;
+ struct pw_thread_loop *thread_loop;
+ struct pw_context *context;
+
+ struct pw_core *core;
+ struct spa_hook core_listener;
+ int seq;
+} pwaudio;
+
+typedef struct PWVoice {
+ pwaudio *g;
+ bool enabled;
+ struct pw_stream *stream;
+ struct spa_hook stream_listener;
+ struct spa_audio_info_raw info;
+ uint32_t frame_size;
+ struct spa_ringbuffer ring;
+ uint8_t buffer[RINGBUFFER_SIZE];
+
+ uint32_t mode;
+ struct pw_properties *props;
+} PWVoice;
+
+typedef struct PWVoiceOut {
+ HWVoiceOut hw;
+ PWVoice v;
+} PWVoiceOut;
+
+typedef struct PWVoiceIn {
+ HWVoiceIn hw;
+ PWVoice v;
+} PWVoiceIn;
+
+static void
+stream_destroy(void *data)
+{
+ PWVoice *v = (PWVoice *) data;
+ spa_hook_remove(&v->stream_listener);
+ v->stream = NULL;
+}
+
+/* output data processing function to read stuffs from the buffer */
+static void
+playback_on_process(void *data)
+{
+ PWVoice *v = (PWVoice *) data;
+ void *p;
+ struct pw_buffer *b;
+ struct spa_buffer *buf;
+ uint32_t n_frames, req, index, n_bytes;
+ int32_t avail;
+
+ if (!v->stream) {
+ return;
+ }
+
+ /* obtain a buffer to read from */
+ b = pw_stream_dequeue_buffer(v->stream);
+ if (b == NULL) {
+ pw_log_warn("out of buffers: %m");
+ return;
+ }
+
+ buf = b->buffer;
+ p = buf->datas[0].data;
+ if (p == NULL) {
+ return;
+ }
+ req = b->requested * v->frame_size;
+ if (req == 0) {
+ req = 4096 * v->frame_size;
+ }
+ n_frames = SPA_MIN(req, buf->datas[0].maxsize);
+ n_bytes = n_frames * v->frame_size;
+
+ /* get no of available bytes to read data from buffer */
+
+ avail = spa_ringbuffer_get_read_index(&v->ring, &index);
+
+ if (!v->enabled) {
+ avail = 0;
+ }
+
+ if (avail == 0) {
+ memset(p, 0, n_bytes);
+ } else {
+ if (avail < (int32_t) n_bytes) {
+ n_bytes = avail;
+ }
+
+ spa_ringbuffer_read_data(&v->ring,
+ v->buffer, RINGBUFFER_SIZE,
+ index & RINGBUFFER_MASK, p, n_bytes);
+
+ index += n_bytes;
+ spa_ringbuffer_read_update(&v->ring, index);
+ }
+
+ buf->datas[0].chunk->offset = 0;
+ buf->datas[0].chunk->stride = v->frame_size;
+ buf->datas[0].chunk->size = n_bytes;
+
+ /* queue the buffer for playback */
+ pw_stream_queue_buffer(v->stream, b);
+}
+
+/* output data processing function to generate stuffs in the buffer */
+static void
+capture_on_process(void *data)
+{
+ PWVoice *v = (PWVoice *) data;
+ void *p;
+ struct pw_buffer *b;
+ struct spa_buffer *buf;
+ int32_t filled;
+ uint32_t index, offs, n_bytes;
+
+ if (!v->stream) {
+ return;
+ }
+
+ /* obtain a buffer */
+ b = pw_stream_dequeue_buffer(v->stream);
+ if (b == NULL) {
+ pw_log_warn("out of buffers: %m");
+ return;
+ }
+
+ /* Write data into buffer */
+ buf = b->buffer;
+ p = buf->datas[0].data;
+ if (p == NULL) {
+ return;
+ }
+ offs = SPA_MIN(buf->datas[0].chunk->offset, buf->datas[0].maxsize);
+ n_bytes = SPA_MIN(buf->datas[0].chunk->size, buf->datas[0].maxsize - offs);
+
+ filled = spa_ringbuffer_get_write_index(&v->ring, &index);
+
+ if (!v->enabled) {
+ n_bytes = 0;
+ }
+
+ if (filled < 0) {
+ pw_log_warn("%p: underrun write:%u filled:%d", p, index, filled);
+ } else {
+ if ((uint32_t) filled + n_bytes > RINGBUFFER_SIZE) {
+ pw_log_warn("%p: overrun write:%u filled:%d + size:%u > max:%u",
+ p, index, filled, n_bytes, RINGBUFFER_SIZE);
+ }
+ }
+ spa_ringbuffer_write_data(&v->ring,
+ v->buffer, RINGBUFFER_SIZE,
+ index & RINGBUFFER_MASK,
+ SPA_PTROFF(p, offs, void), n_bytes);
+ index += n_bytes;
+ spa_ringbuffer_write_update(&v->ring, index);
+
+ /* queue the buffer for playback */
+ pw_stream_queue_buffer(v->stream, b);
+}
+
+static void
+on_stream_state_changed(void *_data, enum pw_stream_state old,
+ enum pw_stream_state state, const char *error)
+{
+ PWVoice *v = (PWVoice *) _data;
+
+ pw_log_debug("stream state: \"%s\"\n", pw_stream_state_as_string(state));
+
+ switch (state) {
+ case PW_STREAM_STATE_ERROR:
+ case PW_STREAM_STATE_UNCONNECTED:
+ {
+ break;
+ }
+ case PW_STREAM_STATE_PAUSED:
+ pw_log_debug("node id: %d\n", pw_stream_get_node_id(v->stream));
+ break;
+ case PW_STREAM_STATE_CONNECTING:
+ case PW_STREAM_STATE_STREAMING:
+ break;
+ }
+}
+
+static const struct pw_stream_events capture_stream_events = {
+ PW_VERSION_STREAM_EVENTS,
+ .destroy = stream_destroy,
+ .state_changed = on_stream_state_changed,
+ .process = capture_on_process
+};
+
+static const struct pw_stream_events playback_stream_events = {
+ PW_VERSION_STREAM_EVENTS,
+ .destroy = stream_destroy,
+ .state_changed = on_stream_state_changed,
+ .process = playback_on_process
+};
+
+static size_t
+qpw_read(HWVoiceIn *hw, void *data, size_t len)
+{
+ PWVoiceIn *pw = (PWVoiceIn *) hw;
+ PWVoice *v = &pw->v;
+ pwaudio *c = v->g;
+ const char *error = NULL;
+ size_t l;
+ int32_t avail;
+ uint32_t index;
+
+ pw_thread_loop_lock(c->thread_loop);
+ if (pw_stream_get_state(v->stream, &error) != PW_STREAM_STATE_STREAMING) {
+ /* wait for stream to become ready */
+ l = 0;
+ goto done_unlock;
+ }
+ /* get no of available bytes to read data from buffer */
+ avail = spa_ringbuffer_get_read_index(&v->ring, &index);
+
+ if (avail < (int32_t) len) {
+ len = avail;
+ }
+
+ spa_ringbuffer_read_data(&v->ring,
+ v->buffer, RINGBUFFER_SIZE,
+ index & RINGBUFFER_MASK, data, len);
+ index += len;
+ spa_ringbuffer_read_update(&v->ring, index);
+ l = len;
+
+done_unlock:
+ pw_thread_loop_unlock(c->thread_loop);
+ return l;
+}
+
+static size_t
+qpw_write(HWVoiceOut *hw, void *data, size_t len)
+{
+ PWVoiceOut *pw = (PWVoiceOut *) hw;
+ PWVoice *v = &pw->v;
+ pwaudio *c = v->g;
+ const char *error = NULL;
+ const int periods = 3;
+ size_t l;
+ int32_t filled, avail;
+ uint32_t index;
+
+ pw_thread_loop_lock(c->thread_loop);
+ if (pw_stream_get_state(v->stream, &error) != PW_STREAM_STATE_STREAMING) {
+ /* wait for stream to become ready */
+ l = 0;
+ goto done_unlock;
+ }
+ filled = spa_ringbuffer_get_write_index(&v->ring, &index);
+
+ avail = BUFFER_SAMPLES * v->frame_size * periods - filled;
+
+ pw_log_debug("%u %u %u %zu", filled, avail, index, len);
+
+ if (len > avail) {
+ len = avail;
+ }
+
+ if (filled < 0) {
+ pw_log_warn("%p: underrun write:%u filled:%d", pw, index, filled);
+ } else {
+ if ((uint32_t) filled + len > RINGBUFFER_SIZE) {
+ pw_log_warn("%p: overrun write:%u filled:%d + size:%zu > max:%u",
+ pw, index, filled, len, RINGBUFFER_SIZE);
+ }
+ }
+
+ spa_ringbuffer_write_data(&v->ring,
+ v->buffer, RINGBUFFER_SIZE,
+ index & RINGBUFFER_MASK, data, len);
+ index += len;
+ spa_ringbuffer_write_update(&v->ring, index);
+ l = len;
+
+done_unlock:
+ pw_thread_loop_unlock(c->thread_loop);
+ return l;
+}
+
+static int
+audfmt_to_pw(AudioFormat fmt, int endianness)
+{
+ int format;
+
+ switch (fmt) {
+ case AUDIO_FORMAT_S8:
+ format = SPA_AUDIO_FORMAT_S8;
+ break;
+ case AUDIO_FORMAT_U8:
+ format = SPA_AUDIO_FORMAT_U8;
+ break;
+ case AUDIO_FORMAT_S16:
+ format = endianness ? SPA_AUDIO_FORMAT_S16_BE : SPA_AUDIO_FORMAT_S16_LE;
+ break;
+ case AUDIO_FORMAT_U16:
+ format = endianness ? SPA_AUDIO_FORMAT_U16_BE : SPA_AUDIO_FORMAT_U16_LE;
+ break;
+ case AUDIO_FORMAT_S32:
+ format = endianness ? SPA_AUDIO_FORMAT_S32_BE : SPA_AUDIO_FORMAT_S32_LE;
+ break;
+ case AUDIO_FORMAT_U32:
+ format = endianness ? SPA_AUDIO_FORMAT_U32_BE : SPA_AUDIO_FORMAT_U32_LE;
+ break;
+ case AUDIO_FORMAT_F32:
+ format = endianness ? SPA_AUDIO_FORMAT_F32_BE : SPA_AUDIO_FORMAT_F32_LE;
+ break;
+ default:
+ dolog("Internal logic error: Bad audio format %d\n", fmt);
+ format = SPA_AUDIO_FORMAT_U8;
+ break;
+ }
+ return format;
+}
+
+static AudioFormat
+pw_to_audfmt(enum spa_audio_format fmt, int *endianness,
+ uint32_t *frame_size)
+{
+ switch (fmt) {
+ case SPA_AUDIO_FORMAT_S8:
+ *frame_size = 1;
+ return AUDIO_FORMAT_S8;
+ case SPA_AUDIO_FORMAT_U8:
+ *frame_size = 1;
+ return AUDIO_FORMAT_U8;
+ case SPA_AUDIO_FORMAT_S16_BE:
+ *frame_size = 2;
+ *endianness = 1;
+ return AUDIO_FORMAT_S16;
+ case SPA_AUDIO_FORMAT_S16_LE:
+ *frame_size = 2;
+ *endianness = 0;
+ return AUDIO_FORMAT_S16;
+ case SPA_AUDIO_FORMAT_U16_BE:
+ *frame_size = 2;
+ *endianness = 1;
+ return AUDIO_FORMAT_U16;
+ case SPA_AUDIO_FORMAT_U16_LE:
+ *frame_size = 2;
+ *endianness = 0;
+ return AUDIO_FORMAT_U16;
+ case SPA_AUDIO_FORMAT_S32_BE:
+ *frame_size = 4;
+ *endianness = 1;
+ return AUDIO_FORMAT_S32;
+ case SPA_AUDIO_FORMAT_S32_LE:
+ *frame_size = 4;
+ *endianness = 0;
+ return AUDIO_FORMAT_S32;
+ case SPA_AUDIO_FORMAT_U32_BE:
+ *frame_size = 4;
+ *endianness = 1;
+ return AUDIO_FORMAT_U32;
+ case SPA_AUDIO_FORMAT_U32_LE:
+ *frame_size = 4;
+ *endianness = 0;
+ return AUDIO_FORMAT_U32;
+ case SPA_AUDIO_FORMAT_F32_BE:
+ *frame_size = 4;
+ *endianness = 1;
+ return AUDIO_FORMAT_F32;
+ case SPA_AUDIO_FORMAT_F32_LE:
+ *frame_size = 4;
+ *endianness = 0;
+ return AUDIO_FORMAT_F32;
+ default:
+ *frame_size = 1;
+ dolog("Internal logic error: Bad spa_audio_format %d\n", fmt);
+ return AUDIO_FORMAT_U8;
+ }
+}
+
+static int
+create_stream(pwaudio *c, PWVoice *v, const char *name)
+{
+ int res;
+ uint32_t n_params;
+ const struct spa_pod *params[2];
+ uint8_t buffer[1024];
+ struct spa_pod_builder b;
+
+ v->stream = pw_stream_new(c->core, name, NULL);
+
+ if (v->stream == NULL) {
+ res = -errno;
+ goto error;
+ }
+
+ if (v->mode == MODE_SOURCE) {
+ pw_stream_add_listener(v->stream,
+ &v->stream_listener, &capture_stream_events, v);
+ } else {
+ pw_stream_add_listener(v->stream,
+ &v->stream_listener, &playback_stream_events, v);
+ }
+
+ n_params = 0;
+ spa_pod_builder_init(&b, buffer, sizeof(buffer));
+ params[n_params++] = spa_format_audio_raw_build(&b,
+ SPA_PARAM_EnumFormat,
+ &v->info);
+
+ /* connect the stream to a sink or source */
+ res = pw_stream_connect(v->stream,
+ v->mode ==
+ MODE_SOURCE ? PW_DIRECTION_INPUT :
+ PW_DIRECTION_OUTPUT, PW_ID_ANY,
+ PW_STREAM_FLAG_AUTOCONNECT |
+ PW_STREAM_FLAG_MAP_BUFFERS |
+ PW_STREAM_FLAG_RT_PROCESS, params, n_params);
+ if (res < 0) {
+ goto error;
+ }
+
+ return 0;
+error:
+ return res;
+}
+
+static void
+pw_destroy(pwaudio *c)
+{
+ if (c->thread_loop) {
+ pw_thread_loop_stop(c->thread_loop);
+ }
+ if (c->core) {
+ pw_core_disconnect(c->core);
+ }
+
+ g_free(c);
+}
+
+static int
+qpw_stream_new(pwaudio *c, PWVoice *v, const char *name)
+{
+ int r;
+
+ pw_thread_loop_lock(c->thread_loop);
+
+ switch (v->info.channels) {
+ case 8:
+ v->info.position[0] = SPA_AUDIO_CHANNEL_FL;
+ v->info.position[1] = SPA_AUDIO_CHANNEL_FR;
+ v->info.position[2] = SPA_AUDIO_CHANNEL_FC;
+ v->info.position[3] = SPA_AUDIO_CHANNEL_LFE;
+ v->info.position[4] = SPA_AUDIO_CHANNEL_RL;
+ v->info.position[5] = SPA_AUDIO_CHANNEL_RR;
+ v->info.position[6] = SPA_AUDIO_CHANNEL_SL;
+ v->info.position[7] = SPA_AUDIO_CHANNEL_SR;
+ break;
+ case 6:
+ v->info.position[0] = SPA_AUDIO_CHANNEL_FL;
+ v->info.position[1] = SPA_AUDIO_CHANNEL_FR;
+ v->info.position[2] = SPA_AUDIO_CHANNEL_FC;
+ v->info.position[3] = SPA_AUDIO_CHANNEL_LFE;
+ v->info.position[4] = SPA_AUDIO_CHANNEL_RL;
+ v->info.position[5] = SPA_AUDIO_CHANNEL_RR;
+ break;
+ case 5:
+ v->info.position[0] = SPA_AUDIO_CHANNEL_FL;
+ v->info.position[1] = SPA_AUDIO_CHANNEL_FR;
+ v->info.position[2] = SPA_AUDIO_CHANNEL_FC;
+ v->info.position[3] = SPA_AUDIO_CHANNEL_LFE;
+ v->info.position[4] = SPA_AUDIO_CHANNEL_RC;
+ break;
+ case 4:
+ v->info.position[0] = SPA_AUDIO_CHANNEL_FL;
+ v->info.position[1] = SPA_AUDIO_CHANNEL_FR;
+ v->info.position[2] = SPA_AUDIO_CHANNEL_FC;
+ v->info.position[3] = SPA_AUDIO_CHANNEL_RC;
+ break;
+ case 3:
+ v->info.position[0] = SPA_AUDIO_CHANNEL_FL;
+ v->info.position[1] = SPA_AUDIO_CHANNEL_FR;
+ v->info.position[2] = SPA_AUDIO_CHANNEL_LFE;
+ break;
+ case 2:
+ v->info.position[0] = SPA_AUDIO_CHANNEL_FL;
+ v->info.position[1] = SPA_AUDIO_CHANNEL_FR;
+ break;
+ case 1:
+ v->info.position[0] = SPA_AUDIO_CHANNEL_MONO;
+ break;
+ default:
+ for (size_t i = 0; i < v->info.channels; i++) {
+ v->info.position[i] = SPA_AUDIO_CHANNEL_UNKNOWN;
+ }
+ break;
+ }
+
+ /* create a new unconnected pwstream */
+ r = create_stream(c, v, name);
+ if (r < 0) {
+ goto error;
+ }
+
+ pw_thread_loop_unlock(c->thread_loop);
+ return r;
+
+error:
+ AUD_log(AUDIO_CAP, "Failed to create stream.");
+ pw_thread_loop_unlock(c->thread_loop);
+ pw_destroy(c);
+ return -1;
+}
+
+static int
+qpw_init_out(HWVoiceOut *hw, struct audsettings *as, void *drv_opaque)
+{
+ PWVoiceOut *pw = (PWVoiceOut *) hw;
+ PWVoice *v = &pw->v;
+ struct audsettings obt_as = *as;
+ pwaudio *c = v->g = drv_opaque;
+ AudiodevPipewireOptions *popts = &c->dev->u.pipewire;
+ AudiodevPipewirePerDirectionOptions *ppdo = popts->out;
+ int r;
+ v->enabled = false;
+
+ v->mode = MODE_SINK;
+
+ pw_thread_loop_lock(c->thread_loop);
+
+ v->info.format = audfmt_to_pw(as->fmt, as->endianness);
+ v->info.channels = as->nchannels;
+ v->info.rate = as->freq;
+
+ obt_as.fmt =
+ pw_to_audfmt(v->info.format, &obt_as.endianness, &v->frame_size);
+ v->frame_size *= as->nchannels;
+
+ /* call the function that creates a new stream for playback */
+ r = qpw_stream_new(c, v, ppdo->stream_name ? : c->dev->id);
+ if (r < 0) {
+ pw_log_error("qpw_stream_new for playback failed\n ");
+ goto fail;
+ }
+
+ /* report the audio format we support */
+ audio_pcm_init_info(&hw->info, &obt_as);
+
+ /* report the buffer size to qemu */
+ hw->samples = BUFFER_SAMPLES;
+
+ pw_thread_loop_unlock(c->thread_loop);
+ return 0;
+fail:
+ pw_thread_loop_unlock(c->thread_loop);
+ return -1;
+}
+
+static int
+qpw_init_in(HWVoiceIn *hw, struct audsettings *as, void *drv_opaque)
+{
+ PWVoiceIn *pw = (PWVoiceIn *) hw;
+ PWVoice *v = &pw->v;
+ struct audsettings obt_as = *as;
+ pwaudio *c = v->g = drv_opaque;
+ AudiodevPipewireOptions *popts = &c->dev->u.pipewire;
+ AudiodevPipewirePerDirectionOptions *ppdo = popts->in;
+ int r;
+ v->enabled = false;
+
+ v->mode = MODE_SOURCE;
+ pw_thread_loop_lock(c->thread_loop);
+
+ v->info.format = audfmt_to_pw(as->fmt, as->endianness);
+ v->info.channels = as->nchannels;
+ v->info.rate = as->freq;
+
+ obt_as.fmt =
+ pw_to_audfmt(v->info.format, &obt_as.endianness, &v->frame_size);
+ v->frame_size *= as->nchannels;
+
+ /* call the function that creates a new stream for recording */
+ r = qpw_stream_new(c, v, ppdo->stream_name ? : c->dev->id);
+ if (r < 0) {
+ pw_log_error("qpw_stream_new for recording failed\n ");
+ goto fail;
+ }
+
+ /* report the audio format we support */
+ audio_pcm_init_info(&hw->info, &obt_as);
+
+ /* report the buffer size to qemu */
+ hw->samples = BUFFER_SAMPLES;
+
+ pw_thread_loop_unlock(c->thread_loop);
+ return 0;
+fail:
+ pw_thread_loop_unlock(c->thread_loop);
+ return -1;
+}
+
+static void
+qpw_fini_out(HWVoiceOut *hw)
+{
+ PWVoiceOut *pw = (PWVoiceOut *) hw;
+ PWVoice *v = &pw->v;
+
+ if (v->stream) {
+ pwaudio *c = v->g;
+ pw_thread_loop_lock(c->thread_loop);
+ pw_stream_destroy(v->stream);
+ v->stream = NULL;
+ pw_thread_loop_unlock(c->thread_loop);
+ }
+}
+
+static void
+qpw_fini_in(HWVoiceIn *hw)
+{
+ PWVoiceIn *pw = (PWVoiceIn *) hw;
+ PWVoice *v = &pw->v;
+
+ if (v->stream) {
+ pwaudio *c = v->g;
+ pw_thread_loop_lock(c->thread_loop);
+ pw_stream_destroy(v->stream);
+ v->stream = NULL;
+ pw_thread_loop_unlock(c->thread_loop);
+ }
+}
+
+static void
+qpw_enable_out(HWVoiceOut *hw, bool enable)
+{
+ PWVoiceOut *po = (PWVoiceOut *) hw;
+ PWVoice *v = &po->v;
+ v->enabled = enable;
+}
+
+static void
+qpw_enable_in(HWVoiceIn *hw, bool enable)
+{
+ PWVoiceIn *pi = (PWVoiceIn *) hw;
+ PWVoice *v = &pi->v;
+ v->enabled = enable;
+}
+
+static void
+on_core_error(void *data, uint32_t id, int seq, int res, const char *message)
+{
+ pwaudio *pw = data;
+
+ pw_log_warn("error id:%u seq:%d res:%d (%s): %s",
+ id, seq, res, spa_strerror(res), message);
+
+ pw_thread_loop_signal(pw->thread_loop, FALSE);
+}
+
+static void
+on_core_done(void *data, uint32_t id, int seq)
+{
+ pwaudio *pw = data;
+ if (id == PW_ID_CORE) {
+ pw->seq = seq;
+ pw_thread_loop_signal(pw->thread_loop, FALSE);
+ }
+}
+
+static const struct pw_core_events core_events = {
+ PW_VERSION_CORE_EVENTS,
+ .done = on_core_done,
+ .error = on_core_error,
+};
+
+static void *
+qpw_audio_init(Audiodev *dev)
+{
+ pwaudio *pw;
+ pw = g_new0(pwaudio, 1);
+ pw_init(NULL, NULL);
+
+ AudiodevPipewireOptions *popts;
+ pw_log_debug("Initialize PW context\n");
+ assert(dev->driver == AUDIODEV_DRIVER_PIPEWIRE);
+ popts = &dev->u.pipewire;
+
+ if (!popts->has_latency) {
+ popts->has_latency = true;
+ popts->latency = 15000;
+ }
+
+ pw->dev = dev;
+ pw->thread_loop = pw_thread_loop_new("Pipewire thread loop", NULL);
+ if (pw->thread_loop == NULL) {
+ goto fail;
+ }
+ pw->context =
+ pw_context_new(pw_thread_loop_get_loop(pw->thread_loop), NULL, 0);
+
+ if (pw_thread_loop_start(pw->thread_loop) < 0) {
+ goto fail;
+ }
+
+ pw_thread_loop_lock(pw->thread_loop);
+
+ pw->core = pw_context_connect(pw->context, NULL, 0);
+ if (pw->core == NULL) {
+ goto fail;
+ }
+
+ pw_core_add_listener(pw->core, &pw->core_listener, &core_events, pw);
+
+ pw_thread_loop_unlock(pw->thread_loop);
+
+ return pw;
+
+fail:
+ AUD_log(AUDIO_CAP, "Failed to initialize PW context");
+ pw_thread_loop_unlock(pw->thread_loop);
+ pw_context_destroy(pw->context);
+ pw_thread_loop_destroy(pw->thread_loop);
+ g_free(pw);
+ return NULL;
+}
+
+static void
+qpw_audio_fini(void *opaque)
+{
+ pwaudio *pw = opaque;
+
+ pw_thread_loop_stop(pw->thread_loop);
+
+ if (pw->core) {
+ spa_hook_remove(&pw->core_listener);
+ spa_zero(pw->core_listener);
+ pw_core_disconnect(pw->core);
+ }
+
+ if (pw->context) {
+ pw_context_destroy(pw->context);
+ }
+ pw_thread_loop_destroy(pw->thread_loop);
+
+ g_free(pw);
+}
+
+static struct audio_pcm_ops qpw_pcm_ops = {
+ .init_out = qpw_init_out,
+ .fini_out = qpw_fini_out,
+ .write = qpw_write,
+ .buffer_get_free = audio_generic_buffer_get_free,
+ .run_buffer_out = audio_generic_run_buffer_out,
+ .enable_out = qpw_enable_out,
+
+ .init_in = qpw_init_in,
+ .fini_in = qpw_fini_in,
+ .read = qpw_read,
+ .run_buffer_in = audio_generic_run_buffer_in,
+ .enable_in = qpw_enable_in
+};
+
+static struct audio_driver pw_audio_driver = {
+ .name = "pipewire",
+ .descr = "http://www.pipewire.org/",
+ .init = qpw_audio_init,
+ .fini = qpw_audio_fini,
+ .pcm_ops = &qpw_pcm_ops,
+ .can_be_default = 1,
+ .max_voices_out = INT_MAX,
+ .max_voices_in = INT_MAX,
+ .voice_size_out = sizeof(PWVoiceOut),
+ .voice_size_in = sizeof(PWVoiceIn),
+};
+
+static void
+register_audio_pw(void)
+{
+ audio_driver_register(&pw_audio_driver);
+}
+
+type_init(register_audio_pw);
diff --git a/meson.build b/meson.build
index 6cb2b1a42f..2aa0b397ea 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(),
}
@@ -3978,6 +3985,7 @@ 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 6b0900205e..6c5f260a6a 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)
+#
+#
+# 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 beeb4475ba..2fd88ccf15 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 5d969a94c0..236a714087 100644
--- a/scripts/meson-buildoptions.sh
+++ b/scripts/meson-buildoptions.sh
@@ -1,7 +1,8 @@
# This file is generated by meson-buildoptions.py, do not edit!
meson_options_help() {
- printf "%s\n" ' --audio-drv-list=CHOICES Set audio driver list [default] (choices: alsa/co'
- printf "%s\n" ' reaudio/default/dsound/jack/oss/pa/sdl/sndio)'
+ printf "%s\n" ' --audio-drv-list=CHOICES Set audio driver list [default] (choices: al'
+ printf "%s\n" ' sa/coreaudio/default/dsound/jack/oss/pa/'
+ printf "%s\n" ' pipewire/sdl/sndio)'
printf "%s\n" ' --block-drv-ro-whitelist=VALUE'
printf "%s\n" ' set block driver read-only whitelist (by default'
printf "%s\n" ' affects only QEMU, not tools like qemu-img)'
@@ -135,6 +136,7 @@ meson_options_help() {
printf "%s\n" ' oss OSS sound support'
printf "%s\n" ' pa PulseAudio sound support'
printf "%s\n" ' parallels parallels image format support'
+ printf "%s\n" ' pipewire Pipewire sound support'
printf "%s\n" ' png PNG support with libpng'
printf "%s\n" ' pvrdma Enable PVRDMA support'
printf "%s\n" ' qcow1 qcow1 image format support'
@@ -369,6 +371,8 @@ _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 ;;
--
2.39.1Hi Dorinda
On Tue, Feb 28, 2023 at 2:01 AM 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>
>
I didn't look at buffering/latency details yet, but here are some
other comments for this iteration.
thanks
---
> v5:
> silence output to the console, use pw debug log
> use SPDX identifier
> fix typo
> change version release
>
> audio/audio.c | 3 +
> audio/audio_template.h | 4 +
> audio/meson.build | 1 +
> audio/pwaudio.c | 811 ++++++++++++++++++++++++++++++++++
> meson.build | 8 +
> meson_options.txt | 4 +-
> qapi/audio.json | 45 ++
> qemu-options.hx | 17 +
> scripts/meson-buildoptions.sh | 8 +-
> 9 files changed, 898 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..7448f0abd0
> --- /dev/null
> +++ b/audio/pwaudio.c
> @@ -0,0 +1,811 @@
> +/*
> + * 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 <spa/param/audio/format-utils.h>
> +#include <spa/utils/ringbuffer.h>
> +#include <spa/utils/result.h>
> +
> +#include <pipewire/pipewire.h>
> +
> +#define AUDIO_CAP "pipewire"
> +#define RINGBUFFER_SIZE (1u << 22)
> +#define RINGBUFFER_MASK (RINGBUFFER_SIZE - 1)
> +#define BUFFER_SAMPLES 512
> +
> +#include "audio_int.h"
> +
> +enum {
> + MODE_SINK,
> + MODE_SOURCE
> +};
> +
> +typedef struct pwaudio {
> + Audiodev *dev;
> + struct pw_thread_loop *thread_loop;
> + struct pw_context *context;
> +
> + struct pw_core *core;
> + struct spa_hook core_listener;
> + int seq;
> +} pwaudio;
> +
> +typedef struct PWVoice {
> + pwaudio *g;
> + bool enabled;
> + struct pw_stream *stream;
> + struct spa_hook stream_listener;
> + struct spa_audio_info_raw info;
> + uint32_t frame_size;
> + struct spa_ringbuffer ring;
> + uint8_t buffer[RINGBUFFER_SIZE];
> +
> + uint32_t mode;
> + struct pw_properties *props;
> +} PWVoice;
> +
> +typedef struct PWVoiceOut {
> + HWVoiceOut hw;
> + PWVoice v;
> +} PWVoiceOut;
> +
> +typedef struct PWVoiceIn {
> + HWVoiceIn hw;
> + PWVoice v;
> +} PWVoiceIn;
> +
> +static void
> +stream_destroy(void *data)
> +{
> + PWVoice *v = (PWVoice *) data;
> + spa_hook_remove(&v->stream_listener);
> + v->stream = NULL;
> +}
> +
> +/* output data processing function to read stuffs from the buffer */
> +static void
> +playback_on_process(void *data)
> +{
> + PWVoice *v = (PWVoice *) data;
> + void *p;
> + struct pw_buffer *b;
> + struct spa_buffer *buf;
> + uint32_t n_frames, req, index, n_bytes;
> + int32_t avail;
> +
> + if (!v->stream) {
> + return;
> + }
> +
> + /* obtain a buffer to read from */
> + b = pw_stream_dequeue_buffer(v->stream);
> + if (b == NULL) {
> + pw_log_warn("out of buffers: %m");
We don't use %m in QEMU, but error_report("..%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;
> + }
> + n_frames = SPA_MIN(req, buf->datas[0].maxsize);
> + n_bytes = n_frames * v->frame_size;
> +
> + /* get no of available bytes to read data from buffer */
> +
> + avail = spa_ringbuffer_get_read_index(&v->ring, &index);
> +
> + if (!v->enabled) {
> + avail = 0;
> + }
> +
> + if (avail == 0) {
> + memset(p, 0, n_bytes);
> + } else {
> + if (avail < (int32_t) n_bytes) {
> + n_bytes = avail;
> + }
> +
> + spa_ringbuffer_read_data(&v->ring,
> + v->buffer, RINGBUFFER_SIZE,
> + index & RINGBUFFER_MASK, p, n_bytes);
> +
> + index += n_bytes;
> + spa_ringbuffer_read_update(&v->ring, index);
> + }
> +
> + buf->datas[0].chunk->offset = 0;
> + buf->datas[0].chunk->stride = v->frame_size;
> + buf->datas[0].chunk->size = n_bytes;
> +
> + /* queue the buffer for playback */
> + pw_stream_queue_buffer(v->stream, b);
> +}
> +
> +/* output data processing function to generate stuffs in the buffer */
> +static void
> +capture_on_process(void *data)
> +{
> + PWVoice *v = (PWVoice *) data;
> + void *p;
> + struct pw_buffer *b;
> + struct spa_buffer *buf;
> + int32_t filled;
> + uint32_t index, offs, n_bytes;
> +
> + if (!v->stream) {
> + return;
> + }
> +
> + /* obtain a buffer */
> + b = pw_stream_dequeue_buffer(v->stream);
> + if (b == NULL) {
> + pw_log_warn("out of buffers: %m");
> + return;
> + }
> +
> + /* Write data into buffer */
> + buf = b->buffer;
> + p = buf->datas[0].data;
> + if (p == NULL) {
> + return;
> + }
> + offs = SPA_MIN(buf->datas[0].chunk->offset, buf->datas[0].maxsize);
> + n_bytes = SPA_MIN(buf->datas[0].chunk->size, buf->datas[0].maxsize - offs);
> +
> + filled = spa_ringbuffer_get_write_index(&v->ring, &index);
> +
> + if (!v->enabled) {
> + n_bytes = 0;
> + }
> +
> + if (filled < 0) {
> + pw_log_warn("%p: underrun write:%u filled:%d", p, index, filled);
> + } else {
> + if ((uint32_t) filled + n_bytes > RINGBUFFER_SIZE) {
> + pw_log_warn("%p: overrun write:%u filled:%d + size:%u > max:%u",
> + p, index, filled, n_bytes, RINGBUFFER_SIZE);
> + }
> + }
> + spa_ringbuffer_write_data(&v->ring,
> + v->buffer, RINGBUFFER_SIZE,
> + index & RINGBUFFER_MASK,
> + SPA_PTROFF(p, offs, void), n_bytes);
> + index += n_bytes;
> + spa_ringbuffer_write_update(&v->ring, index);
> +
> + /* queue the buffer for playback */
> + pw_stream_queue_buffer(v->stream, b);
> +}
> +
> +static void
> +on_stream_state_changed(void *_data, enum pw_stream_state old,
> + enum pw_stream_state state, const char *error)
> +{
> + PWVoice *v = (PWVoice *) _data;
> +
> + pw_log_debug("stream state: \"%s\"\n", pw_stream_state_as_string(state));
> +
> + switch (state) {
> + case PW_STREAM_STATE_ERROR:
> + case PW_STREAM_STATE_UNCONNECTED:
> + {
> + break;
> + }
> + case PW_STREAM_STATE_PAUSED:
> + pw_log_debug("node id: %d\n", pw_stream_get_node_id(v->stream));
> + break;
> + case PW_STREAM_STATE_CONNECTING:
> + case PW_STREAM_STATE_STREAMING:
> + break;
> + }
> +}
> +
> +static const struct pw_stream_events capture_stream_events = {
> + PW_VERSION_STREAM_EVENTS,
> + .destroy = stream_destroy,
> + .state_changed = on_stream_state_changed,
> + .process = capture_on_process
> +};
> +
> +static const struct pw_stream_events playback_stream_events = {
> + PW_VERSION_STREAM_EVENTS,
> + .destroy = stream_destroy,
> + .state_changed = on_stream_state_changed,
> + .process = playback_on_process
> +};
> +
> +static size_t
> +qpw_read(HWVoiceIn *hw, void *data, size_t len)
> +{
> + PWVoiceIn *pw = (PWVoiceIn *) hw;
> + PWVoice *v = &pw->v;
> + pwaudio *c = v->g;
> + const char *error = NULL;
> + size_t l;
> + int32_t avail;
> + uint32_t index;
> +
> + pw_thread_loop_lock(c->thread_loop);
> + if (pw_stream_get_state(v->stream, &error) != PW_STREAM_STATE_STREAMING) {
> + /* wait for stream to become ready */
> + l = 0;
> + goto done_unlock;
> + }
> + /* get no of available bytes to read data from buffer */
> + avail = spa_ringbuffer_get_read_index(&v->ring, &index);
> +
> + if (avail < (int32_t) len) {
> + len = avail;
> + }
> +
> + spa_ringbuffer_read_data(&v->ring,
> + v->buffer, RINGBUFFER_SIZE,
> + index & RINGBUFFER_MASK, data, len);
> + index += len;
> + spa_ringbuffer_read_update(&v->ring, index);
> + l = len;
> +
> +done_unlock:
> + pw_thread_loop_unlock(c->thread_loop);
> + return l;
> +}
> +
> +static size_t
> +qpw_write(HWVoiceOut *hw, void *data, size_t len)
> +{
> + PWVoiceOut *pw = (PWVoiceOut *) hw;
> + PWVoice *v = &pw->v;
> + pwaudio *c = v->g;
> + const char *error = NULL;
> + const int periods = 3;
> + size_t l;
> + int32_t filled, avail;
> + uint32_t index;
> +
> + pw_thread_loop_lock(c->thread_loop);
> + if (pw_stream_get_state(v->stream, &error) != PW_STREAM_STATE_STREAMING) {
> + /* wait for stream to become ready */
> + l = 0;
> + goto done_unlock;
> + }
> + filled = spa_ringbuffer_get_write_index(&v->ring, &index);
> +
> + avail = BUFFER_SAMPLES * v->frame_size * periods - filled;
> +
> + pw_log_debug("%u %u %u %zu", filled, avail, index, len);
use trace
> +
> + if (len > avail) {
> + len = avail;
> + }
> +
> + if (filled < 0) {
> + pw_log_warn("%p: underrun write:%u filled:%d", pw, index, filled);
> + } else {
> + if ((uint32_t) filled + len > RINGBUFFER_SIZE) {
> + pw_log_warn("%p: overrun write:%u filled:%d + size:%zu > max:%u",
> + pw, index, filled, len, RINGBUFFER_SIZE);
> + }
> + }
> +
> + spa_ringbuffer_write_data(&v->ring,
> + v->buffer, RINGBUFFER_SIZE,
> + index & RINGBUFFER_MASK, data, len);
> + index += len;
> + spa_ringbuffer_write_update(&v->ring, index);
> + l = len;
> +
> +done_unlock:
> + pw_thread_loop_unlock(c->thread_loop);
> + return l;
> +}
> +
> +static int
> +audfmt_to_pw(AudioFormat fmt, int endianness)
> +{
> + int format;
> +
> + switch (fmt) {
> + case AUDIO_FORMAT_S8:
> + format = SPA_AUDIO_FORMAT_S8;
> + break;
> + case AUDIO_FORMAT_U8:
> + format = SPA_AUDIO_FORMAT_U8;
> + break;
> + case AUDIO_FORMAT_S16:
> + format = endianness ? SPA_AUDIO_FORMAT_S16_BE : SPA_AUDIO_FORMAT_S16_LE;
> + break;
> + case AUDIO_FORMAT_U16:
> + format = endianness ? SPA_AUDIO_FORMAT_U16_BE : SPA_AUDIO_FORMAT_U16_LE;
> + break;
> + case AUDIO_FORMAT_S32:
> + format = endianness ? SPA_AUDIO_FORMAT_S32_BE : SPA_AUDIO_FORMAT_S32_LE;
> + break;
> + case AUDIO_FORMAT_U32:
> + format = endianness ? SPA_AUDIO_FORMAT_U32_BE : SPA_AUDIO_FORMAT_U32_LE;
> + break;
> + case AUDIO_FORMAT_F32:
> + format = endianness ? SPA_AUDIO_FORMAT_F32_BE : SPA_AUDIO_FORMAT_F32_LE;
> + break;
> + default:
> + dolog("Internal logic error: Bad audio format %d\n", fmt);
> + format = SPA_AUDIO_FORMAT_U8;
> + break;
> + }
> + return format;
> +}
> +
> +static AudioFormat
> +pw_to_audfmt(enum spa_audio_format fmt, int *endianness,
> + uint32_t *frame_size)
> +{
> + switch (fmt) {
> + case SPA_AUDIO_FORMAT_S8:
> + *frame_size = 1;
> + return AUDIO_FORMAT_S8;
> + case SPA_AUDIO_FORMAT_U8:
> + *frame_size = 1;
> + return AUDIO_FORMAT_U8;
> + case SPA_AUDIO_FORMAT_S16_BE:
> + *frame_size = 2;
> + *endianness = 1;
> + return AUDIO_FORMAT_S16;
> + case SPA_AUDIO_FORMAT_S16_LE:
> + *frame_size = 2;
> + *endianness = 0;
> + return AUDIO_FORMAT_S16;
> + case SPA_AUDIO_FORMAT_U16_BE:
> + *frame_size = 2;
> + *endianness = 1;
> + return AUDIO_FORMAT_U16;
> + case SPA_AUDIO_FORMAT_U16_LE:
> + *frame_size = 2;
> + *endianness = 0;
> + return AUDIO_FORMAT_U16;
> + case SPA_AUDIO_FORMAT_S32_BE:
> + *frame_size = 4;
> + *endianness = 1;
> + return AUDIO_FORMAT_S32;
> + case SPA_AUDIO_FORMAT_S32_LE:
> + *frame_size = 4;
> + *endianness = 0;
> + return AUDIO_FORMAT_S32;
> + case SPA_AUDIO_FORMAT_U32_BE:
> + *frame_size = 4;
> + *endianness = 1;
> + return AUDIO_FORMAT_U32;
> + case SPA_AUDIO_FORMAT_U32_LE:
> + *frame_size = 4;
> + *endianness = 0;
> + return AUDIO_FORMAT_U32;
> + case SPA_AUDIO_FORMAT_F32_BE:
> + *frame_size = 4;
> + *endianness = 1;
> + return AUDIO_FORMAT_F32;
> + case SPA_AUDIO_FORMAT_F32_LE:
> + *frame_size = 4;
> + *endianness = 0;
> + return AUDIO_FORMAT_F32;
> + default:
> + *frame_size = 1;
> + dolog("Internal logic error: Bad spa_audio_format %d\n", fmt);
> + return AUDIO_FORMAT_U8;
> + }
> +}
> +
> +static int
> +create_stream(pwaudio *c, PWVoice *v, const char *name)
> +{
> + int res;
> + uint32_t n_params;
> + const struct spa_pod *params[2];
> + uint8_t buffer[1024];
> + struct spa_pod_builder b;
> +
> + v->stream = pw_stream_new(c->core, name, NULL);
> +
> + if (v->stream == NULL) {
> + res = -errno;
> + goto error;
> + }
> +
> + if (v->mode == MODE_SOURCE) {
> + pw_stream_add_listener(v->stream,
> + &v->stream_listener, &capture_stream_events, v);
> + } else {
> + pw_stream_add_listener(v->stream,
> + &v->stream_listener, &playback_stream_events, v);
> + }
> +
> + n_params = 0;
> + spa_pod_builder_init(&b, buffer, sizeof(buffer));
> + params[n_params++] = spa_format_audio_raw_build(&b,
> + SPA_PARAM_EnumFormat,
> + &v->info);
> +
> + /* connect the stream to a sink or source */
> + res = pw_stream_connect(v->stream,
> + v->mode ==
> + MODE_SOURCE ? PW_DIRECTION_INPUT :
> + PW_DIRECTION_OUTPUT, PW_ID_ANY,
> + PW_STREAM_FLAG_AUTOCONNECT |
> + PW_STREAM_FLAG_MAP_BUFFERS |
> + PW_STREAM_FLAG_RT_PROCESS, params, n_params);
> + if (res < 0) {
> + goto error;
> + }
> +
> + return 0;
> +error:
> + return res;
You can return directly instead.
> +}
> +
> +static void
> +pw_destroy(pwaudio *c)
> +{
> + if (c->thread_loop) {
> + pw_thread_loop_stop(c->thread_loop);
> + }
> + if (c->core) {
> + pw_core_disconnect(c->core);
> + }
> +
> + g_free(c);
> +}
> +
> +static int
> +qpw_stream_new(pwaudio *c, PWVoice *v, const char *name)
> +{
> + int r;
> +
> + pw_thread_loop_lock(c->thread_loop);
You already have the lock when called.
> +
> + switch (v->info.channels) {
> + case 8:
> + v->info.position[0] = SPA_AUDIO_CHANNEL_FL;
> + v->info.position[1] = SPA_AUDIO_CHANNEL_FR;
> + v->info.position[2] = SPA_AUDIO_CHANNEL_FC;
> + v->info.position[3] = SPA_AUDIO_CHANNEL_LFE;
> + v->info.position[4] = SPA_AUDIO_CHANNEL_RL;
> + v->info.position[5] = SPA_AUDIO_CHANNEL_RR;
> + v->info.position[6] = SPA_AUDIO_CHANNEL_SL;
> + v->info.position[7] = SPA_AUDIO_CHANNEL_SR;
> + break;
> + case 6:
> + v->info.position[0] = SPA_AUDIO_CHANNEL_FL;
> + v->info.position[1] = SPA_AUDIO_CHANNEL_FR;
> + v->info.position[2] = SPA_AUDIO_CHANNEL_FC;
> + v->info.position[3] = SPA_AUDIO_CHANNEL_LFE;
> + v->info.position[4] = SPA_AUDIO_CHANNEL_RL;
> + v->info.position[5] = SPA_AUDIO_CHANNEL_RR;
> + break;
> + case 5:
> + v->info.position[0] = SPA_AUDIO_CHANNEL_FL;
> + v->info.position[1] = SPA_AUDIO_CHANNEL_FR;
> + v->info.position[2] = SPA_AUDIO_CHANNEL_FC;
> + v->info.position[3] = SPA_AUDIO_CHANNEL_LFE;
> + v->info.position[4] = SPA_AUDIO_CHANNEL_RC;
> + break;
> + case 4:
> + v->info.position[0] = SPA_AUDIO_CHANNEL_FL;
> + v->info.position[1] = SPA_AUDIO_CHANNEL_FR;
> + v->info.position[2] = SPA_AUDIO_CHANNEL_FC;
> + v->info.position[3] = SPA_AUDIO_CHANNEL_RC;
> + break;
> + case 3:
> + v->info.position[0] = SPA_AUDIO_CHANNEL_FL;
> + v->info.position[1] = SPA_AUDIO_CHANNEL_FR;
> + v->info.position[2] = SPA_AUDIO_CHANNEL_LFE;
> + break;
> + case 2:
> + v->info.position[0] = SPA_AUDIO_CHANNEL_FL;
> + v->info.position[1] = SPA_AUDIO_CHANNEL_FR;
> + break;
> + case 1:
> + v->info.position[0] = SPA_AUDIO_CHANNEL_MONO;
> + break;
> + default:
> + for (size_t i = 0; i < v->info.channels; i++) {
> + v->info.position[i] = SPA_AUDIO_CHANNEL_UNKNOWN;
> + }
> + break;
> + }
> +
> + /* create a new unconnected pwstream */
> + r = create_stream(c, v, name);
> + if (r < 0) {
> + goto error;
> + }
> +
> + pw_thread_loop_unlock(c->thread_loop);
> + return r;
> +
> +error:
> + AUD_log(AUDIO_CAP, "Failed to create stream.");
> + pw_thread_loop_unlock(c->thread_loop);
> + pw_destroy(c);
You are freeing c (and other data) on stream creation error. This will
later crash during fini(), or other events, I guess.
There is a single "goto error", you could move it there.
> + return -1;
> +}
> +
> +static int
> +qpw_init_out(HWVoiceOut *hw, struct audsettings *as, void *drv_opaque)
> +{
> + PWVoiceOut *pw = (PWVoiceOut *) hw;
> + PWVoice *v = &pw->v;
> + struct audsettings obt_as = *as;
> + pwaudio *c = v->g = drv_opaque;
> + AudiodevPipewireOptions *popts = &c->dev->u.pipewire;
> + AudiodevPipewirePerDirectionOptions *ppdo = popts->out;
> + int r;
> + v->enabled = false;
> +
> + v->mode = MODE_SINK;
> +
> + pw_thread_loop_lock(c->thread_loop);
> +
> + v->info.format = audfmt_to_pw(as->fmt, as->endianness);
> + v->info.channels = as->nchannels;
> + v->info.rate = as->freq;
> +
> + obt_as.fmt =
> + pw_to_audfmt(v->info.format, &obt_as.endianness, &v->frame_size);
> + v->frame_size *= as->nchannels;
> +
> + /* call the function that creates a new stream for playback */
> + r = qpw_stream_new(c, v, ppdo->stream_name ? : c->dev->id);
> + if (r < 0) {
> + pw_log_error("qpw_stream_new for playback failed\n ");
Use error_report() instead
> + 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;
There is a single "goto fail", you could move that here.
> +}
> +
> +static int
> +qpw_init_in(HWVoiceIn *hw, struct audsettings *as, void *drv_opaque)
> +{
> + PWVoiceIn *pw = (PWVoiceIn *) hw;
> + PWVoice *v = &pw->v;
> + struct audsettings obt_as = *as;
> + pwaudio *c = v->g = drv_opaque;
> + AudiodevPipewireOptions *popts = &c->dev->u.pipewire;
> + AudiodevPipewirePerDirectionOptions *ppdo = popts->in;
> + int r;
> + v->enabled = false;
> +
> + v->mode = MODE_SOURCE;
> + pw_thread_loop_lock(c->thread_loop);
> +
> + v->info.format = audfmt_to_pw(as->fmt, as->endianness);
> + v->info.channels = as->nchannels;
> + v->info.rate = as->freq;
> +
> + obt_as.fmt =
> + pw_to_audfmt(v->info.format, &obt_as.endianness, &v->frame_size);
> + v->frame_size *= as->nchannels;
> +
> + /* call the function that creates a new stream for recording */
> + r = qpw_stream_new(c, v, ppdo->stream_name ? : c->dev->id);
> + if (r < 0) {
> + pw_log_error("qpw_stream_new for recording failed\n ");
Use error_report() instead
> + 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;
There is a single "goto fail", you could move that here.
> +}
> +
> +static void
> +qpw_fini_out(HWVoiceOut *hw)
> +{
> + PWVoiceOut *pw = (PWVoiceOut *) hw;
> + PWVoice *v = &pw->v;
> +
> + if (v->stream) {
> + pwaudio *c = v->g;
> + pw_thread_loop_lock(c->thread_loop);
> + pw_stream_destroy(v->stream);
> + v->stream = NULL;
> + pw_thread_loop_unlock(c->thread_loop);
> + }
> +}
> +
> +static void
> +qpw_fini_in(HWVoiceIn *hw)
> +{
> + PWVoiceIn *pw = (PWVoiceIn *) hw;
> + PWVoice *v = &pw->v;
> +
> + if (v->stream) {
> + pwaudio *c = v->g;
> + pw_thread_loop_lock(c->thread_loop);
> + pw_stream_destroy(v->stream);
> + v->stream = NULL;
> + pw_thread_loop_unlock(c->thread_loop);
> + }
> +}
> +
> +static void
> +qpw_enable_out(HWVoiceOut *hw, bool enable)
> +{
> + PWVoiceOut *po = (PWVoiceOut *) hw;
> + PWVoice *v = &po->v;
> + v->enabled = enable;
> +}
> +
> +static void
> +qpw_enable_in(HWVoiceIn *hw, bool enable)
> +{
> + PWVoiceIn *pi = (PWVoiceIn *) hw;
> + PWVoice *v = &pi->v;
> + v->enabled = enable;
> +}
> +
> +static void
> +on_core_error(void *data, uint32_t id, int seq, int res, const char *message)
> +{
> + pwaudio *pw = data;
> +
> + pw_log_warn("error id:%u seq:%d res:%d (%s): %s",
> + id, seq, res, spa_strerror(res), message);
Use error_report() instead
> +
> + 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);
What are those thread_loop_signal() for? Maybe leave a comment?
> + }
> +}
> +
> +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;
> + pw_log_debug("Initialize PW context\n");
Can you replace the logging with traces? That's what QEMU uses.
> + assert(dev->driver == AUDIODEV_DRIVER_PIPEWIRE);
> + popts = &dev->u.pipewire;
> +
> + if (!popts->has_latency) {
> + popts->has_latency = true;
> + popts->latency = 15000;
> + }
> +
> + pw->dev = dev;
> + pw->thread_loop = pw_thread_loop_new("Pipewire thread loop", NULL);
> + if (pw->thread_loop == NULL) {
> + goto fail;
> + }
Either we handle allocation failures everywhere, or we don't. Unless
the documentation is explicit about handling allocation/initialization
errors, I would say we don't have to, in init/fini().
> + pw->context =
> + pw_context_new(pw_thread_loop_get_loop(pw->thread_loop), NULL, 0);
> +
Why not handle errors here?
> + if (pw_thread_loop_start(pw->thread_loop) < 0) {
> + goto fail;
> + }
> +
> + pw_thread_loop_lock(pw->thread_loop);
> +
> + pw->core = pw_context_connect(pw->context, NULL, 0);
> + if (pw->core == NULL) {
> + goto fail;
> + }
> +
> + pw_core_add_listener(pw->core, &pw->core_listener, &core_events, pw);
> +
> + pw_thread_loop_unlock(pw->thread_loop);
> +
> + return pw;
> +
> +fail:
> + AUD_log(AUDIO_CAP, "Failed to initialize PW context");
> + pw_thread_loop_unlock(pw->thread_loop);
Bad after first "goto fail"
> + pw_context_destroy(pw->context);
> + pw_thread_loop_destroy(pw->thread_loop);
> + g_free(pw);
> + return NULL;
> +}
> +
> +static void
> +qpw_audio_fini(void *opaque)
> +{
> + pwaudio *pw = opaque;
> +
> + pw_thread_loop_stop(pw->thread_loop);
> +
> + if (pw->core) {
> + spa_hook_remove(&pw->core_listener);
> + spa_zero(pw->core_listener);
> + pw_core_disconnect(pw->core);
> + }
> +
> + if (pw->context) {
> + pw_context_destroy(pw->context);
> + }
> + pw_thread_loop_destroy(pw->thread_loop);
> +
> + g_free(pw);
> +}
> +
> +static struct audio_pcm_ops qpw_pcm_ops = {
> + .init_out = qpw_init_out,
> + .fini_out = qpw_fini_out,
> + .write = qpw_write,
> + .buffer_get_free = audio_generic_buffer_get_free,
> + .run_buffer_out = audio_generic_run_buffer_out,
> + .enable_out = qpw_enable_out,
> +
> + .init_in = qpw_init_in,
> + .fini_in = qpw_fini_in,
> + .read = qpw_read,
> + .run_buffer_in = audio_generic_run_buffer_in,
> + .enable_in = qpw_enable_in
> +};
> +
> +static struct audio_driver pw_audio_driver = {
> + .name = "pipewire",
> + .descr = "http://www.pipewire.org/",
> + .init = qpw_audio_init,
> + .fini = qpw_audio_fini,
> + .pcm_ops = &qpw_pcm_ops,
> + .can_be_default = 1,
> + .max_voices_out = INT_MAX,
> + .max_voices_in = INT_MAX,
> + .voice_size_out = sizeof(PWVoiceOut),
> + .voice_size_in = sizeof(PWVoiceIn),
> +};
> +
> +static void
> +register_audio_pw(void)
> +{
> + audio_driver_register(&pw_audio_driver);
> +}
> +
> +type_init(register_audio_pw);
> diff --git a/meson.build b/meson.build
> index 6cb2b1a42f..2aa0b397ea 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(),
> }
> @@ -3978,6 +3985,7 @@ 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 6b0900205e..6c5f260a6a 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)
> +#
> +#
> +# 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 beeb4475ba..2fd88ccf15 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 5d969a94c0..236a714087 100644
> --- a/scripts/meson-buildoptions.sh
> +++ b/scripts/meson-buildoptions.sh
> @@ -1,7 +1,8 @@
> # This file is generated by meson-buildoptions.py, do not edit!
> meson_options_help() {
> - printf "%s\n" ' --audio-drv-list=CHOICES Set audio driver list [default] (choices: alsa/co'
> - printf "%s\n" ' reaudio/default/dsound/jack/oss/pa/sdl/sndio)'
> + printf "%s\n" ' --audio-drv-list=CHOICES Set audio driver list [default] (choices: al'
> + printf "%s\n" ' sa/coreaudio/default/dsound/jack/oss/pa/'
> + printf "%s\n" ' pipewire/sdl/sndio)'
> printf "%s\n" ' --block-drv-ro-whitelist=VALUE'
> printf "%s\n" ' set block driver read-only whitelist (by default'
> printf "%s\n" ' affects only QEMU, not tools like qemu-img)'
> @@ -135,6 +136,7 @@ meson_options_help() {
> printf "%s\n" ' oss OSS sound support'
> printf "%s\n" ' pa PulseAudio sound support'
> printf "%s\n" ' parallels parallels image format support'
> + printf "%s\n" ' pipewire Pipewire sound support'
> printf "%s\n" ' png PNG support with libpng'
> printf "%s\n" ' pvrdma Enable PVRDMA support'
> printf "%s\n" ' qcow1 qcow1 image format support'
> @@ -369,6 +371,8 @@ _meson_option_parse() {
> --disable-pa) printf "%s" -Dpa=disabled ;;
> --enable-parallels) printf "%s" -Dparallels=enabled ;;
> --disable-parallels) printf "%s" -Dparallels=disabled ;;
> + --enable-pipewire) printf "%s" -Dpipewire=enabled ;;
> + --disable-pipewire) printf "%s" -Dpipewire=disabled ;;
> --with-pkgversion=*) quote_sh "-Dpkgversion=$2" ;;
> --enable-png) printf "%s" -Dpng=enabled ;;
> --disable-png) printf "%s" -Dpng=disabled ;;
> --
> 2.39.1
>
>
--
Marc-André Lureau
Thanks for the feedback!
your comments have been addressed in the latest patch version.
What are those thread_loop_signal() for? Maybe leave a comment?
the explanation of the function is in the reference header file.
regards,
Dorinda.
On Wed, Mar 1, 2023 at 12:08 PM Marc-André Lureau <
marcandre.lureau@gmail.com> wrote:
> Hi Dorinda
>
> On Tue, Feb 28, 2023 at 2:01 AM 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>
> >
>
> I didn't look at buffering/latency details yet, but here are some
> other comments for this iteration.
>
> thanks
>
> ---
> > v5:
> > silence output to the console, use pw debug log
> > use SPDX identifier
> > fix typo
> > change version release
> >
> > audio/audio.c | 3 +
> > audio/audio_template.h | 4 +
> > audio/meson.build | 1 +
> > audio/pwaudio.c | 811 ++++++++++++++++++++++++++++++++++
> > meson.build | 8 +
> > meson_options.txt | 4 +-
> > qapi/audio.json | 45 ++
> > qemu-options.hx | 17 +
> > scripts/meson-buildoptions.sh | 8 +-
> > 9 files changed, 898 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..7448f0abd0
> > --- /dev/null
> > +++ b/audio/pwaudio.c
> > @@ -0,0 +1,811 @@
> > +/*
> > + * 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 <spa/param/audio/format-utils.h>
> > +#include <spa/utils/ringbuffer.h>
> > +#include <spa/utils/result.h>
> > +
> > +#include <pipewire/pipewire.h>
> > +
> > +#define AUDIO_CAP "pipewire"
> > +#define RINGBUFFER_SIZE (1u << 22)
> > +#define RINGBUFFER_MASK (RINGBUFFER_SIZE - 1)
> > +#define BUFFER_SAMPLES 512
> > +
> > +#include "audio_int.h"
> > +
> > +enum {
> > + MODE_SINK,
> > + MODE_SOURCE
> > +};
> > +
> > +typedef struct pwaudio {
> > + Audiodev *dev;
> > + struct pw_thread_loop *thread_loop;
> > + struct pw_context *context;
> > +
> > + struct pw_core *core;
> > + struct spa_hook core_listener;
> > + int seq;
> > +} pwaudio;
> > +
> > +typedef struct PWVoice {
> > + pwaudio *g;
> > + bool enabled;
> > + struct pw_stream *stream;
> > + struct spa_hook stream_listener;
> > + struct spa_audio_info_raw info;
> > + uint32_t frame_size;
> > + struct spa_ringbuffer ring;
> > + uint8_t buffer[RINGBUFFER_SIZE];
> > +
> > + uint32_t mode;
> > + struct pw_properties *props;
> > +} PWVoice;
> > +
> > +typedef struct PWVoiceOut {
> > + HWVoiceOut hw;
> > + PWVoice v;
> > +} PWVoiceOut;
> > +
> > +typedef struct PWVoiceIn {
> > + HWVoiceIn hw;
> > + PWVoice v;
> > +} PWVoiceIn;
> > +
> > +static void
> > +stream_destroy(void *data)
> > +{
> > + PWVoice *v = (PWVoice *) data;
> > + spa_hook_remove(&v->stream_listener);
> > + v->stream = NULL;
> > +}
> > +
> > +/* output data processing function to read stuffs from the buffer */
> > +static void
> > +playback_on_process(void *data)
> > +{
> > + PWVoice *v = (PWVoice *) data;
> > + void *p;
> > + struct pw_buffer *b;
> > + struct spa_buffer *buf;
> > + uint32_t n_frames, req, index, n_bytes;
> > + int32_t avail;
> > +
> > + if (!v->stream) {
> > + return;
> > + }
> > +
> > + /* obtain a buffer to read from */
> > + b = pw_stream_dequeue_buffer(v->stream);
> > + if (b == NULL) {
> > + pw_log_warn("out of buffers: %m");
>
> We don't use %m in QEMU, but error_report("..%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;
> > + }
> > + n_frames = SPA_MIN(req, buf->datas[0].maxsize);
> > + n_bytes = n_frames * v->frame_size;
> > +
> > + /* get no of available bytes to read data from buffer */
> > +
> > + avail = spa_ringbuffer_get_read_index(&v->ring, &index);
> > +
> > + if (!v->enabled) {
> > + avail = 0;
> > + }
> > +
> > + if (avail == 0) {
> > + memset(p, 0, n_bytes);
> > + } else {
> > + if (avail < (int32_t) n_bytes) {
> > + n_bytes = avail;
> > + }
> > +
> > + spa_ringbuffer_read_data(&v->ring,
> > + v->buffer, RINGBUFFER_SIZE,
> > + index & RINGBUFFER_MASK, p,
> n_bytes);
> > +
> > + index += n_bytes;
> > + spa_ringbuffer_read_update(&v->ring, index);
> > + }
> > +
> > + buf->datas[0].chunk->offset = 0;
> > + buf->datas[0].chunk->stride = v->frame_size;
> > + buf->datas[0].chunk->size = n_bytes;
> > +
> > + /* queue the buffer for playback */
> > + pw_stream_queue_buffer(v->stream, b);
> > +}
> > +
> > +/* output data processing function to generate stuffs in the buffer */
> > +static void
> > +capture_on_process(void *data)
> > +{
> > + PWVoice *v = (PWVoice *) data;
> > + void *p;
> > + struct pw_buffer *b;
> > + struct spa_buffer *buf;
> > + int32_t filled;
> > + uint32_t index, offs, n_bytes;
> > +
> > + if (!v->stream) {
> > + return;
> > + }
> > +
> > + /* obtain a buffer */
> > + b = pw_stream_dequeue_buffer(v->stream);
> > + if (b == NULL) {
> > + pw_log_warn("out of buffers: %m");
> > + return;
> > + }
> > +
> > + /* Write data into buffer */
> > + buf = b->buffer;
> > + p = buf->datas[0].data;
> > + if (p == NULL) {
> > + return;
> > + }
> > + offs = SPA_MIN(buf->datas[0].chunk->offset, buf->datas[0].maxsize);
> > + n_bytes = SPA_MIN(buf->datas[0].chunk->size, buf->datas[0].maxsize
> - offs);
> > +
> > + filled = spa_ringbuffer_get_write_index(&v->ring, &index);
> > +
> > + if (!v->enabled) {
> > + n_bytes = 0;
> > + }
> > +
> > + if (filled < 0) {
> > + pw_log_warn("%p: underrun write:%u filled:%d", p, index,
> filled);
> > + } else {
> > + if ((uint32_t) filled + n_bytes > RINGBUFFER_SIZE) {
> > + pw_log_warn("%p: overrun write:%u filled:%d + size:%u >
> max:%u",
> > + p, index, filled, n_bytes, RINGBUFFER_SIZE);
> > + }
> > + }
> > + spa_ringbuffer_write_data(&v->ring,
> > + v->buffer, RINGBUFFER_SIZE,
> > + index & RINGBUFFER_MASK,
> > + SPA_PTROFF(p, offs, void), n_bytes);
> > + index += n_bytes;
> > + spa_ringbuffer_write_update(&v->ring, index);
> > +
> > + /* queue the buffer for playback */
> > + pw_stream_queue_buffer(v->stream, b);
> > +}
> > +
> > +static void
> > +on_stream_state_changed(void *_data, enum pw_stream_state old,
> > + enum pw_stream_state state, const char *error)
> > +{
> > + PWVoice *v = (PWVoice *) _data;
> > +
> > + pw_log_debug("stream state: \"%s\"\n",
> pw_stream_state_as_string(state));
> > +
> > + switch (state) {
> > + case PW_STREAM_STATE_ERROR:
> > + case PW_STREAM_STATE_UNCONNECTED:
> > + {
> > + break;
> > + }
> > + case PW_STREAM_STATE_PAUSED:
> > + pw_log_debug("node id: %d\n", pw_stream_get_node_id(v->stream));
> > + break;
> > + case PW_STREAM_STATE_CONNECTING:
> > + case PW_STREAM_STATE_STREAMING:
> > + break;
> > + }
> > +}
> > +
> > +static const struct pw_stream_events capture_stream_events = {
> > + PW_VERSION_STREAM_EVENTS,
> > + .destroy = stream_destroy,
> > + .state_changed = on_stream_state_changed,
> > + .process = capture_on_process
> > +};
> > +
> > +static const struct pw_stream_events playback_stream_events = {
> > + PW_VERSION_STREAM_EVENTS,
> > + .destroy = stream_destroy,
> > + .state_changed = on_stream_state_changed,
> > + .process = playback_on_process
> > +};
> > +
> > +static size_t
> > +qpw_read(HWVoiceIn *hw, void *data, size_t len)
> > +{
> > + PWVoiceIn *pw = (PWVoiceIn *) hw;
> > + PWVoice *v = &pw->v;
> > + pwaudio *c = v->g;
> > + const char *error = NULL;
> > + size_t l;
> > + int32_t avail;
> > + uint32_t index;
> > +
> > + pw_thread_loop_lock(c->thread_loop);
> > + if (pw_stream_get_state(v->stream, &error) !=
> PW_STREAM_STATE_STREAMING) {
> > + /* wait for stream to become ready */
> > + l = 0;
> > + goto done_unlock;
> > + }
> > + /* get no of available bytes to read data from buffer */
> > + avail = spa_ringbuffer_get_read_index(&v->ring, &index);
> > +
> > + if (avail < (int32_t) len) {
> > + len = avail;
> > + }
> > +
> > + spa_ringbuffer_read_data(&v->ring,
> > + v->buffer, RINGBUFFER_SIZE,
> > + index & RINGBUFFER_MASK, data, len);
> > + index += len;
> > + spa_ringbuffer_read_update(&v->ring, index);
> > + l = len;
> > +
> > +done_unlock:
> > + pw_thread_loop_unlock(c->thread_loop);
> > + return l;
> > +}
> > +
> > +static size_t
> > +qpw_write(HWVoiceOut *hw, void *data, size_t len)
> > +{
> > + PWVoiceOut *pw = (PWVoiceOut *) hw;
> > + PWVoice *v = &pw->v;
> > + pwaudio *c = v->g;
> > + const char *error = NULL;
> > + const int periods = 3;
> > + size_t l;
> > + int32_t filled, avail;
> > + uint32_t index;
> > +
> > + pw_thread_loop_lock(c->thread_loop);
> > + if (pw_stream_get_state(v->stream, &error) !=
> PW_STREAM_STATE_STREAMING) {
> > + /* wait for stream to become ready */
> > + l = 0;
> > + goto done_unlock;
> > + }
> > + filled = spa_ringbuffer_get_write_index(&v->ring, &index);
> > +
> > + avail = BUFFER_SAMPLES * v->frame_size * periods - filled;
> > +
> > + pw_log_debug("%u %u %u %zu", filled, avail, index, len);
>
> use trace
>
> > +
> > + if (len > avail) {
> > + len = avail;
> > + }
> > +
> > + if (filled < 0) {
> > + pw_log_warn("%p: underrun write:%u filled:%d", pw, index,
> filled);
> > + } else {
> > + if ((uint32_t) filled + len > RINGBUFFER_SIZE) {
> > + pw_log_warn("%p: overrun write:%u filled:%d + size:%zu >
> max:%u",
> > + pw, index, filled, len, RINGBUFFER_SIZE);
> > + }
> > + }
> > +
> > + spa_ringbuffer_write_data(&v->ring,
> > + v->buffer, RINGBUFFER_SIZE,
> > + index & RINGBUFFER_MASK, data, len);
> > + index += len;
> > + spa_ringbuffer_write_update(&v->ring, index);
> > + l = len;
> > +
> > +done_unlock:
> > + pw_thread_loop_unlock(c->thread_loop);
> > + return l;
> > +}
> > +
> > +static int
> > +audfmt_to_pw(AudioFormat fmt, int endianness)
> > +{
> > + int format;
> > +
> > + switch (fmt) {
> > + case AUDIO_FORMAT_S8:
> > + format = SPA_AUDIO_FORMAT_S8;
> > + break;
> > + case AUDIO_FORMAT_U8:
> > + format = SPA_AUDIO_FORMAT_U8;
> > + break;
> > + case AUDIO_FORMAT_S16:
> > + format = endianness ? SPA_AUDIO_FORMAT_S16_BE :
> SPA_AUDIO_FORMAT_S16_LE;
> > + break;
> > + case AUDIO_FORMAT_U16:
> > + format = endianness ? SPA_AUDIO_FORMAT_U16_BE :
> SPA_AUDIO_FORMAT_U16_LE;
> > + break;
> > + case AUDIO_FORMAT_S32:
> > + format = endianness ? SPA_AUDIO_FORMAT_S32_BE :
> SPA_AUDIO_FORMAT_S32_LE;
> > + break;
> > + case AUDIO_FORMAT_U32:
> > + format = endianness ? SPA_AUDIO_FORMAT_U32_BE :
> SPA_AUDIO_FORMAT_U32_LE;
> > + break;
> > + case AUDIO_FORMAT_F32:
> > + format = endianness ? SPA_AUDIO_FORMAT_F32_BE :
> SPA_AUDIO_FORMAT_F32_LE;
> > + break;
> > + default:
> > + dolog("Internal logic error: Bad audio format %d\n", fmt);
> > + format = SPA_AUDIO_FORMAT_U8;
> > + break;
> > + }
> > + return format;
> > +}
> > +
> > +static AudioFormat
> > +pw_to_audfmt(enum spa_audio_format fmt, int *endianness,
> > + uint32_t *frame_size)
> > +{
> > + switch (fmt) {
> > + case SPA_AUDIO_FORMAT_S8:
> > + *frame_size = 1;
> > + return AUDIO_FORMAT_S8;
> > + case SPA_AUDIO_FORMAT_U8:
> > + *frame_size = 1;
> > + return AUDIO_FORMAT_U8;
> > + case SPA_AUDIO_FORMAT_S16_BE:
> > + *frame_size = 2;
> > + *endianness = 1;
> > + return AUDIO_FORMAT_S16;
> > + case SPA_AUDIO_FORMAT_S16_LE:
> > + *frame_size = 2;
> > + *endianness = 0;
> > + return AUDIO_FORMAT_S16;
> > + case SPA_AUDIO_FORMAT_U16_BE:
> > + *frame_size = 2;
> > + *endianness = 1;
> > + return AUDIO_FORMAT_U16;
> > + case SPA_AUDIO_FORMAT_U16_LE:
> > + *frame_size = 2;
> > + *endianness = 0;
> > + return AUDIO_FORMAT_U16;
> > + case SPA_AUDIO_FORMAT_S32_BE:
> > + *frame_size = 4;
> > + *endianness = 1;
> > + return AUDIO_FORMAT_S32;
> > + case SPA_AUDIO_FORMAT_S32_LE:
> > + *frame_size = 4;
> > + *endianness = 0;
> > + return AUDIO_FORMAT_S32;
> > + case SPA_AUDIO_FORMAT_U32_BE:
> > + *frame_size = 4;
> > + *endianness = 1;
> > + return AUDIO_FORMAT_U32;
> > + case SPA_AUDIO_FORMAT_U32_LE:
> > + *frame_size = 4;
> > + *endianness = 0;
> > + return AUDIO_FORMAT_U32;
> > + case SPA_AUDIO_FORMAT_F32_BE:
> > + *frame_size = 4;
> > + *endianness = 1;
> > + return AUDIO_FORMAT_F32;
> > + case SPA_AUDIO_FORMAT_F32_LE:
> > + *frame_size = 4;
> > + *endianness = 0;
> > + return AUDIO_FORMAT_F32;
> > + default:
> > + *frame_size = 1;
> > + dolog("Internal logic error: Bad spa_audio_format %d\n", fmt);
> > + return AUDIO_FORMAT_U8;
> > + }
> > +}
> > +
> > +static int
> > +create_stream(pwaudio *c, PWVoice *v, const char *name)
> > +{
> > + int res;
> > + uint32_t n_params;
> > + const struct spa_pod *params[2];
> > + uint8_t buffer[1024];
> > + struct spa_pod_builder b;
> > +
> > + v->stream = pw_stream_new(c->core, name, NULL);
> > +
> > + if (v->stream == NULL) {
> > + res = -errno;
> > + goto error;
> > + }
> > +
> > + if (v->mode == MODE_SOURCE) {
> > + pw_stream_add_listener(v->stream,
> > + &v->stream_listener,
> &capture_stream_events, v);
> > + } else {
> > + pw_stream_add_listener(v->stream,
> > + &v->stream_listener,
> &playback_stream_events, v);
> > + }
> > +
> > + n_params = 0;
> > + spa_pod_builder_init(&b, buffer, sizeof(buffer));
> > + params[n_params++] = spa_format_audio_raw_build(&b,
> > + SPA_PARAM_EnumFormat,
> > + &v->info);
> > +
> > + /* connect the stream to a sink or source */
> > + res = pw_stream_connect(v->stream,
> > + v->mode ==
> > + MODE_SOURCE ? PW_DIRECTION_INPUT :
> > + PW_DIRECTION_OUTPUT, PW_ID_ANY,
> > + PW_STREAM_FLAG_AUTOCONNECT |
> > + PW_STREAM_FLAG_MAP_BUFFERS |
> > + PW_STREAM_FLAG_RT_PROCESS, params,
> n_params);
> > + if (res < 0) {
> > + goto error;
> > + }
> > +
> > + return 0;
> > +error:
> > + return res;
>
> You can return directly instead.
>
> > +}
> > +
> > +static void
> > +pw_destroy(pwaudio *c)
> > +{
> > + if (c->thread_loop) {
> > + pw_thread_loop_stop(c->thread_loop);
> > + }
> > + if (c->core) {
> > + pw_core_disconnect(c->core);
> > + }
> > +
> > + g_free(c);
> > +}
> > +
> > +static int
> > +qpw_stream_new(pwaudio *c, PWVoice *v, const char *name)
> > +{
> > + int r;
> > +
> > + pw_thread_loop_lock(c->thread_loop);
>
> You already have the lock when called.
>
> > +
> > + switch (v->info.channels) {
> > + case 8:
> > + v->info.position[0] = SPA_AUDIO_CHANNEL_FL;
> > + v->info.position[1] = SPA_AUDIO_CHANNEL_FR;
> > + v->info.position[2] = SPA_AUDIO_CHANNEL_FC;
> > + v->info.position[3] = SPA_AUDIO_CHANNEL_LFE;
> > + v->info.position[4] = SPA_AUDIO_CHANNEL_RL;
> > + v->info.position[5] = SPA_AUDIO_CHANNEL_RR;
> > + v->info.position[6] = SPA_AUDIO_CHANNEL_SL;
> > + v->info.position[7] = SPA_AUDIO_CHANNEL_SR;
> > + break;
> > + case 6:
> > + v->info.position[0] = SPA_AUDIO_CHANNEL_FL;
> > + v->info.position[1] = SPA_AUDIO_CHANNEL_FR;
> > + v->info.position[2] = SPA_AUDIO_CHANNEL_FC;
> > + v->info.position[3] = SPA_AUDIO_CHANNEL_LFE;
> > + v->info.position[4] = SPA_AUDIO_CHANNEL_RL;
> > + v->info.position[5] = SPA_AUDIO_CHANNEL_RR;
> > + break;
> > + case 5:
> > + v->info.position[0] = SPA_AUDIO_CHANNEL_FL;
> > + v->info.position[1] = SPA_AUDIO_CHANNEL_FR;
> > + v->info.position[2] = SPA_AUDIO_CHANNEL_FC;
> > + v->info.position[3] = SPA_AUDIO_CHANNEL_LFE;
> > + v->info.position[4] = SPA_AUDIO_CHANNEL_RC;
> > + break;
> > + case 4:
> > + v->info.position[0] = SPA_AUDIO_CHANNEL_FL;
> > + v->info.position[1] = SPA_AUDIO_CHANNEL_FR;
> > + v->info.position[2] = SPA_AUDIO_CHANNEL_FC;
> > + v->info.position[3] = SPA_AUDIO_CHANNEL_RC;
> > + break;
> > + case 3:
> > + v->info.position[0] = SPA_AUDIO_CHANNEL_FL;
> > + v->info.position[1] = SPA_AUDIO_CHANNEL_FR;
> > + v->info.position[2] = SPA_AUDIO_CHANNEL_LFE;
> > + break;
> > + case 2:
> > + v->info.position[0] = SPA_AUDIO_CHANNEL_FL;
> > + v->info.position[1] = SPA_AUDIO_CHANNEL_FR;
> > + break;
> > + case 1:
> > + v->info.position[0] = SPA_AUDIO_CHANNEL_MONO;
> > + break;
> > + default:
> > + for (size_t i = 0; i < v->info.channels; i++) {
> > + v->info.position[i] = SPA_AUDIO_CHANNEL_UNKNOWN;
> > + }
> > + break;
> > + }
> > +
> > + /* create a new unconnected pwstream */
> > + r = create_stream(c, v, name);
> > + if (r < 0) {
> > + goto error;
> > + }
> > +
> > + pw_thread_loop_unlock(c->thread_loop);
> > + return r;
> > +
> > +error:
> > + AUD_log(AUDIO_CAP, "Failed to create stream.");
> > + pw_thread_loop_unlock(c->thread_loop);
> > + pw_destroy(c);
>
> You are freeing c (and other data) on stream creation error. This will
> later crash during fini(), or other events, I guess.
>
> There is a single "goto error", you could move it there.
>
>
> > + return -1;
> > +}
> > +
> > +static int
> > +qpw_init_out(HWVoiceOut *hw, struct audsettings *as, void *drv_opaque)
> > +{
> > + PWVoiceOut *pw = (PWVoiceOut *) hw;
> > + PWVoice *v = &pw->v;
> > + struct audsettings obt_as = *as;
> > + pwaudio *c = v->g = drv_opaque;
> > + AudiodevPipewireOptions *popts = &c->dev->u.pipewire;
> > + AudiodevPipewirePerDirectionOptions *ppdo = popts->out;
> > + int r;
> > + v->enabled = false;
> > +
> > + v->mode = MODE_SINK;
> > +
> > + pw_thread_loop_lock(c->thread_loop);
> > +
> > + v->info.format = audfmt_to_pw(as->fmt, as->endianness);
> > + v->info.channels = as->nchannels;
> > + v->info.rate = as->freq;
> > +
> > + obt_as.fmt =
> > + pw_to_audfmt(v->info.format, &obt_as.endianness,
> &v->frame_size);
> > + v->frame_size *= as->nchannels;
> > +
> > + /* call the function that creates a new stream for playback */
> > + r = qpw_stream_new(c, v, ppdo->stream_name ? : c->dev->id);
> > + if (r < 0) {
> > + pw_log_error("qpw_stream_new for playback failed\n ");
>
> Use error_report() instead
>
> > + 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;
>
> There is a single "goto fail", you could move that here.
>
> > +}
> > +
> > +static int
> > +qpw_init_in(HWVoiceIn *hw, struct audsettings *as, void *drv_opaque)
> > +{
> > + PWVoiceIn *pw = (PWVoiceIn *) hw;
> > + PWVoice *v = &pw->v;
> > + struct audsettings obt_as = *as;
> > + pwaudio *c = v->g = drv_opaque;
> > + AudiodevPipewireOptions *popts = &c->dev->u.pipewire;
> > + AudiodevPipewirePerDirectionOptions *ppdo = popts->in;
> > + int r;
> > + v->enabled = false;
> > +
> > + v->mode = MODE_SOURCE;
> > + pw_thread_loop_lock(c->thread_loop);
> > +
> > + v->info.format = audfmt_to_pw(as->fmt, as->endianness);
> > + v->info.channels = as->nchannels;
> > + v->info.rate = as->freq;
> > +
> > + obt_as.fmt =
> > + pw_to_audfmt(v->info.format, &obt_as.endianness,
> &v->frame_size);
> > + v->frame_size *= as->nchannels;
> > +
> > + /* call the function that creates a new stream for recording */
> > + r = qpw_stream_new(c, v, ppdo->stream_name ? : c->dev->id);
> > + if (r < 0) {
> > + pw_log_error("qpw_stream_new for recording failed\n ");
>
> Use error_report() instead
>
> > + 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;
>
> There is a single "goto fail", you could move that here.
>
> > +}
> > +
> > +static void
> > +qpw_fini_out(HWVoiceOut *hw)
> > +{
> > + PWVoiceOut *pw = (PWVoiceOut *) hw;
> > + PWVoice *v = &pw->v;
> > +
> > + if (v->stream) {
> > + pwaudio *c = v->g;
> > + pw_thread_loop_lock(c->thread_loop);
> > + pw_stream_destroy(v->stream);
> > + v->stream = NULL;
> > + pw_thread_loop_unlock(c->thread_loop);
> > + }
> > +}
> > +
> > +static void
> > +qpw_fini_in(HWVoiceIn *hw)
> > +{
> > + PWVoiceIn *pw = (PWVoiceIn *) hw;
> > + PWVoice *v = &pw->v;
> > +
> > + if (v->stream) {
> > + pwaudio *c = v->g;
> > + pw_thread_loop_lock(c->thread_loop);
> > + pw_stream_destroy(v->stream);
> > + v->stream = NULL;
> > + pw_thread_loop_unlock(c->thread_loop);
> > + }
> > +}
> > +
> > +static void
> > +qpw_enable_out(HWVoiceOut *hw, bool enable)
> > +{
> > + PWVoiceOut *po = (PWVoiceOut *) hw;
> > + PWVoice *v = &po->v;
> > + v->enabled = enable;
> > +}
> > +
> > +static void
> > +qpw_enable_in(HWVoiceIn *hw, bool enable)
> > +{
> > + PWVoiceIn *pi = (PWVoiceIn *) hw;
> > + PWVoice *v = &pi->v;
> > + v->enabled = enable;
> > +}
> > +
> > +static void
> > +on_core_error(void *data, uint32_t id, int seq, int res, const char
> *message)
> > +{
> > + pwaudio *pw = data;
> > +
> > + pw_log_warn("error id:%u seq:%d res:%d (%s): %s",
> > + id, seq, res, spa_strerror(res), message);
>
> Use error_report() instead
>
> > +
> > + 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);
>
> What are those thread_loop_signal() for? Maybe leave a comment?
>
> > + }
> > +}
> > +
> > +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;
> > + pw_log_debug("Initialize PW context\n");
>
> Can you replace the logging with traces? That's what QEMU uses.
>
> > + assert(dev->driver == AUDIODEV_DRIVER_PIPEWIRE);
> > + popts = &dev->u.pipewire;
> > +
> > + if (!popts->has_latency) {
> > + popts->has_latency = true;
> > + popts->latency = 15000;
> > + }
> > +
> > + pw->dev = dev;
> > + pw->thread_loop = pw_thread_loop_new("Pipewire thread loop", NULL);
> > + if (pw->thread_loop == NULL) {
> > + goto fail;
> > + }
>
> Either we handle allocation failures everywhere, or we don't. Unless
> the documentation is explicit about handling allocation/initialization
> errors, I would say we don't have to, in init/fini().
>
> > + pw->context =
> > + pw_context_new(pw_thread_loop_get_loop(pw->thread_loop), NULL,
> 0);
> > +
>
> Why not handle errors here?
>
> > + if (pw_thread_loop_start(pw->thread_loop) < 0) {
> > + goto fail;
> > + }
> > +
> > + pw_thread_loop_lock(pw->thread_loop);
> > +
> > + pw->core = pw_context_connect(pw->context, NULL, 0);
> > + if (pw->core == NULL) {
> > + goto fail;
> > + }
> > +
> > + pw_core_add_listener(pw->core, &pw->core_listener, &core_events,
> pw);
> > +
> > + pw_thread_loop_unlock(pw->thread_loop);
> > +
> > + return pw;
> > +
> > +fail:
> > + AUD_log(AUDIO_CAP, "Failed to initialize PW context");
> > + pw_thread_loop_unlock(pw->thread_loop);
>
> Bad after first "goto fail"
>
> > + pw_context_destroy(pw->context);
> > + pw_thread_loop_destroy(pw->thread_loop);
> > + g_free(pw);
> > + return NULL;
> > +}
> > +
> > +static void
> > +qpw_audio_fini(void *opaque)
> > +{
> > + pwaudio *pw = opaque;
> > +
> > + pw_thread_loop_stop(pw->thread_loop);
> > +
> > + if (pw->core) {
> > + spa_hook_remove(&pw->core_listener);
> > + spa_zero(pw->core_listener);
> > + pw_core_disconnect(pw->core);
> > + }
> > +
> > + if (pw->context) {
> > + pw_context_destroy(pw->context);
> > + }
> > + pw_thread_loop_destroy(pw->thread_loop);
> > +
> > + g_free(pw);
> > +}
> > +
> > +static struct audio_pcm_ops qpw_pcm_ops = {
> > + .init_out = qpw_init_out,
> > + .fini_out = qpw_fini_out,
> > + .write = qpw_write,
> > + .buffer_get_free = audio_generic_buffer_get_free,
> > + .run_buffer_out = audio_generic_run_buffer_out,
> > + .enable_out = qpw_enable_out,
> > +
> > + .init_in = qpw_init_in,
> > + .fini_in = qpw_fini_in,
> > + .read = qpw_read,
> > + .run_buffer_in = audio_generic_run_buffer_in,
> > + .enable_in = qpw_enable_in
> > +};
> > +
> > +static struct audio_driver pw_audio_driver = {
> > + .name = "pipewire",
> > + .descr = "http://www.pipewire.org/",
> > + .init = qpw_audio_init,
> > + .fini = qpw_audio_fini,
> > + .pcm_ops = &qpw_pcm_ops,
> > + .can_be_default = 1,
> > + .max_voices_out = INT_MAX,
> > + .max_voices_in = INT_MAX,
> > + .voice_size_out = sizeof(PWVoiceOut),
> > + .voice_size_in = sizeof(PWVoiceIn),
> > +};
> > +
> > +static void
> > +register_audio_pw(void)
> > +{
> > + audio_driver_register(&pw_audio_driver);
> > +}
> > +
> > +type_init(register_audio_pw);
> > diff --git a/meson.build b/meson.build
> > index 6cb2b1a42f..2aa0b397ea 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(),
> > }
> > @@ -3978,6 +3985,7 @@ 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 6b0900205e..6c5f260a6a 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)
> > +#
> > +#
> > +# 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 beeb4475ba..2fd88ccf15 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 5d969a94c0..236a714087 100644
> > --- a/scripts/meson-buildoptions.sh
> > +++ b/scripts/meson-buildoptions.sh
> > @@ -1,7 +1,8 @@
> > # This file is generated by meson-buildoptions.py, do not edit!
> > meson_options_help() {
> > - printf "%s\n" ' --audio-drv-list=CHOICES Set audio driver list
> [default] (choices: alsa/co'
> > - printf "%s\n" '
> reaudio/default/dsound/jack/oss/pa/sdl/sndio)'
> > + printf "%s\n" ' --audio-drv-list=CHOICES Set audio driver list
> [default] (choices: al'
> > + printf "%s\n" '
> sa/coreaudio/default/dsound/jack/oss/pa/'
> > + printf "%s\n" ' pipewire/sdl/sndio)'
> > printf "%s\n" ' --block-drv-ro-whitelist=VALUE'
> > printf "%s\n" ' set block driver read-only
> whitelist (by default'
> > printf "%s\n" ' affects only QEMU, not
> tools like qemu-img)'
> > @@ -135,6 +136,7 @@ meson_options_help() {
> > printf "%s\n" ' oss OSS sound support'
> > printf "%s\n" ' pa PulseAudio sound support'
> > printf "%s\n" ' parallels parallels image format support'
> > + printf "%s\n" ' pipewire Pipewire sound support'
> > printf "%s\n" ' png PNG support with libpng'
> > printf "%s\n" ' pvrdma Enable PVRDMA support'
> > printf "%s\n" ' qcow1 qcow1 image format support'
> > @@ -369,6 +371,8 @@ _meson_option_parse() {
> > --disable-pa) printf "%s" -Dpa=disabled ;;
> > --enable-parallels) printf "%s" -Dparallels=enabled ;;
> > --disable-parallels) printf "%s" -Dparallels=disabled ;;
> > + --enable-pipewire) printf "%s" -Dpipewire=enabled ;;
> > + --disable-pipewire) printf "%s" -Dpipewire=disabled ;;
> > --with-pkgversion=*) quote_sh "-Dpkgversion=$2" ;;
> > --enable-png) printf "%s" -Dpng=enabled ;;
> > --disable-png) printf "%s" -Dpng=disabled ;;
> > --
> > 2.39.1
> >
> >
>
>
> --
> Marc-André Lureau
>
>
Hi On Fri, Mar 3, 2023 at 8:06 PM Dorinda Bassey >> What are those thread_loop_signal() for? Maybe leave a comment? > > the explanation of the function is in the reference header file. > Yes, I read the reference documentation before asking: "Signal all threads waiting with pw_thread_loop_wait." (https://docs.pipewire.org/group__pw__thread__loop.html#gaf9bc8dd348d05b095139f5a55ac5a4b0) Unfortunately, you are not calling pw_thread_loop_wait yourself, so that doesn't help me what this is supposed to do. When signaling things and expecting a certain state and side-effect from a different thread or context, it's nice to document it. I guess this will break the thread loop? What happens next? thanks -- Marc-André Lureau
Hi Unfortunately, you are not calling pw_thread_loop_wait yourself, so > that doesn't help me what this is supposed to do. When signaling > things and expecting a certain state and side-effect from a different > thread or context, it's nice to document it. > > I guess this will break the thread loop? What happens next? > In this case When the on_core_done event is received for an object with id `PW_ID_CORE` this function would call the thread_loop_signal(loop, FALSE) to stop and exit the thread loop. I would add a comment for this, to be clear. Thanks, Dorinda. On Tue, Mar 7, 2023 at 2:49 PM Marc-André Lureau <marcandre.lureau@gmail.com> wrote: > Hi > > On Fri, Mar 3, 2023 at 8:06 PM Dorinda Bassey > >> What are those thread_loop_signal() for? Maybe leave a comment? > > > > the explanation of the function is in the reference header file. > > > > Yes, I read the reference documentation before asking: "Signal all > threads waiting with pw_thread_loop_wait." > ( > https://docs.pipewire.org/group__pw__thread__loop.html#gaf9bc8dd348d05b095139f5a55ac5a4b0 > ) > > Unfortunately, you are not calling pw_thread_loop_wait yourself, so > that doesn't help me what this is supposed to do. When signaling > things and expecting a certain state and side-effect from a different > thread or context, it's nice to document it. > > I guess this will break the thread loop? What happens next? > > thanks > > -- > Marc-André Lureau > >
© 2016 - 2024 Red Hat, Inc.