| Message ID | 20230412135730.51012-1-dbassey@redhat.com (mailing list archive) |
|---|---|
| State | New, archived |
| Headers | show |
| Series | [v11] audio/pwaudio.c: Add Pipewire audio backend for QEMU | expand |
Hi Volker, It seems that for some unknown reason using audio_pcm_info_clear_buf in playback_process causes segmentation fault. Hence I moved the handling of buffer underruns from the playback process to the qpw_write process because that is the underlying cause of buffer underrun. Regards, Dorinda. On Wed, Apr 12, 2023 at 3:57 PM Dorinda Bassey <dbassey@redhat.com> wrote: > This commit adds a new audiodev backend to allow QEMU to use Pipewire as > both an audio sink and source. This backend is available on most systems > > Add Pipewire entry points for QEMU Pipewire audio backend > Add wrappers for QEMU Pipewire audio backend in qpw_pcm_ops() > qpw_write function returns the current state of the stream to pwaudio > and Writes some data to the server for playback streams using pipewire > spa_ringbuffer implementation. > qpw_read function returns the current state of the stream to pwaudio and > reads some data from the server for capture streams using pipewire > spa_ringbuffer implementation. These functions qpw_write and qpw_read > are called during playback and capture. > Added some functions that convert pw audio formats to QEMU audio format > and vice versa which would be needed in the pipewire audio sink and > source functions qpw_init_in() & qpw_init_out(). > These methods that implement playback and recording will create streams > for playback and capture that will start processing and will result in > the on_process callbacks to be called. > Built a connection to the Pipewire sound system server in the > qpw_audio_init() method. > > Signed-off-by: Dorinda Bassey <dbassey@redhat.com> > --- > v11: > handle buffer underruns in qpw_write > use local variable > change param name frame_size > fix format specifier > change trace value to trace quantum > > audio/audio.c | 3 + > audio/audio_template.h | 4 + > audio/meson.build | 1 + > audio/pwaudio.c | 913 ++++++++++++++++++++++++++++++++++ > audio/trace-events | 8 + > meson.build | 8 + > meson_options.txt | 4 +- > qapi/audio.json | 44 ++ > qemu-options.hx | 21 + > scripts/meson-buildoptions.sh | 8 +- > 10 files changed, 1011 insertions(+), 3 deletions(-) > create mode 100644 audio/pwaudio.c > > diff --git a/audio/audio.c b/audio/audio.c > index 70b096713c..90c7c49d11 100644 > --- a/audio/audio.c > +++ b/audio/audio.c > @@ -2061,6 +2061,9 @@ void audio_create_pdos(Audiodev *dev) > #ifdef CONFIG_AUDIO_PA > CASE(PA, pa, Pa); > #endif > +#ifdef CONFIG_AUDIO_PIPEWIRE > + CASE(PIPEWIRE, pipewire, Pipewire); > +#endif > #ifdef CONFIG_AUDIO_SDL > CASE(SDL, sdl, Sdl); > #endif > diff --git a/audio/audio_template.h b/audio/audio_template.h > index e42326c20d..dc0c74aa74 100644 > --- a/audio/audio_template.h > +++ b/audio/audio_template.h > @@ -362,6 +362,10 @@ AudiodevPerDirectionOptions *glue(audio_get_pdo_, > TYPE)(Audiodev *dev) > case AUDIODEV_DRIVER_PA: > return qapi_AudiodevPaPerDirectionOptions_base(dev->u.pa.TYPE); > #endif > +#ifdef CONFIG_AUDIO_PIPEWIRE > + case AUDIODEV_DRIVER_PIPEWIRE: > + return > qapi_AudiodevPipewirePerDirectionOptions_base(dev->u.pipewire.TYPE); > +#endif > #ifdef CONFIG_AUDIO_SDL > case AUDIODEV_DRIVER_SDL: > return qapi_AudiodevSdlPerDirectionOptions_base(dev->u.sdl.TYPE); > diff --git a/audio/meson.build b/audio/meson.build > index 0722224ba9..65a49c1a10 100644 > --- a/audio/meson.build > +++ b/audio/meson.build > @@ -19,6 +19,7 @@ foreach m : [ > ['sdl', sdl, files('sdlaudio.c')], > ['jack', jack, files('jackaudio.c')], > ['sndio', sndio, files('sndioaudio.c')], > + ['pipewire', pipewire, files('pwaudio.c')], > ['spice', spice, files('spiceaudio.c')] > ] > if m[1].found() > diff --git a/audio/pwaudio.c b/audio/pwaudio.c > new file mode 100644 > index 0000000000..adf1a538c0 > --- /dev/null > +++ b/audio/pwaudio.c > @@ -0,0 +1,913 @@ > +/* > + * QEMU Pipewire audio driver > + * > + * Copyright (c) 2023 Red Hat Inc. > + * > + * Author: Dorinda Bassey <dbassey@redhat.com> > + * > + * SPDX-License-Identifier: GPL-2.0-or-later > + */ > + > +#include "qemu/osdep.h" > +#include "qemu/module.h" > +#include "audio.h" > +#include <errno.h> > +#include "qemu/error-report.h" > +#include <spa/param/audio/format-utils.h> > +#include <spa/utils/ringbuffer.h> > +#include <spa/utils/result.h> > +#include <spa/param/props.h> > + > +#include <pipewire/pipewire.h> > +#include "trace.h" > + > +#define AUDIO_CAP "pipewire" > +#define RINGBUFFER_SIZE (1u << 22) > +#define RINGBUFFER_MASK (RINGBUFFER_SIZE - 1) > + > +#include "audio_int.h" > + > +typedef struct pwvolume { > + uint32_t channels; > + float values[SPA_AUDIO_MAX_CHANNELS]; > +} pwvolume; > + > +typedef struct pwaudio { > + Audiodev *dev; > + struct pw_thread_loop *thread_loop; > + struct pw_context *context; > + > + struct pw_core *core; > + struct spa_hook core_listener; > + int last_seq, pending_seq, error; > +} pwaudio; > + > +typedef struct PWVoice { > + pwaudio *g; > + struct pw_stream *stream; > + struct spa_hook stream_listener; > + struct spa_audio_info_raw info; > + uint32_t highwater_mark; > + uint32_t sample_size, req; > + struct spa_ringbuffer ring; > + uint8_t buffer[RINGBUFFER_SIZE]; > + > + pwvolume volume; > + bool muted; > +} PWVoice; > + > +typedef struct PWVoiceOut { > + HWVoiceOut hw; > + PWVoice v; > +} PWVoiceOut; > + > +typedef struct PWVoiceIn { > + HWVoiceIn hw; > + PWVoice v; > +} PWVoiceIn; > + > +static void > +stream_destroy(void *data) > +{ > + PWVoice *v = (PWVoice *) data; > + spa_hook_remove(&v->stream_listener); > + v->stream = NULL; > +} > + > +/* output data processing function to read stuffs from the buffer */ > +static void > +playback_on_process(void *data) > +{ > + PWVoice *v = (PWVoice *) data; > + void *p; > + struct pw_buffer *b; > + struct spa_buffer *buf; > + uint32_t req, index, n_bytes; > + int32_t avail; > + > + assert(v->stream); > + > + /* obtain a buffer to read from */ > + b = pw_stream_dequeue_buffer(v->stream); > + if (b == NULL) { > + error_report("out of buffers: %s", strerror(errno)); > + return; > + } > + > + buf = b->buffer; > + p = buf->datas[0].data; > + if (p == NULL) { > + return; > + } > + /* calculate the total no of bytes to read data from buffer */ > + req = b->requested * v->sample_size; > + if (req == 0) { > + req = v->req; > + } > + n_bytes = SPA_MIN(req, buf->datas[0].maxsize); > + > + /* get no of available bytes to read data from buffer */ > + > + avail = spa_ringbuffer_get_read_index(&v->ring, &index); > + > + if (avail <= 0) { > + /* underrun, can't really happen but if it does we */ > + /* do nothing and wait for more data */ > + error_report("%p: underrun read:%u avail:%d", p, index, avail); > + } 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->sample_size; > + buf->datas[0].chunk->size = n_bytes; > + > + /* queue the buffer for playback */ > + pw_stream_queue_buffer(v->stream, b); > +} > + > +/* output data processing function to generate stuffs in the buffer */ > +static void > +capture_on_process(void *data) > +{ > + PWVoice *v = (PWVoice *) data; > + void *p; > + struct pw_buffer *b; > + struct spa_buffer *buf; > + int32_t filled; > + uint32_t index, offs, n_bytes; > + > + assert(v->stream); > + > + /* obtain a buffer */ > + b = pw_stream_dequeue_buffer(v->stream); > + if (b == NULL) { > + error_report("out of buffers: %s", strerror(errno)); > + return; > + } > + > + /* Write data into buffer */ > + buf = b->buffer; > + p = buf->datas[0].data; > + if (p == NULL) { > + return; > + } > + offs = SPA_MIN(buf->datas[0].chunk->offset, buf->datas[0].maxsize); > + n_bytes = SPA_MIN(buf->datas[0].chunk->size, buf->datas[0].maxsize - > offs); > + > + filled = spa_ringbuffer_get_write_index(&v->ring, &index); > + > + > + if (filled < 0) { > + error_report("%p: underrun write:%u filled:%d", p, index, filled); > + } else { > + if ((uint32_t) filled + n_bytes > RINGBUFFER_SIZE) { > + error_report("%p: overrun write:%u filled:%d + size:%u > > max:%u", > + p, index, filled, n_bytes, RINGBUFFER_SIZE); > + } > + } > + spa_ringbuffer_write_data(&v->ring, > + v->buffer, RINGBUFFER_SIZE, > + index & RINGBUFFER_MASK, > + SPA_PTROFF(p, offs, void), n_bytes); > + index += n_bytes; > + spa_ringbuffer_write_update(&v->ring, index); > + > + /* queue the buffer for playback */ > + pw_stream_queue_buffer(v->stream, b); > +} > + > +static void > +on_stream_state_changed(void *data, enum pw_stream_state old, > + enum pw_stream_state state, const char *error) > +{ > + PWVoice *v = (PWVoice *) data; > + > + trace_pw_state_changed(pw_stream_get_node_id(v->stream), > + pw_stream_state_as_string(state)); > + > + switch (state) { > + case PW_STREAM_STATE_ERROR: > + case PW_STREAM_STATE_UNCONNECTED: > + break; > + case PW_STREAM_STATE_PAUSED: > + case PW_STREAM_STATE_CONNECTING: > + case PW_STREAM_STATE_STREAMING: > + break; > + } > +} > + > +static const struct pw_stream_events capture_stream_events = { > + PW_VERSION_STREAM_EVENTS, > + .destroy = stream_destroy, > + .state_changed = on_stream_state_changed, > + .process = capture_on_process > +}; > + > +static const struct pw_stream_events playback_stream_events = { > + PW_VERSION_STREAM_EVENTS, > + .destroy = stream_destroy, > + .state_changed = on_stream_state_changed, > + .process = playback_on_process > +}; > + > +static size_t > +qpw_read(HWVoiceIn *hw, void *data, size_t len) > +{ > + PWVoiceIn *pw = (PWVoiceIn *) hw; > + PWVoice *v = &pw->v; > + pwaudio *c = v->g; > + const char *error = NULL; > + size_t l; > + int32_t avail; > + uint32_t index; > + > + pw_thread_loop_lock(c->thread_loop); > + if (pw_stream_get_state(v->stream, &error) != > PW_STREAM_STATE_STREAMING) { > + /* wait for stream to become ready */ > + l = 0; > + goto done_unlock; > + } > + /* get no of available bytes to read data from buffer */ > + avail = spa_ringbuffer_get_read_index(&v->ring, &index); > + > + trace_pw_read(avail, index, len); > + > + if (avail < (int32_t) len) { > + len = avail; > + } > + > + spa_ringbuffer_read_data(&v->ring, > + v->buffer, RINGBUFFER_SIZE, > + index & RINGBUFFER_MASK, data, len); > + index += len; > + spa_ringbuffer_read_update(&v->ring, index); > + l = len; > + > +done_unlock: > + pw_thread_loop_unlock(c->thread_loop); > + return l; > +} > + > +static size_t qpw_buffer_get_free(HWVoiceOut *hw) > +{ > + PWVoiceOut *pw = (PWVoiceOut *)hw; > + PWVoice *v = &pw->v; > + pwaudio *c = v->g; > + const char *error = NULL; > + int32_t filled, avail; > + uint32_t index; > + > + pw_thread_loop_lock(c->thread_loop); > + if (pw_stream_get_state(v->stream, &error) != > PW_STREAM_STATE_STREAMING) { > + /* wait for stream to become ready */ > + avail = 0; > + goto done_unlock; > + } > + > + filled = spa_ringbuffer_get_write_index(&v->ring, &index); > + avail = v->highwater_mark - filled; > + > +done_unlock: > + pw_thread_loop_unlock(c->thread_loop); > + return avail; > +} > + > +static size_t > +qpw_write(HWVoiceOut *hw, void *data, size_t len) > +{ > + PWVoiceOut *pw = (PWVoiceOut *) hw; > + PWVoice *v = &pw->v; > + pwaudio *c = v->g; > + const char *error = NULL; > + int32_t filled, avail; > + uint32_t index; > + > + pw_thread_loop_lock(c->thread_loop); > + if (pw_stream_get_state(v->stream, &error) != > PW_STREAM_STATE_STREAMING) { > + /* wait for stream to become ready */ > + len = 0; > + goto done_unlock; > + } > + filled = spa_ringbuffer_get_write_index(&v->ring, &index); > + avail = v->highwater_mark - filled; > + > + trace_pw_write(filled, avail, index, len); > + > + if (len > avail) { > + len = avail; > + } > + > + if (filled < 0) { > + audio_pcm_info_clear_buf(&hw->info, data, len / > hw->info.bytes_per_frame); > + error_report("%p: underrun write:%u filled:%d", pw, index, > filled); > + } else { > + if ((uint32_t) filled + len > RINGBUFFER_SIZE) { > + error_report("%p: overrun write:%u filled:%d + size:%zu > > max:%u", > + pw, index, filled, len, RINGBUFFER_SIZE); > + } > + } > + > + spa_ringbuffer_write_data(&v->ring, > + v->buffer, RINGBUFFER_SIZE, > + index & RINGBUFFER_MASK, data, len); > + index += len; > + spa_ringbuffer_write_update(&v->ring, index); > + > +done_unlock: > + pw_thread_loop_unlock(c->thread_loop); > + return len; > +} > + > +static int > +audfmt_to_pw(AudioFormat fmt, int endianness) > +{ > + int format; > + > + switch (fmt) { > + case AUDIO_FORMAT_S8: > + format = SPA_AUDIO_FORMAT_S8; > + break; > + case AUDIO_FORMAT_U8: > + format = SPA_AUDIO_FORMAT_U8; > + break; > + case AUDIO_FORMAT_S16: > + format = endianness ? SPA_AUDIO_FORMAT_S16_BE : > SPA_AUDIO_FORMAT_S16_LE; > + break; > + case AUDIO_FORMAT_U16: > + format = endianness ? SPA_AUDIO_FORMAT_U16_BE : > SPA_AUDIO_FORMAT_U16_LE; > + break; > + case AUDIO_FORMAT_S32: > + format = endianness ? SPA_AUDIO_FORMAT_S32_BE : > SPA_AUDIO_FORMAT_S32_LE; > + break; > + case AUDIO_FORMAT_U32: > + format = endianness ? SPA_AUDIO_FORMAT_U32_BE : > SPA_AUDIO_FORMAT_U32_LE; > + break; > + case AUDIO_FORMAT_F32: > + format = endianness ? SPA_AUDIO_FORMAT_F32_BE : > SPA_AUDIO_FORMAT_F32_LE; > + break; > + default: > + dolog("Internal logic error: Bad audio format %d\n", fmt); > + format = SPA_AUDIO_FORMAT_U8; > + break; > + } > + return format; > +} > + > +static AudioFormat > +pw_to_audfmt(enum spa_audio_format fmt, int *endianness, > + uint32_t *sample_size) > +{ > + switch (fmt) { > + case SPA_AUDIO_FORMAT_S8: > + *sample_size = 1; > + return AUDIO_FORMAT_S8; > + case SPA_AUDIO_FORMAT_U8: > + *sample_size = 1; > + return AUDIO_FORMAT_U8; > + case SPA_AUDIO_FORMAT_S16_BE: > + *sample_size = 2; > + *endianness = 1; > + return AUDIO_FORMAT_S16; > + case SPA_AUDIO_FORMAT_S16_LE: > + *sample_size = 2; > + *endianness = 0; > + return AUDIO_FORMAT_S16; > + case SPA_AUDIO_FORMAT_U16_BE: > + *sample_size = 2; > + *endianness = 1; > + return AUDIO_FORMAT_U16; > + case SPA_AUDIO_FORMAT_U16_LE: > + *sample_size = 2; > + *endianness = 0; > + return AUDIO_FORMAT_U16; > + case SPA_AUDIO_FORMAT_S32_BE: > + *sample_size = 4; > + *endianness = 1; > + return AUDIO_FORMAT_S32; > + case SPA_AUDIO_FORMAT_S32_LE: > + *sample_size = 4; > + *endianness = 0; > + return AUDIO_FORMAT_S32; > + case SPA_AUDIO_FORMAT_U32_BE: > + *sample_size = 4; > + *endianness = 1; > + return AUDIO_FORMAT_U32; > + case SPA_AUDIO_FORMAT_U32_LE: > + *sample_size = 4; > + *endianness = 0; > + return AUDIO_FORMAT_U32; > + case SPA_AUDIO_FORMAT_F32_BE: > + *sample_size = 4; > + *endianness = 1; > + return AUDIO_FORMAT_F32; > + case SPA_AUDIO_FORMAT_F32_LE: > + *sample_size = 4; > + *endianness = 0; > + return AUDIO_FORMAT_F32; > + default: > + *sample_size = 1; > + dolog("Internal logic error: Bad spa_audio_format %d\n", fmt); > + return AUDIO_FORMAT_U8; > + } > +} > + > +static int > +create_stream(pwaudio *c, PWVoice *v, const char *stream_name, > + const char *name, enum spa_direction dir) > +{ > + int res; > + uint32_t n_params; > + const struct spa_pod *params[2]; > + uint8_t buffer[1024]; > + struct spa_pod_builder b; > + uint64_t buf_samples; > + struct pw_properties *props; > + > + props = pw_properties_new(NULL, NULL); > + > + /* 75% of the timer period for faster updates */ > + buf_samples = (uint64_t)v->g->dev->timer_period * v->info.rate > + * 3 / 4 / 1000000; > + trace_pw_timer(v->g->dev->timer_period); > + pw_properties_setf(props, PW_KEY_NODE_LATENCY, "%" PRIu64 "/%u", > + buf_samples, v->info.rate); > + > + if (name) { > + pw_properties_set(props, PW_KEY_TARGET_OBJECT, name); > + } > + v->stream = pw_stream_new(c->core, stream_name, props); > + > + if (v->stream == NULL) { > + return -1; > + } > + > + if (dir == SPA_DIRECTION_INPUT) { > + pw_stream_add_listener(v->stream, > + &v->stream_listener, &capture_stream_events, > v); > + } else { > + pw_stream_add_listener(v->stream, > + &v->stream_listener, &playback_stream_events, > v); > + } > + > + n_params = 0; > + spa_pod_builder_init(&b, buffer, sizeof(buffer)); > + params[n_params++] = spa_format_audio_raw_build(&b, > + SPA_PARAM_EnumFormat, > + &v->info); > + > + /* connect the stream to a sink or source */ > + res = pw_stream_connect(v->stream, > + dir == > + SPA_DIRECTION_INPUT ? PW_DIRECTION_INPUT : > + PW_DIRECTION_OUTPUT, PW_ID_ANY, > + PW_STREAM_FLAG_AUTOCONNECT | > + PW_STREAM_FLAG_INACTIVE | > + PW_STREAM_FLAG_MAP_BUFFERS | > + PW_STREAM_FLAG_RT_PROCESS, params, n_params); > + if (res < 0) { > + pw_stream_destroy(v->stream); > + return -1; > + } > + > + return 0; > +} > + > +static int > +qpw_stream_new(pwaudio *c, PWVoice *v, const char *stream_name, > + const char *name, enum spa_direction dir) > +{ > + int r; > + > + switch (v->info.channels) { > + case 8: > + v->info.position[0] = SPA_AUDIO_CHANNEL_FL; > + v->info.position[1] = SPA_AUDIO_CHANNEL_FR; > + v->info.position[2] = SPA_AUDIO_CHANNEL_FC; > + v->info.position[3] = SPA_AUDIO_CHANNEL_LFE; > + v->info.position[4] = SPA_AUDIO_CHANNEL_RL; > + v->info.position[5] = SPA_AUDIO_CHANNEL_RR; > + v->info.position[6] = SPA_AUDIO_CHANNEL_SL; > + v->info.position[7] = SPA_AUDIO_CHANNEL_SR; > + break; > + case 6: > + v->info.position[0] = SPA_AUDIO_CHANNEL_FL; > + v->info.position[1] = SPA_AUDIO_CHANNEL_FR; > + v->info.position[2] = SPA_AUDIO_CHANNEL_FC; > + v->info.position[3] = SPA_AUDIO_CHANNEL_LFE; > + v->info.position[4] = SPA_AUDIO_CHANNEL_RL; > + v->info.position[5] = SPA_AUDIO_CHANNEL_RR; > + break; > + case 5: > + v->info.position[0] = SPA_AUDIO_CHANNEL_FL; > + v->info.position[1] = SPA_AUDIO_CHANNEL_FR; > + v->info.position[2] = SPA_AUDIO_CHANNEL_FC; > + v->info.position[3] = SPA_AUDIO_CHANNEL_LFE; > + v->info.position[4] = SPA_AUDIO_CHANNEL_RC; > + break; > + case 4: > + v->info.position[0] = SPA_AUDIO_CHANNEL_FL; > + v->info.position[1] = SPA_AUDIO_CHANNEL_FR; > + v->info.position[2] = SPA_AUDIO_CHANNEL_FC; > + v->info.position[3] = SPA_AUDIO_CHANNEL_RC; > + break; > + case 3: > + v->info.position[0] = SPA_AUDIO_CHANNEL_FL; > + v->info.position[1] = SPA_AUDIO_CHANNEL_FR; > + v->info.position[2] = SPA_AUDIO_CHANNEL_LFE; > + break; > + case 2: > + v->info.position[0] = SPA_AUDIO_CHANNEL_FL; > + v->info.position[1] = SPA_AUDIO_CHANNEL_FR; > + break; > + case 1: > + v->info.position[0] = SPA_AUDIO_CHANNEL_MONO; > + break; > + default: > + for (size_t i = 0; i < v->info.channels; i++) { > + v->info.position[i] = SPA_AUDIO_CHANNEL_UNKNOWN; > + } > + break; > + } > + > + /* create a new unconnected pwstream */ > + r = create_stream(c, v, stream_name, name, dir); > + if (r < 0) { > + AUD_log(AUDIO_CAP, "Failed to create stream."); > + return -1; > + } > + > + return r; > +} > + > +static int > +qpw_init_out(HWVoiceOut *hw, struct audsettings *as, void *drv_opaque) > +{ > + PWVoiceOut *pw = (PWVoiceOut *) hw; > + PWVoice *v = &pw->v; > + struct audsettings obt_as = *as; > + pwaudio *c = v->g = drv_opaque; > + AudiodevPipewireOptions *popts = &c->dev->u.pipewire; > + AudiodevPipewirePerDirectionOptions *ppdo = popts->out; > + int r; > + > + pw_thread_loop_lock(c->thread_loop); > + > + v->info.format = audfmt_to_pw(as->fmt, as->endianness); > + v->info.channels = as->nchannels; > + v->info.rate = as->freq; > + > + obt_as.fmt = > + pw_to_audfmt(v->info.format, &obt_as.endianness, &v->sample_size); > + v->sample_size *= as->nchannels; > + > + v->req = (uint64_t)c->dev->timer_period * v->info.rate > + * 1 / 2 / 1000000 * v->sample_size; > + > + /* call the function that creates a new stream for playback */ > + r = qpw_stream_new(c, v, ppdo->stream_name ? : c->dev->id, > + ppdo->name, SPA_DIRECTION_OUTPUT); > + if (r < 0) { > + error_report("qpw_stream_new for playback failed"); > + pw_thread_loop_unlock(c->thread_loop); > + return -1; > + } > + > + /* report the audio format we support */ > + audio_pcm_init_info(&hw->info, &obt_as); > + > + /* report the buffer size to qemu */ > + hw->samples = audio_buffer_frames( > + qapi_AudiodevPipewirePerDirectionOptions_base(ppdo), &obt_as, > 46440); > + v->highwater_mark = MIN(RINGBUFFER_SIZE, > + (ppdo->has_latency ? ppdo->latency : 46440) > + * (uint64_t)v->info.rate / 1000000 * > v->sample_size); > + > + pw_thread_loop_unlock(c->thread_loop); > + return 0; > +} > + > +static int > +qpw_init_in(HWVoiceIn *hw, struct audsettings *as, void *drv_opaque) > +{ > + PWVoiceIn *pw = (PWVoiceIn *) hw; > + PWVoice *v = &pw->v; > + struct audsettings obt_as = *as; > + pwaudio *c = v->g = drv_opaque; > + AudiodevPipewireOptions *popts = &c->dev->u.pipewire; > + AudiodevPipewirePerDirectionOptions *ppdo = popts->in; > + int r; > + > + pw_thread_loop_lock(c->thread_loop); > + > + v->info.format = audfmt_to_pw(as->fmt, as->endianness); > + v->info.channels = as->nchannels; > + v->info.rate = as->freq; > + > + obt_as.fmt = > + pw_to_audfmt(v->info.format, &obt_as.endianness, &v->sample_size); > + v->sample_size *= as->nchannels; > + > + /* call the function that creates a new stream for recording */ > + r = qpw_stream_new(c, v, ppdo->stream_name ? : c->dev->id, > + ppdo->name, SPA_DIRECTION_INPUT); > + if (r < 0) { > + error_report("qpw_stream_new for recording failed"); > + pw_thread_loop_unlock(c->thread_loop); > + return -1; > + } > + > + /* report the audio format we support */ > + audio_pcm_init_info(&hw->info, &obt_as); > + > + /* report the buffer size to qemu */ > + hw->samples = audio_buffer_frames( > + qapi_AudiodevPipewirePerDirectionOptions_base(ppdo), &obt_as, > 46440); > + > + pw_thread_loop_unlock(c->thread_loop); > + return 0; > +} > + > +static void > +qpw_fini_out(HWVoiceOut *hw) > +{ > + PWVoiceOut *pw = (PWVoiceOut *) hw; > + PWVoice *v = &pw->v; > + > + if (v->stream) { > + pwaudio *c = v->g; > + pw_thread_loop_lock(c->thread_loop); > + pw_stream_destroy(v->stream); > + v->stream = NULL; > + pw_thread_loop_unlock(c->thread_loop); > + } > +} > + > +static void > +qpw_fini_in(HWVoiceIn *hw) > +{ > + PWVoiceIn *pw = (PWVoiceIn *) hw; > + PWVoice *v = &pw->v; > + > + if (v->stream) { > + pwaudio *c = v->g; > + pw_thread_loop_lock(c->thread_loop); > + pw_stream_destroy(v->stream); > + v->stream = NULL; > + pw_thread_loop_unlock(c->thread_loop); > + } > +} > + > +static void > +qpw_enable_out(HWVoiceOut *hw, bool enable) > +{ > + PWVoiceOut *po = (PWVoiceOut *) hw; > + PWVoice *v = &po->v; > + pwaudio *c = v->g; > + pw_thread_loop_lock(c->thread_loop); > + pw_stream_set_active(v->stream, enable); > + pw_thread_loop_unlock(c->thread_loop); > +} > + > +static void > +qpw_enable_in(HWVoiceIn *hw, bool enable) > +{ > + PWVoiceIn *pi = (PWVoiceIn *) hw; > + PWVoice *v = &pi->v; > + pwaudio *c = v->g; > + pw_thread_loop_lock(c->thread_loop); > + pw_stream_set_active(v->stream, enable); > + pw_thread_loop_unlock(c->thread_loop); > +} > + > +static void > +qpw_volume_out(HWVoiceOut *hw, Volume *vol) > +{ > + PWVoiceOut *pw = (PWVoiceOut *) hw; > + PWVoice *v = &pw->v; > + pwaudio *c = v->g; > + int i, ret; > + > + pw_thread_loop_lock(c->thread_loop); > + v->volume.channels = vol->channels; > + > + for (i = 0; i < vol->channels; ++i) { > + v->volume.values[i] = (float)vol->vol[i] / 255; > + } > + > + ret = pw_stream_set_control(v->stream, > + SPA_PROP_channelVolumes, v->volume.channels, v->volume.values, 0); > + trace_pw_vol(ret == 0 ? "success" : "failed"); > + > + v->muted = vol->mute; > + float val = v->muted ? 1.f : 0.f; > + ret = pw_stream_set_control(v->stream, SPA_PROP_mute, 1, &val, 0); > + pw_thread_loop_unlock(c->thread_loop); > +} > + > +static void > +qpw_volume_in(HWVoiceIn *hw, Volume *vol) > +{ > + PWVoiceIn *pw = (PWVoiceIn *) hw; > + PWVoice *v = &pw->v; > + pwaudio *c = v->g; > + int i, ret; > + > + pw_thread_loop_lock(c->thread_loop); > + v->volume.channels = vol->channels; > + > + for (i = 0; i < vol->channels; ++i) { > + v->volume.values[i] = (float)vol->vol[i] / 255; > + } > + > + ret = pw_stream_set_control(v->stream, > + SPA_PROP_channelVolumes, v->volume.channels, v->volume.values, 0); > + trace_pw_vol(ret == 0 ? "success" : "failed"); > + > + v->muted = vol->mute; > + float val = v->muted ? 1.f : 0.f; > + ret = pw_stream_set_control(v->stream, SPA_PROP_mute, 1, &val, 0); > + pw_thread_loop_unlock(c->thread_loop); > +} > + > +static int wait_resync(pwaudio *pw) > +{ > + int res; > + pw->pending_seq = pw_core_sync(pw->core, PW_ID_CORE, pw->pending_seq); > + > + while (true) { > + pw_thread_loop_wait(pw->thread_loop); > + > + res = pw->error; > + if (res < 0) { > + pw->error = 0; > + return res; > + } > + if (pw->pending_seq == pw->last_seq) { > + break; > + } > + } > + return 0; > +} > +static void > +on_core_error(void *data, uint32_t id, int seq, int res, const char > *message) > +{ > + pwaudio *pw = data; > + > + error_report("error id:%u seq:%d res:%d (%s): %s", > + id, seq, res, spa_strerror(res), message); > + > + /* stop and exit the thread loop */ > + pw_thread_loop_signal(pw->thread_loop, FALSE); > +} > + > +static void > +on_core_done(void *data, uint32_t id, int seq) > +{ > + pwaudio *pw = data; > + assert(id == PW_ID_CORE); > + pw->last_seq = seq; > + if (pw->pending_seq == seq) { > + /* stop and exit the thread loop */ > + pw_thread_loop_signal(pw->thread_loop, FALSE); > + } > +} > + > +static const struct pw_core_events core_events = { > + PW_VERSION_CORE_EVENTS, > + .done = on_core_done, > + .error = on_core_error, > +}; > + > +static void * > +qpw_audio_init(Audiodev *dev) > +{ > + g_autofree pwaudio *pw = g_new0(pwaudio, 1); > + pw_init(NULL, NULL); > + > + trace_pw_audio_init(); > + assert(dev->driver == AUDIODEV_DRIVER_PIPEWIRE); > + > + pw->dev = dev; > + pw->thread_loop = pw_thread_loop_new("Pipewire thread loop", NULL); > + if (pw->thread_loop == NULL) { > + error_report("Could not create Pipewire loop"); > + goto fail; > + } > + > + pw->context = > + pw_context_new(pw_thread_loop_get_loop(pw->thread_loop), NULL, 0); > + if (pw->context == NULL) { > + error_report("Could not create Pipewire context"); > + goto fail; > + } > + > + if (pw_thread_loop_start(pw->thread_loop) < 0) { > + error_report("Could not start Pipewire loop"); > + goto fail; > + } > + > + pw_thread_loop_lock(pw->thread_loop); > + > + pw->core = pw_context_connect(pw->context, NULL, 0); > + if (pw->core == NULL) { > + pw_thread_loop_unlock(pw->thread_loop); > + goto fail; > + } > + > + if (pw_core_add_listener(pw->core, &pw->core_listener, > + &core_events, pw) < 0) { > + pw_thread_loop_unlock(pw->thread_loop); > + goto fail; > + } > + if (wait_resync(pw) < 0) { > + pw_thread_loop_unlock(pw->thread_loop); > + } > + > + pw_thread_loop_unlock(pw->thread_loop); > + > + return g_steal_pointer(&pw); > + > +fail: > + AUD_log(AUDIO_CAP, "Failed to initialize PW context"); > + if (pw->thread_loop) { > + pw_thread_loop_stop(pw->thread_loop); > + } > + if (pw->context) { > + g_clear_pointer(&pw->context, pw_context_destroy); > + } > + if (pw->thread_loop) { > + g_clear_pointer(&pw->thread_loop, pw_thread_loop_destroy); > + } > + return NULL; > +} > + > +static void > +qpw_audio_fini(void *opaque) > +{ > + pwaudio *pw = opaque; > + > + if (pw->thread_loop) { > + pw_thread_loop_stop(pw->thread_loop); > + } > + > + if (pw->core) { > + spa_hook_remove(&pw->core_listener); > + spa_zero(pw->core_listener); > + pw_core_disconnect(pw->core); > + } > + > + if (pw->context) { > + pw_context_destroy(pw->context); > + } > + pw_thread_loop_destroy(pw->thread_loop); > + > + g_free(pw); > +} > + > +static struct audio_pcm_ops qpw_pcm_ops = { > + .init_out = qpw_init_out, > + .fini_out = qpw_fini_out, > + .write = qpw_write, > + .buffer_get_free = qpw_buffer_get_free, > + .run_buffer_out = audio_generic_run_buffer_out, > + .enable_out = qpw_enable_out, > + .volume_out = qpw_volume_out, > + .volume_in = qpw_volume_in, > + > + .init_in = qpw_init_in, > + .fini_in = qpw_fini_in, > + .read = qpw_read, > + .run_buffer_in = audio_generic_run_buffer_in, > + .enable_in = qpw_enable_in > +}; > + > +static struct audio_driver pw_audio_driver = { > + .name = "pipewire", > + .descr = "http://www.pipewire.org/", > + .init = qpw_audio_init, > + .fini = qpw_audio_fini, > + .pcm_ops = &qpw_pcm_ops, > + .can_be_default = 1, > + .max_voices_out = INT_MAX, > + .max_voices_in = INT_MAX, > + .voice_size_out = sizeof(PWVoiceOut), > + .voice_size_in = sizeof(PWVoiceIn), > +}; > + > +static void > +register_audio_pw(void) > +{ > + audio_driver_register(&pw_audio_driver); > +} > + > +type_init(register_audio_pw); > diff --git a/audio/trace-events b/audio/trace-events > index e1ab643add..c764e5641b 100644 > --- a/audio/trace-events > +++ b/audio/trace-events > @@ -18,6 +18,14 @@ dbus_audio_register(const char *s, const char *dir) > "sender = %s, dir = %s" > dbus_audio_put_buffer_out(size_t len) "len = %zu" > dbus_audio_read(size_t len) "len = %zu" > > +# pwaudio.c > +pw_state_changed(int nodeid, const char *s) "node id: %d stream state: %s" > +pw_read(int32_t avail, uint32_t index, size_t len) "avail=%d index=%u > len=%zu" > +pw_write(int32_t filled, int32_t avail, uint32_t index, size_t len) > "filled=%d avail=%d index=%u len=%zu" > +pw_vol(const char *ret) "set volume: %s" > +pw_timer(uint64_t buf_samples) "timer period = %" PRIu64 > +pw_audio_init(void) "Initialize Pipewire context" > + > # audio.c > audio_timer_start(int interval) "interval %d ms" > audio_timer_stop(void) "" > diff --git a/meson.build b/meson.build > index 29f8644d6d..31bf280c0d 100644 > --- a/meson.build > +++ b/meson.build > @@ -730,6 +730,12 @@ if not get_option('jack').auto() or have_system > jack = dependency('jack', required: get_option('jack'), > method: 'pkg-config', kwargs: static_kwargs) > endif > +pipewire = not_found > +if not get_option('pipewire').auto() or (targetos == 'linux' and > have_system) > + pipewire = dependency('libpipewire-0.3', version: '>=0.3.60', > + required: get_option('pipewire'), > + method: 'pkg-config', kwargs: static_kwargs) > +endif > sndio = not_found > if not get_option('sndio').auto() or have_system > sndio = dependency('sndio', required: get_option('sndio'), > @@ -1667,6 +1673,7 @@ if have_system > 'jack': jack.found(), > 'oss': oss.found(), > 'pa': pulse.found(), > + 'pipewire': pipewire.found(), > 'sdl': sdl.found(), > 'sndio': sndio.found(), > } > @@ -3980,6 +3987,7 @@ if targetos == 'linux' > summary_info += {'ALSA support': alsa} > summary_info += {'PulseAudio support': pulse} > endif > +summary_info += {'Pipewire support': pipewire} > summary_info += {'JACK support': jack} > summary_info += {'brlapi support': brlapi} > summary_info += {'vde support': vde} > diff --git a/meson_options.txt b/meson_options.txt > index fc9447d267..9ae1ec7f47 100644 > --- a/meson_options.txt > +++ b/meson_options.txt > @@ -21,7 +21,7 @@ option('tls_priority', type : 'string', value : 'NORMAL', > option('default_devices', type : 'boolean', value : true, > description: 'Include a default selection of devices in emulators') > option('audio_drv_list', type: 'array', value: ['default'], > - choices: ['alsa', 'coreaudio', 'default', 'dsound', 'jack', 'oss', > 'pa', 'sdl', 'sndio'], > + choices: ['alsa', 'coreaudio', 'default', 'dsound', 'jack', 'oss', > 'pa', 'pipewire', 'sdl', 'sndio'], > description: 'Set audio driver list') > option('block_drv_rw_whitelist', type : 'string', value : '', > description: 'set block driver read-write whitelist (by default > affects only QEMU, not tools like qemu-img)') > @@ -255,6 +255,8 @@ option('oss', type: 'feature', value: 'auto', > description: 'OSS sound support') > option('pa', type: 'feature', value: 'auto', > description: 'PulseAudio sound support') > +option('pipewire', type: 'feature', value: 'auto', > + description: 'Pipewire sound support') > option('sndio', type: 'feature', value: 'auto', > description: 'sndio sound support') > > diff --git a/qapi/audio.json b/qapi/audio.json > index 4e54c00f51..e03396a7bc 100644 > --- a/qapi/audio.json > +++ b/qapi/audio.json > @@ -324,6 +324,47 @@ > '*out': 'AudiodevPaPerDirectionOptions', > '*server': 'str' } } > > +## > +# @AudiodevPipewirePerDirectionOptions: > +# > +# Options of the Pipewire backend that are used for both playback and > +# recording. > +# > +# @name: name of the sink/source to use > +# > +# @stream-name: name of the Pipewire stream created by qemu. Can be > +# used to identify the stream in Pipewire when you > +# create multiple Pipewire devices or run multiple qemu > +# instances (default: audiodev's id) > +# > +# @latency: latency you want Pipewire to achieve in microseconds > +# (default 46000) > +# > +# Since: 8.1 > +## > +{ 'struct': 'AudiodevPipewirePerDirectionOptions', > + 'base': 'AudiodevPerDirectionOptions', > + 'data': { > + '*name': 'str', > + '*stream-name': 'str', > + '*latency': 'uint32' } } > + > +## > +# @AudiodevPipewireOptions: > +# > +# Options of the Pipewire audio backend. > +# > +# @in: options of the capture stream > +# > +# @out: options of the playback stream > +# > +# Since: 8.1 > +## > +{ 'struct': 'AudiodevPipewireOptions', > + 'data': { > + '*in': 'AudiodevPipewirePerDirectionOptions', > + '*out': 'AudiodevPipewirePerDirectionOptions' } } > + > ## > # @AudiodevSdlPerDirectionOptions: > # > @@ -416,6 +457,7 @@ > { 'name': 'jack', 'if': 'CONFIG_AUDIO_JACK' }, > { 'name': 'oss', 'if': 'CONFIG_AUDIO_OSS' }, > { 'name': 'pa', 'if': 'CONFIG_AUDIO_PA' }, > + { 'name': 'pipewire', 'if': 'CONFIG_AUDIO_PIPEWIRE' }, > { 'name': 'sdl', 'if': 'CONFIG_AUDIO_SDL' }, > { 'name': 'sndio', 'if': 'CONFIG_AUDIO_SNDIO' }, > { 'name': 'spice', 'if': 'CONFIG_SPICE' }, > @@ -456,6 +498,8 @@ > 'if': 'CONFIG_AUDIO_OSS' }, > 'pa': { 'type': 'AudiodevPaOptions', > 'if': 'CONFIG_AUDIO_PA' }, > + 'pipewire': { 'type': 'AudiodevPipewireOptions', > + 'if': 'CONFIG_AUDIO_PIPEWIRE' }, > 'sdl': { 'type': 'AudiodevSdlOptions', > 'if': 'CONFIG_AUDIO_SDL' }, > 'sndio': { 'type': 'AudiodevSndioOptions', > diff --git a/qemu-options.hx b/qemu-options.hx > index 59bdf67a2c..2d908717bd 100644 > --- a/qemu-options.hx > +++ b/qemu-options.hx > @@ -779,6 +779,12 @@ DEF("audiodev", HAS_ARG, QEMU_OPTION_audiodev, > " in|out.name= source/sink device name\n" > " in|out.latency= desired latency in microseconds\n" > #endif > +#ifdef CONFIG_AUDIO_PIPEWIRE > + "-audiodev pipewire,id=id[,prop[=value][,...]]\n" > + " in|out.name= source/sink device name\n" > + " in|out.stream-name= name of pipewire stream\n" > + " in|out.latency= desired latency in microseconds\n" > +#endif > #ifdef CONFIG_AUDIO_SDL > "-audiodev sdl,id=id[,prop[=value][,...]]\n" > " in|out.buffer-count= number of buffers\n" > @@ -942,6 +948,21 @@ SRST > Desired latency in microseconds. The PulseAudio server will try > to honor this value but actual latencies may be lower or higher. > > +``-audiodev pipewire,id=id[,prop[=value][,...]]`` > + Creates a backend using Pipewire. This backend is available on > + most systems. > + > + Pipewire specific options are: > + > + ``in|out.latency=usecs`` > + Desired latency in microseconds. > + > + ``in|out.name=sink`` > + Use the specified source/sink for recording/playback. > + > + ``in|out.stream-name`` > + Specify the name of pipewire stream. > + > ``-audiodev sdl,id=id[,prop[=value][,...]]`` > Creates a backend using SDL. This backend is available on most > systems, but you should use your platform's native backend if > diff --git a/scripts/meson-buildoptions.sh b/scripts/meson-buildoptions.sh > index 009fab1515..ba1057b62c 100644 > --- a/scripts/meson-buildoptions.sh > +++ b/scripts/meson-buildoptions.sh > @@ -1,7 +1,8 @@ > # This file is generated by meson-buildoptions.py, do not edit! > meson_options_help() { > - printf "%s\n" ' --audio-drv-list=CHOICES Set audio driver list > [default] (choices: alsa/co' > - printf "%s\n" ' > reaudio/default/dsound/jack/oss/pa/sdl/sndio)' > + printf "%s\n" ' --audio-drv-list=CHOICES Set audio driver list > [default] (choices: al' > + printf "%s\n" ' > sa/coreaudio/default/dsound/jack/oss/pa/' > + printf "%s\n" ' pipewire/sdl/sndio)' > printf "%s\n" ' --block-drv-ro-whitelist=VALUE' > printf "%s\n" ' set block driver read-only > whitelist (by default' > printf "%s\n" ' affects only QEMU, not tools > like qemu-img)' > @@ -136,6 +137,7 @@ meson_options_help() { > printf "%s\n" ' oss OSS sound support' > printf "%s\n" ' pa PulseAudio sound support' > printf "%s\n" ' parallels parallels image format support' > + printf "%s\n" ' pipewire Pipewire sound support' > printf "%s\n" ' png PNG support with libpng' > printf "%s\n" ' pvrdma Enable PVRDMA support' > printf "%s\n" ' qcow1 qcow1 image format support' > @@ -370,6 +372,8 @@ _meson_option_parse() { > --disable-pa) printf "%s" -Dpa=disabled ;; > --enable-parallels) printf "%s" -Dparallels=enabled ;; > --disable-parallels) printf "%s" -Dparallels=disabled ;; > + --enable-pipewire) printf "%s" -Dpipewire=enabled ;; > + --disable-pipewire) printf "%s" -Dpipewire=disabled ;; > --with-pkgversion=*) quote_sh "-Dpkgversion=$2" ;; > --enable-png) printf "%s" -Dpng=enabled ;; > --disable-png) printf "%s" -Dpng=disabled ;; > -- > 2.39.1 > >
Am 12.04.23 um 15:59 schrieb Dorinda Bassey: > Hi Volker, > > It seems that for some unknown reason using audio_pcm_info_clear_buf in playback_process causes segmentation fault. Hence I moved the handling of buffer underruns from the playback process to the qpw_write process because that is the underlying cause of buffer underrun. > Hi Dorinda, I guess you made a mistake somewhere if you see a segmentation fault. This patch for v10 works fine on my computer. diff --git a/audio/pwaudio.c b/audio/pwaudio.c index f9da86059f..0f272d6744 100644 --- a/audio/pwaudio.c +++ b/audio/pwaudio.c @@ -79,7 +79,7 @@ stream_destroy(void *data) static void playback_on_process(void *data) { - PWVoice *v = (PWVoice *) data; + PWVoice *v = data; void *p; struct pw_buffer *b; struct spa_buffer *buf; @@ -108,19 +108,28 @@ playback_on_process(void *data) n_bytes = SPA_MIN(req, buf->datas[0].maxsize); /* get no of available bytes to read data from buffer */ - avail = spa_ringbuffer_get_read_index(&v->ring, &index); + if (avail <= 0) { + PWVoiceOut *vo = container_of(data, PWVoiceOut, v); - if (avail < (int32_t) n_bytes) { - n_bytes = avail; - } + audio_pcm_info_clear_buf(&vo->hw.info, p, n_bytes / v->frame_size); + } else { + if ((uint32_t)avail < n_bytes) { + /* + * PipeWire immediately calls this callback again if we provide + * less than n_bytes. Then audio_pcm_info_clear_buf() fills the + * rest of the buffer with silence. + */ + n_bytes = avail; + } - spa_ringbuffer_read_data(&v->ring, - v->buffer, RINGBUFFER_SIZE, - index & RINGBUFFER_MASK, p, n_bytes); + 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); + index += n_bytes; + spa_ringbuffer_read_update(&v->ring, index); + } buf->datas[0].chunk->offset = 0; buf->datas[0].chunk->stride = v->frame_size;
Hi Dorinda, > This commit adds a new audiodev backend to allow QEMU to use Pipewire as > both an audio sink and source. This backend is available on most systems > > Add Pipewire entry points for QEMU Pipewire audio backend > Add wrappers for QEMU Pipewire audio backend in qpw_pcm_ops() > qpw_write function returns the current state of the stream to pwaudio > and Writes some data to the server for playback streams using pipewire > spa_ringbuffer implementation. > qpw_read function returns the current state of the stream to pwaudio and > reads some data from the server for capture streams using pipewire > spa_ringbuffer implementation. These functions qpw_write and qpw_read > are called during playback and capture. > Added some functions that convert pw audio formats to QEMU audio format > and vice versa which would be needed in the pipewire audio sink and > source functions qpw_init_in() & qpw_init_out(). > These methods that implement playback and recording will create streams > for playback and capture that will start processing and will result in > the on_process callbacks to be called. > Built a connection to the Pipewire sound system server in the > qpw_audio_init() method. > > Signed-off-by: Dorinda Bassey <dbassey@redhat.com> > --- > v11: > handle buffer underruns in qpw_write > use local variable > change param name frame_size > fix format specifier > change trace value to trace quantum > > audio/audio.c | 3 + > audio/audio_template.h | 4 + > audio/meson.build | 1 + > audio/pwaudio.c | 913 ++++++++++++++++++++++++++++++++++ > audio/trace-events | 8 + > meson.build | 8 + > meson_options.txt | 4 +- > qapi/audio.json | 44 ++ > qemu-options.hx | 21 + > scripts/meson-buildoptions.sh | 8 +- > 10 files changed, 1011 insertions(+), 3 deletions(-) > create mode 100644 audio/pwaudio.c > > diff --git a/audio/audio.c b/audio/audio.c > index 70b096713c..90c7c49d11 100644 > --- a/audio/audio.c > +++ b/audio/audio.c > @@ -2061,6 +2061,9 @@ void audio_create_pdos(Audiodev *dev) > #ifdef CONFIG_AUDIO_PA > CASE(PA, pa, Pa); > #endif > +#ifdef CONFIG_AUDIO_PIPEWIRE > + CASE(PIPEWIRE, pipewire, Pipewire); > +#endif > #ifdef CONFIG_AUDIO_SDL > CASE(SDL, sdl, Sdl); > #endif > diff --git a/audio/audio_template.h b/audio/audio_template.h > index e42326c20d..dc0c74aa74 100644 > --- a/audio/audio_template.h > +++ b/audio/audio_template.h > @@ -362,6 +362,10 @@ AudiodevPerDirectionOptions *glue(audio_get_pdo_, TYPE)(Audiodev *dev) > case AUDIODEV_DRIVER_PA: > return qapi_AudiodevPaPerDirectionOptions_base(dev->u.pa.TYPE); > #endif > +#ifdef CONFIG_AUDIO_PIPEWIRE > + case AUDIODEV_DRIVER_PIPEWIRE: > + return qapi_AudiodevPipewirePerDirectionOptions_base(dev->u.pipewire.TYPE); > +#endif > #ifdef CONFIG_AUDIO_SDL > case AUDIODEV_DRIVER_SDL: > return qapi_AudiodevSdlPerDirectionOptions_base(dev->u.sdl.TYPE); > diff --git a/audio/meson.build b/audio/meson.build > index 0722224ba9..65a49c1a10 100644 > --- a/audio/meson.build > +++ b/audio/meson.build > @@ -19,6 +19,7 @@ foreach m : [ > ['sdl', sdl, files('sdlaudio.c')], > ['jack', jack, files('jackaudio.c')], > ['sndio', sndio, files('sndioaudio.c')], > + ['pipewire', pipewire, files('pwaudio.c')], > ['spice', spice, files('spiceaudio.c')] > ] > if m[1].found() > diff --git a/audio/pwaudio.c b/audio/pwaudio.c > new file mode 100644 > index 0000000000..adf1a538c0 > --- /dev/null > +++ b/audio/pwaudio.c > @@ -0,0 +1,913 @@ > +/* > + * QEMU Pipewire audio driver > + * > + * Copyright (c) 2023 Red Hat Inc. > + * > + * Author: Dorinda Bassey <dbassey@redhat.com> > + * > + * SPDX-License-Identifier: GPL-2.0-or-later > + */ > + > +#include "qemu/osdep.h" > +#include "qemu/module.h" > +#include "audio.h" > +#include <errno.h> > +#include "qemu/error-report.h" > +#include <spa/param/audio/format-utils.h> > +#include <spa/utils/ringbuffer.h> > +#include <spa/utils/result.h> > +#include <spa/param/props.h> > + > +#include <pipewire/pipewire.h> > +#include "trace.h" > + > +#define AUDIO_CAP "pipewire" > +#define RINGBUFFER_SIZE (1u << 22) > +#define RINGBUFFER_MASK (RINGBUFFER_SIZE - 1) > + > +#include "audio_int.h" > + > +typedef struct pwvolume { > + uint32_t channels; > + float values[SPA_AUDIO_MAX_CHANNELS]; > +} pwvolume; > + > +typedef struct pwaudio { > + Audiodev *dev; > + struct pw_thread_loop *thread_loop; > + struct pw_context *context; > + > + struct pw_core *core; > + struct spa_hook core_listener; > + int last_seq, pending_seq, error; > +} pwaudio; > + > +typedef struct PWVoice { > + pwaudio *g; > + struct pw_stream *stream; > + struct spa_hook stream_listener; > + struct spa_audio_info_raw info; > + uint32_t highwater_mark; > + uint32_t sample_size, req; The older patches used the correct name frame_size instead of sample_size. Please revert this. Further below I've written an explanation. > + struct spa_ringbuffer ring; > + uint8_t buffer[RINGBUFFER_SIZE]; > + > + pwvolume volume; > + bool muted; > +} PWVoice; > + > +typedef struct PWVoiceOut { > + HWVoiceOut hw; > + PWVoice v; > +} PWVoiceOut; > + > +typedef struct PWVoiceIn { > + HWVoiceIn hw; > + PWVoice v; > +} PWVoiceIn; > + > +static void > +stream_destroy(void *data) > +{ > + PWVoice *v = (PWVoice *) data; > + spa_hook_remove(&v->stream_listener); > + v->stream = NULL; > +} > + > +/* output data processing function to read stuffs from the buffer */ > +static void > +playback_on_process(void *data) > +{ > + PWVoice *v = (PWVoice *) data; > + void *p; > + struct pw_buffer *b; > + struct spa_buffer *buf; > + uint32_t req, index, n_bytes; > + int32_t avail; > + > + assert(v->stream); > + > + /* obtain a buffer to read from */ > + b = pw_stream_dequeue_buffer(v->stream); > + if (b == NULL) { > + error_report("out of buffers: %s", strerror(errno)); > + return; > + } > + > + buf = b->buffer; > + p = buf->datas[0].data; > + if (p == NULL) { > + return; > + } > + /* calculate the total no of bytes to read data from buffer */ > + req = b->requested * v->sample_size; > + if (req == 0) { > + req = v->req; > + } > + n_bytes = SPA_MIN(req, buf->datas[0].maxsize); > + > + /* get no of available bytes to read data from buffer */ > + > + avail = spa_ringbuffer_get_read_index(&v->ring, &index); > + > + if (avail <= 0) { > + /* underrun, can't really happen but if it does we */ > + /* do nothing and wait for more data */ > + error_report("%p: underrun read:%u avail:%d", p, index, avail); Please don't do that. The PipeWire audio threads have a higher priority than the QEMU thread. On a heavily loaded system the callbacks will be called even if QEMU nearly stalled. So this is the right place to detect a buffer underflow and continue to write silent audio frames. There's no need for an error report. Buffer underflows are expected and the QEMU users can hear the problem. > + } 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->sample_size; > + buf->datas[0].chunk->size = n_bytes; > + > + /* queue the buffer for playback */ > + pw_stream_queue_buffer(v->stream, b); > +} > + > +/* output data processing function to generate stuffs in the buffer */ > +static void > +capture_on_process(void *data) > +{ > + PWVoice *v = (PWVoice *) data; > + void *p; > + struct pw_buffer *b; > + struct spa_buffer *buf; > + int32_t filled; > + uint32_t index, offs, n_bytes; > + > + assert(v->stream); > + > + /* obtain a buffer */ > + b = pw_stream_dequeue_buffer(v->stream); > + if (b == NULL) { > + error_report("out of buffers: %s", strerror(errno)); > + return; > + } > + > + /* Write data into buffer */ > + buf = b->buffer; > + p = buf->datas[0].data; > + if (p == NULL) { > + return; > + } > + offs = SPA_MIN(buf->datas[0].chunk->offset, buf->datas[0].maxsize); > + n_bytes = SPA_MIN(buf->datas[0].chunk->size, buf->datas[0].maxsize - offs); > + > + filled = spa_ringbuffer_get_write_index(&v->ring, &index); > + > + > + if (filled < 0) { > + error_report("%p: underrun write:%u filled:%d", p, index, filled); > + } else { > + if ((uint32_t) filled + n_bytes > RINGBUFFER_SIZE) { > + error_report("%p: overrun write:%u filled:%d + size:%u > max:%u", > + p, index, filled, n_bytes, RINGBUFFER_SIZE); > + } > + } > + spa_ringbuffer_write_data(&v->ring, > + v->buffer, RINGBUFFER_SIZE, > + index & RINGBUFFER_MASK, > + SPA_PTROFF(p, offs, void), n_bytes); > + index += n_bytes; > + spa_ringbuffer_write_update(&v->ring, index); > + > + /* queue the buffer for playback */ > + pw_stream_queue_buffer(v->stream, b); > +} > + > +static void > +on_stream_state_changed(void *data, enum pw_stream_state old, > + enum pw_stream_state state, const char *error) > +{ > + PWVoice *v = (PWVoice *) data; > + > + trace_pw_state_changed(pw_stream_get_node_id(v->stream), > + pw_stream_state_as_string(state)); > + > + switch (state) { > + case PW_STREAM_STATE_ERROR: > + case PW_STREAM_STATE_UNCONNECTED: > + break; > + case PW_STREAM_STATE_PAUSED: > + case PW_STREAM_STATE_CONNECTING: > + case PW_STREAM_STATE_STREAMING: > + break; > + } > +} > + > +static const struct pw_stream_events capture_stream_events = { > + PW_VERSION_STREAM_EVENTS, > + .destroy = stream_destroy, > + .state_changed = on_stream_state_changed, > + .process = capture_on_process > +}; > + > +static const struct pw_stream_events playback_stream_events = { > + PW_VERSION_STREAM_EVENTS, > + .destroy = stream_destroy, > + .state_changed = on_stream_state_changed, > + .process = playback_on_process > +}; > + > +static size_t > +qpw_read(HWVoiceIn *hw, void *data, size_t len) > +{ > + PWVoiceIn *pw = (PWVoiceIn *) hw; > + PWVoice *v = &pw->v; > + pwaudio *c = v->g; > + const char *error = NULL; > + size_t l; > + int32_t avail; > + uint32_t index; > + > + pw_thread_loop_lock(c->thread_loop); > + if (pw_stream_get_state(v->stream, &error) != PW_STREAM_STATE_STREAMING) { > + /* wait for stream to become ready */ > + l = 0; > + goto done_unlock; > + } > + /* get no of available bytes to read data from buffer */ > + avail = spa_ringbuffer_get_read_index(&v->ring, &index); > + > + trace_pw_read(avail, index, len); > + > + if (avail < (int32_t) len) { > + len = avail; > + } > + > + spa_ringbuffer_read_data(&v->ring, > + v->buffer, RINGBUFFER_SIZE, > + index & RINGBUFFER_MASK, data, len); > + index += len; > + spa_ringbuffer_read_update(&v->ring, index); > + l = len; > + > +done_unlock: > + pw_thread_loop_unlock(c->thread_loop); > + return l; > +} > + > +static size_t qpw_buffer_get_free(HWVoiceOut *hw) > +{ > + PWVoiceOut *pw = (PWVoiceOut *)hw; > + PWVoice *v = &pw->v; > + pwaudio *c = v->g; > + const char *error = NULL; > + int32_t filled, avail; > + uint32_t index; > + > + pw_thread_loop_lock(c->thread_loop); > + if (pw_stream_get_state(v->stream, &error) != PW_STREAM_STATE_STREAMING) { > + /* wait for stream to become ready */ > + avail = 0; > + goto done_unlock; > + } > + > + filled = spa_ringbuffer_get_write_index(&v->ring, &index); > + avail = v->highwater_mark - filled; > + > +done_unlock: > + pw_thread_loop_unlock(c->thread_loop); > + return avail; > +} > + > +static size_t > +qpw_write(HWVoiceOut *hw, void *data, size_t len) > +{ > + PWVoiceOut *pw = (PWVoiceOut *) hw; > + PWVoice *v = &pw->v; > + pwaudio *c = v->g; > + const char *error = NULL; > + int32_t filled, avail; > + uint32_t index; > + > + pw_thread_loop_lock(c->thread_loop); > + if (pw_stream_get_state(v->stream, &error) != PW_STREAM_STATE_STREAMING) { > + /* wait for stream to become ready */ > + len = 0; > + goto done_unlock; > + } > + filled = spa_ringbuffer_get_write_index(&v->ring, &index); > + avail = v->highwater_mark - filled; > + > + trace_pw_write(filled, avail, index, len); > + > + if (len > avail) { > + len = avail; > + } > + > + if (filled < 0) { > + audio_pcm_info_clear_buf(&hw->info, data, len / hw->info.bytes_per_frame); - audio_pcm_info_clear_buf(&hw->info, data, len / hw->info.bytes_per_frame); There is no way to reach this code. > + error_report("%p: underrun write:%u filled:%d", pw, index, filled); > + } else { > + if ((uint32_t) filled + len > RINGBUFFER_SIZE) { > + error_report("%p: overrun write:%u filled:%d + size:%zu > max:%u", > + pw, index, filled, len, RINGBUFFER_SIZE); > + } > + } > + > + spa_ringbuffer_write_data(&v->ring, > + v->buffer, RINGBUFFER_SIZE, > + index & RINGBUFFER_MASK, data, len); > + index += len; > + spa_ringbuffer_write_update(&v->ring, index); > + > +done_unlock: > + pw_thread_loop_unlock(c->thread_loop); > + return len; > +} > + > +static int > +audfmt_to_pw(AudioFormat fmt, int endianness) > +{ > + int format; > + > + switch (fmt) { > + case AUDIO_FORMAT_S8: > + format = SPA_AUDIO_FORMAT_S8; > + break; > + case AUDIO_FORMAT_U8: > + format = SPA_AUDIO_FORMAT_U8; > + break; > + case AUDIO_FORMAT_S16: > + format = endianness ? SPA_AUDIO_FORMAT_S16_BE : SPA_AUDIO_FORMAT_S16_LE; > + break; > + case AUDIO_FORMAT_U16: > + format = endianness ? SPA_AUDIO_FORMAT_U16_BE : SPA_AUDIO_FORMAT_U16_LE; > + break; > + case AUDIO_FORMAT_S32: > + format = endianness ? SPA_AUDIO_FORMAT_S32_BE : SPA_AUDIO_FORMAT_S32_LE; > + break; > + case AUDIO_FORMAT_U32: > + format = endianness ? SPA_AUDIO_FORMAT_U32_BE : SPA_AUDIO_FORMAT_U32_LE; > + break; > + case AUDIO_FORMAT_F32: > + format = endianness ? SPA_AUDIO_FORMAT_F32_BE : SPA_AUDIO_FORMAT_F32_LE; > + break; > + default: > + dolog("Internal logic error: Bad audio format %d\n", fmt); > + format = SPA_AUDIO_FORMAT_U8; > + break; > + } > + return format; > +} > + > +static AudioFormat > +pw_to_audfmt(enum spa_audio_format fmt, int *endianness, > + uint32_t *sample_size) > +{ > + switch (fmt) { > + case SPA_AUDIO_FORMAT_S8: > + *sample_size = 1; > + return AUDIO_FORMAT_S8; > + case SPA_AUDIO_FORMAT_U8: > + *sample_size = 1; > + return AUDIO_FORMAT_U8; > + case SPA_AUDIO_FORMAT_S16_BE: > + *sample_size = 2; > + *endianness = 1; > + return AUDIO_FORMAT_S16; > + case SPA_AUDIO_FORMAT_S16_LE: > + *sample_size = 2; > + *endianness = 0; > + return AUDIO_FORMAT_S16; > + case SPA_AUDIO_FORMAT_U16_BE: > + *sample_size = 2; > + *endianness = 1; > + return AUDIO_FORMAT_U16; > + case SPA_AUDIO_FORMAT_U16_LE: > + *sample_size = 2; > + *endianness = 0; > + return AUDIO_FORMAT_U16; > + case SPA_AUDIO_FORMAT_S32_BE: > + *sample_size = 4; > + *endianness = 1; > + return AUDIO_FORMAT_S32; > + case SPA_AUDIO_FORMAT_S32_LE: > + *sample_size = 4; > + *endianness = 0; > + return AUDIO_FORMAT_S32; > + case SPA_AUDIO_FORMAT_U32_BE: > + *sample_size = 4; > + *endianness = 1; > + return AUDIO_FORMAT_U32; > + case SPA_AUDIO_FORMAT_U32_LE: > + *sample_size = 4; > + *endianness = 0; > + return AUDIO_FORMAT_U32; > + case SPA_AUDIO_FORMAT_F32_BE: > + *sample_size = 4; > + *endianness = 1; > + return AUDIO_FORMAT_F32; > + case SPA_AUDIO_FORMAT_F32_LE: > + *sample_size = 4; > + *endianness = 0; > + return AUDIO_FORMAT_F32; > + default: > + *sample_size = 1; > + dolog("Internal logic error: Bad spa_audio_format %d\n", fmt); > + return AUDIO_FORMAT_U8; > + } > +} > + > +static int > +create_stream(pwaudio *c, PWVoice *v, const char *stream_name, > + const char *name, enum spa_direction dir) > +{ > + int res; > + uint32_t n_params; > + const struct spa_pod *params[2]; > + uint8_t buffer[1024]; > + struct spa_pod_builder b; > + uint64_t buf_samples; > + struct pw_properties *props; > + > + props = pw_properties_new(NULL, NULL); > + > + /* 75% of the timer period for faster updates */ > + buf_samples = (uint64_t)v->g->dev->timer_period * v->info.rate > + * 3 / 4 / 1000000; > + trace_pw_timer(v->g->dev->timer_period); > + pw_properties_setf(props, PW_KEY_NODE_LATENCY, "%" PRIu64 "/%u", > + buf_samples, v->info.rate); > + > + if (name) { > + pw_properties_set(props, PW_KEY_TARGET_OBJECT, name); > + } > + v->stream = pw_stream_new(c->core, stream_name, props); > + > + if (v->stream == NULL) { > + return -1; > + } > + > + if (dir == SPA_DIRECTION_INPUT) { > + pw_stream_add_listener(v->stream, > + &v->stream_listener, &capture_stream_events, v); > + } else { > + pw_stream_add_listener(v->stream, > + &v->stream_listener, &playback_stream_events, v); > + } > + > + n_params = 0; > + spa_pod_builder_init(&b, buffer, sizeof(buffer)); > + params[n_params++] = spa_format_audio_raw_build(&b, > + SPA_PARAM_EnumFormat, > + &v->info); > + > + /* connect the stream to a sink or source */ > + res = pw_stream_connect(v->stream, > + dir == > + SPA_DIRECTION_INPUT ? PW_DIRECTION_INPUT : > + PW_DIRECTION_OUTPUT, PW_ID_ANY, > + PW_STREAM_FLAG_AUTOCONNECT | > + PW_STREAM_FLAG_INACTIVE | > + PW_STREAM_FLAG_MAP_BUFFERS | > + PW_STREAM_FLAG_RT_PROCESS, params, n_params); > + if (res < 0) { > + pw_stream_destroy(v->stream); > + return -1; > + } > + > + return 0; > +} > + > +static int > +qpw_stream_new(pwaudio *c, PWVoice *v, const char *stream_name, > + const char *name, enum spa_direction dir) > +{ > + int r; > + > + switch (v->info.channels) { > + case 8: > + v->info.position[0] = SPA_AUDIO_CHANNEL_FL; > + v->info.position[1] = SPA_AUDIO_CHANNEL_FR; > + v->info.position[2] = SPA_AUDIO_CHANNEL_FC; > + v->info.position[3] = SPA_AUDIO_CHANNEL_LFE; > + v->info.position[4] = SPA_AUDIO_CHANNEL_RL; > + v->info.position[5] = SPA_AUDIO_CHANNEL_RR; > + v->info.position[6] = SPA_AUDIO_CHANNEL_SL; > + v->info.position[7] = SPA_AUDIO_CHANNEL_SR; > + break; > + case 6: > + v->info.position[0] = SPA_AUDIO_CHANNEL_FL; > + v->info.position[1] = SPA_AUDIO_CHANNEL_FR; > + v->info.position[2] = SPA_AUDIO_CHANNEL_FC; > + v->info.position[3] = SPA_AUDIO_CHANNEL_LFE; > + v->info.position[4] = SPA_AUDIO_CHANNEL_RL; > + v->info.position[5] = SPA_AUDIO_CHANNEL_RR; > + break; > + case 5: > + v->info.position[0] = SPA_AUDIO_CHANNEL_FL; > + v->info.position[1] = SPA_AUDIO_CHANNEL_FR; > + v->info.position[2] = SPA_AUDIO_CHANNEL_FC; > + v->info.position[3] = SPA_AUDIO_CHANNEL_LFE; > + v->info.position[4] = SPA_AUDIO_CHANNEL_RC; > + break; > + case 4: > + v->info.position[0] = SPA_AUDIO_CHANNEL_FL; > + v->info.position[1] = SPA_AUDIO_CHANNEL_FR; > + v->info.position[2] = SPA_AUDIO_CHANNEL_FC; > + v->info.position[3] = SPA_AUDIO_CHANNEL_RC; > + break; > + case 3: > + v->info.position[0] = SPA_AUDIO_CHANNEL_FL; > + v->info.position[1] = SPA_AUDIO_CHANNEL_FR; > + v->info.position[2] = SPA_AUDIO_CHANNEL_LFE; > + break; > + case 2: > + v->info.position[0] = SPA_AUDIO_CHANNEL_FL; > + v->info.position[1] = SPA_AUDIO_CHANNEL_FR; > + break; > + case 1: > + v->info.position[0] = SPA_AUDIO_CHANNEL_MONO; > + break; > + default: > + for (size_t i = 0; i < v->info.channels; i++) { > + v->info.position[i] = SPA_AUDIO_CHANNEL_UNKNOWN; > + } > + break; > + } > + > + /* create a new unconnected pwstream */ > + r = create_stream(c, v, stream_name, name, dir); > + if (r < 0) { > + AUD_log(AUDIO_CAP, "Failed to create stream."); > + return -1; > + } > + > + return r; > +} > + > +static int > +qpw_init_out(HWVoiceOut *hw, struct audsettings *as, void *drv_opaque) > +{ > + PWVoiceOut *pw = (PWVoiceOut *) hw; > + PWVoice *v = &pw->v; > + struct audsettings obt_as = *as; > + pwaudio *c = v->g = drv_opaque; > + AudiodevPipewireOptions *popts = &c->dev->u.pipewire; > + AudiodevPipewirePerDirectionOptions *ppdo = popts->out; > + int r; > + > + pw_thread_loop_lock(c->thread_loop); > + > + v->info.format = audfmt_to_pw(as->fmt, as->endianness); > + v->info.channels = as->nchannels; > + v->info.rate = as->freq; > + > + obt_as.fmt = > + pw_to_audfmt(v->info.format, &obt_as.endianness, &v->sample_size); The third argument of pw_to_audfmt() returns the sample size. > + v->sample_size *= as->nchannels; Here you calculate the frame size from the sample size. The correct name is v->frame_size. I'm aware the rest of QEMU quite often uses samples as a synonym for frames. But new code should get the variable names right. > + > + v->req = (uint64_t)c->dev->timer_period * v->info.rate > + * 1 / 2 / 1000000 * v->sample_size; > + > + /* call the function that creates a new stream for playback */ > + r = qpw_stream_new(c, v, ppdo->stream_name ? : c->dev->id, > + ppdo->name, SPA_DIRECTION_OUTPUT); > + if (r < 0) { > + error_report("qpw_stream_new for playback failed"); > + pw_thread_loop_unlock(c->thread_loop); > + return -1; > + } > + > + /* report the audio format we support */ > + audio_pcm_init_info(&hw->info, &obt_as); > + > + /* report the buffer size to qemu */ > + hw->samples = audio_buffer_frames( > + qapi_AudiodevPipewirePerDirectionOptions_base(ppdo), &obt_as, 46440); > + v->highwater_mark = MIN(RINGBUFFER_SIZE, > + (ppdo->has_latency ? ppdo->latency : 46440) > + * (uint64_t)v->info.rate / 1000000 * v->sample_size); > + > + pw_thread_loop_unlock(c->thread_loop); > + return 0; > +} > + > +static int > +qpw_init_in(HWVoiceIn *hw, struct audsettings *as, void *drv_opaque) > +{ > + PWVoiceIn *pw = (PWVoiceIn *) hw; > + PWVoice *v = &pw->v; > + struct audsettings obt_as = *as; > + pwaudio *c = v->g = drv_opaque; > + AudiodevPipewireOptions *popts = &c->dev->u.pipewire; > + AudiodevPipewirePerDirectionOptions *ppdo = popts->in; > + int r; > + > + pw_thread_loop_lock(c->thread_loop); > + > + v->info.format = audfmt_to_pw(as->fmt, as->endianness); > + v->info.channels = as->nchannels; > + v->info.rate = as->freq; > + > + obt_as.fmt = > + pw_to_audfmt(v->info.format, &obt_as.endianness, &v->sample_size); > + v->sample_size *= as->nchannels; > + > + /* call the function that creates a new stream for recording */ > + r = qpw_stream_new(c, v, ppdo->stream_name ? : c->dev->id, > + ppdo->name, SPA_DIRECTION_INPUT); > + if (r < 0) { > + error_report("qpw_stream_new for recording failed"); > + pw_thread_loop_unlock(c->thread_loop); > + return -1; > + } > + > + /* report the audio format we support */ > + audio_pcm_init_info(&hw->info, &obt_as); > + > + /* report the buffer size to qemu */ > + hw->samples = audio_buffer_frames( > + qapi_AudiodevPipewirePerDirectionOptions_base(ppdo), &obt_as, 46440); > + > + pw_thread_loop_unlock(c->thread_loop); > + return 0; > +} > + > +static void > +qpw_fini_out(HWVoiceOut *hw) > +{ > + PWVoiceOut *pw = (PWVoiceOut *) hw; > + PWVoice *v = &pw->v; > + > + if (v->stream) { > + pwaudio *c = v->g; > + pw_thread_loop_lock(c->thread_loop); > + pw_stream_destroy(v->stream); > + v->stream = NULL; > + pw_thread_loop_unlock(c->thread_loop); > + } > +} > + > +static void > +qpw_fini_in(HWVoiceIn *hw) > +{ > + PWVoiceIn *pw = (PWVoiceIn *) hw; > + PWVoice *v = &pw->v; > + > + if (v->stream) { > + pwaudio *c = v->g; > + pw_thread_loop_lock(c->thread_loop); > + pw_stream_destroy(v->stream); > + v->stream = NULL; > + pw_thread_loop_unlock(c->thread_loop); > + } > +} > + > +static void > +qpw_enable_out(HWVoiceOut *hw, bool enable) > +{ > + PWVoiceOut *po = (PWVoiceOut *) hw; > + PWVoice *v = &po->v; > + pwaudio *c = v->g; > + pw_thread_loop_lock(c->thread_loop); > + pw_stream_set_active(v->stream, enable); > + pw_thread_loop_unlock(c->thread_loop); > +} > + > +static void > +qpw_enable_in(HWVoiceIn *hw, bool enable) > +{ > + PWVoiceIn *pi = (PWVoiceIn *) hw; > + PWVoice *v = &pi->v; > + pwaudio *c = v->g; > + pw_thread_loop_lock(c->thread_loop); > + pw_stream_set_active(v->stream, enable); > + pw_thread_loop_unlock(c->thread_loop); > +} > + > +static void > +qpw_volume_out(HWVoiceOut *hw, Volume *vol) > +{ > + PWVoiceOut *pw = (PWVoiceOut *) hw; > + PWVoice *v = &pw->v; > + pwaudio *c = v->g; > + int i, ret; > + > + pw_thread_loop_lock(c->thread_loop); > + v->volume.channels = vol->channels; > + > + for (i = 0; i < vol->channels; ++i) { > + v->volume.values[i] = (float)vol->vol[i] / 255; > + } > + > + ret = pw_stream_set_control(v->stream, > + SPA_PROP_channelVolumes, v->volume.channels, v->volume.values, 0); > + trace_pw_vol(ret == 0 ? "success" : "failed"); > + > + v->muted = vol->mute; > + float val = v->muted ? 1.f : 0.f; > + ret = pw_stream_set_control(v->stream, SPA_PROP_mute, 1, &val, 0); > + pw_thread_loop_unlock(c->thread_loop); > +} > + > +static void > +qpw_volume_in(HWVoiceIn *hw, Volume *vol) > +{ > + PWVoiceIn *pw = (PWVoiceIn *) hw; > + PWVoice *v = &pw->v; > + pwaudio *c = v->g; > + int i, ret; > + > + pw_thread_loop_lock(c->thread_loop); > + v->volume.channels = vol->channels; > + > + for (i = 0; i < vol->channels; ++i) { > + v->volume.values[i] = (float)vol->vol[i] / 255; > + } > + > + ret = pw_stream_set_control(v->stream, > + SPA_PROP_channelVolumes, v->volume.channels, v->volume.values, 0); > + trace_pw_vol(ret == 0 ? "success" : "failed"); > + > + v->muted = vol->mute; > + float val = v->muted ? 1.f : 0.f; > + ret = pw_stream_set_control(v->stream, SPA_PROP_mute, 1, &val, 0); > + pw_thread_loop_unlock(c->thread_loop); > +} > + > +static int wait_resync(pwaudio *pw) > +{ > + int res; > + pw->pending_seq = pw_core_sync(pw->core, PW_ID_CORE, pw->pending_seq); > + > + while (true) { > + pw_thread_loop_wait(pw->thread_loop); > + > + res = pw->error; > + if (res < 0) { > + pw->error = 0; > + return res; > + } > + if (pw->pending_seq == pw->last_seq) { > + break; > + } > + } > + return 0; > +} > +static void > +on_core_error(void *data, uint32_t id, int seq, int res, const char *message) > +{ > + pwaudio *pw = data; > + > + error_report("error id:%u seq:%d res:%d (%s): %s", > + id, seq, res, spa_strerror(res), message); > + > + /* stop and exit the thread loop */ > + pw_thread_loop_signal(pw->thread_loop, FALSE); > +} > + > +static void > +on_core_done(void *data, uint32_t id, int seq) > +{ > + pwaudio *pw = data; > + assert(id == PW_ID_CORE); > + pw->last_seq = seq; > + if (pw->pending_seq == seq) { > + /* stop and exit the thread loop */ > + pw_thread_loop_signal(pw->thread_loop, FALSE); > + } > +} > + > +static const struct pw_core_events core_events = { > + PW_VERSION_CORE_EVENTS, > + .done = on_core_done, > + .error = on_core_error, > +}; > + > +static void * > +qpw_audio_init(Audiodev *dev) > +{ > + g_autofree pwaudio *pw = g_new0(pwaudio, 1); > + pw_init(NULL, NULL); > + > + trace_pw_audio_init(); > + assert(dev->driver == AUDIODEV_DRIVER_PIPEWIRE); > + > + pw->dev = dev; > + pw->thread_loop = pw_thread_loop_new("Pipewire thread loop", NULL); > + if (pw->thread_loop == NULL) { > + error_report("Could not create Pipewire loop"); > + goto fail; > + } > + > + pw->context = > + pw_context_new(pw_thread_loop_get_loop(pw->thread_loop), NULL, 0); > + if (pw->context == NULL) { > + error_report("Could not create Pipewire context"); > + goto fail; > + } > + > + if (pw_thread_loop_start(pw->thread_loop) < 0) { > + error_report("Could not start Pipewire loop"); > + goto fail; > + } > + > + pw_thread_loop_lock(pw->thread_loop); > + > + pw->core = pw_context_connect(pw->context, NULL, 0); > + if (pw->core == NULL) { > + pw_thread_loop_unlock(pw->thread_loop); > + goto fail; > + } > + > + if (pw_core_add_listener(pw->core, &pw->core_listener, > + &core_events, pw) < 0) { > + pw_thread_loop_unlock(pw->thread_loop); > + goto fail; > + } > + if (wait_resync(pw) < 0) { > + pw_thread_loop_unlock(pw->thread_loop); > + } > + > + pw_thread_loop_unlock(pw->thread_loop); > + > + return g_steal_pointer(&pw); > + > +fail: > + AUD_log(AUDIO_CAP, "Failed to initialize PW context"); > + if (pw->thread_loop) { > + pw_thread_loop_stop(pw->thread_loop); > + } > + if (pw->context) { > + g_clear_pointer(&pw->context, pw_context_destroy); > + } > + if (pw->thread_loop) { > + g_clear_pointer(&pw->thread_loop, pw_thread_loop_destroy); > + } > + return NULL; > +} > + > +static void > +qpw_audio_fini(void *opaque) > +{ > + pwaudio *pw = opaque; > + > + if (pw->thread_loop) { > + pw_thread_loop_stop(pw->thread_loop); > + } > + > + if (pw->core) { > + spa_hook_remove(&pw->core_listener); > + spa_zero(pw->core_listener); > + pw_core_disconnect(pw->core); > + } > + > + if (pw->context) { > + pw_context_destroy(pw->context); > + } > + pw_thread_loop_destroy(pw->thread_loop); > + > + g_free(pw); > +} > + > +static struct audio_pcm_ops qpw_pcm_ops = { > + .init_out = qpw_init_out, > + .fini_out = qpw_fini_out, > + .write = qpw_write, > + .buffer_get_free = qpw_buffer_get_free, > + .run_buffer_out = audio_generic_run_buffer_out, > + .enable_out = qpw_enable_out, > + .volume_out = qpw_volume_out, > + .volume_in = qpw_volume_in, > + > + .init_in = qpw_init_in, > + .fini_in = qpw_fini_in, > + .read = qpw_read, > + .run_buffer_in = audio_generic_run_buffer_in, > + .enable_in = qpw_enable_in > +}; > + > +static struct audio_driver pw_audio_driver = { > + .name = "pipewire", > + .descr = "http://www.pipewire.org/", > + .init = qpw_audio_init, > + .fini = qpw_audio_fini, > + .pcm_ops = &qpw_pcm_ops, > + .can_be_default = 1, > + .max_voices_out = INT_MAX, > + .max_voices_in = INT_MAX, > + .voice_size_out = sizeof(PWVoiceOut), > + .voice_size_in = sizeof(PWVoiceIn), > +}; > + > +static void > +register_audio_pw(void) > +{ > + audio_driver_register(&pw_audio_driver); > +} > + > +type_init(register_audio_pw); > diff --git a/audio/trace-events b/audio/trace-events > index e1ab643add..c764e5641b 100644 > --- a/audio/trace-events > +++ b/audio/trace-events > @@ -18,6 +18,14 @@ dbus_audio_register(const char *s, const char *dir) "sender = %s, dir = %s" > dbus_audio_put_buffer_out(size_t len) "len = %zu" > dbus_audio_read(size_t len) "len = %zu" > > +# pwaudio.c > +pw_state_changed(int nodeid, const char *s) "node id: %d stream state: %s" > +pw_read(int32_t avail, uint32_t index, size_t len) "avail=%d index=%u len=%zu" > +pw_write(int32_t filled, int32_t avail, uint32_t index, size_t len) "filled=%d avail=%d index=%u len=%zu" > +pw_vol(const char *ret) "set volume: %s" > +pw_timer(uint64_t buf_samples) "timer period = %" PRIu64 Sorry, I was not very clear last time. I wrote 'quantum' but I meant the PipeWire scheduling period. -pw_timer(uint64_t buf_samples) "timer period = %" PRIu64 +pw_period(uint64_t quant, uint32_t rate) "period=%" PRIu64 "/%u" This is the same you see with pw-top. S ID QUANT RATE WAIT BUSY W/Q B/Q ERR FORMAT NAME S 28 0 0 --- --- --- --- 0 Dummy-Driver S 29 0 0 --- --- --- --- 0 Freewheel-Driver S 37 0 0 --- --- --- --- 0 Midi-Bridge S 46 0 0 --- --- --- --- 0 alsa_output.pci-0000_00_03.0.hdmi-stereo R 47 256 48000 102,2us 22,1us 0,02 0,00 1 S32LE 2 48000 alsa_output.pci-0000_00_1b.0.analog-stereo R 63 240 32000 65,4us 4,9us 0,01 0,00 1 U8 1 32000 + qemu-system-x86_64 R 67 330 44100 32,8us 33,4us 0,01 0,01 1 S16LE 2 44100 + qemu-system-x86_64 S 48 0 0 --- --- --- --- 0 alsa_input.pci-0000_00_1b.0.analog-stereo S 68 0 0 --- --- --- --- 0 qemu-system-x86_64 With best regards, Volker > +pw_audio_init(void) "Initialize Pipewire context" > + > # audio.c > audio_timer_start(int interval) "interval %d ms" > audio_timer_stop(void) "" > diff --git a/meson.build b/meson.build > index 29f8644d6d..31bf280c0d 100644 > --- a/meson.build > +++ b/meson.build > @@ -730,6 +730,12 @@ if not get_option('jack').auto() or have_system > jack = dependency('jack', required: get_option('jack'), > method: 'pkg-config', kwargs: static_kwargs) > endif > +pipewire = not_found > +if not get_option('pipewire').auto() or (targetos == 'linux' and have_system) > + pipewire = dependency('libpipewire-0.3', version: '>=0.3.60', > + required: get_option('pipewire'), > + method: 'pkg-config', kwargs: static_kwargs) > +endif > sndio = not_found > if not get_option('sndio').auto() or have_system > sndio = dependency('sndio', required: get_option('sndio'), > @@ -1667,6 +1673,7 @@ if have_system > 'jack': jack.found(), > 'oss': oss.found(), > 'pa': pulse.found(), > + 'pipewire': pipewire.found(), > 'sdl': sdl.found(), > 'sndio': sndio.found(), > } > @@ -3980,6 +3987,7 @@ if targetos == 'linux' > summary_info += {'ALSA support': alsa} > summary_info += {'PulseAudio support': pulse} > endif > +summary_info += {'Pipewire support': pipewire} > summary_info += {'JACK support': jack} > summary_info += {'brlapi support': brlapi} > summary_info += {'vde support': vde} > diff --git a/meson_options.txt b/meson_options.txt > index fc9447d267..9ae1ec7f47 100644 > --- a/meson_options.txt > +++ b/meson_options.txt > @@ -21,7 +21,7 @@ option('tls_priority', type : 'string', value : 'NORMAL', > option('default_devices', type : 'boolean', value : true, > description: 'Include a default selection of devices in emulators') > option('audio_drv_list', type: 'array', value: ['default'], > - choices: ['alsa', 'coreaudio', 'default', 'dsound', 'jack', 'oss', 'pa', 'sdl', 'sndio'], > + choices: ['alsa', 'coreaudio', 'default', 'dsound', 'jack', 'oss', 'pa', 'pipewire', 'sdl', 'sndio'], > description: 'Set audio driver list') > option('block_drv_rw_whitelist', type : 'string', value : '', > description: 'set block driver read-write whitelist (by default affects only QEMU, not tools like qemu-img)') > @@ -255,6 +255,8 @@ option('oss', type: 'feature', value: 'auto', > description: 'OSS sound support') > option('pa', type: 'feature', value: 'auto', > description: 'PulseAudio sound support') > +option('pipewire', type: 'feature', value: 'auto', > + description: 'Pipewire sound support') > option('sndio', type: 'feature', value: 'auto', > description: 'sndio sound support') > > diff --git a/qapi/audio.json b/qapi/audio.json > index 4e54c00f51..e03396a7bc 100644 > --- a/qapi/audio.json > +++ b/qapi/audio.json > @@ -324,6 +324,47 @@ > '*out': 'AudiodevPaPerDirectionOptions', > '*server': 'str' } } > > +## > +# @AudiodevPipewirePerDirectionOptions: > +# > +# Options of the Pipewire backend that are used for both playback and > +# recording. > +# > +# @name: name of the sink/source to use > +# > +# @stream-name: name of the Pipewire stream created by qemu. Can be > +# used to identify the stream in Pipewire when you > +# create multiple Pipewire devices or run multiple qemu > +# instances (default: audiodev's id) > +# > +# @latency: latency you want Pipewire to achieve in microseconds > +# (default 46000) > +# > +# Since: 8.1 > +## > +{ 'struct': 'AudiodevPipewirePerDirectionOptions', > + 'base': 'AudiodevPerDirectionOptions', > + 'data': { > + '*name': 'str', > + '*stream-name': 'str', > + '*latency': 'uint32' } } > + > +## > +# @AudiodevPipewireOptions: > +# > +# Options of the Pipewire audio backend. > +# > +# @in: options of the capture stream > +# > +# @out: options of the playback stream > +# > +# Since: 8.1 > +## > +{ 'struct': 'AudiodevPipewireOptions', > + 'data': { > + '*in': 'AudiodevPipewirePerDirectionOptions', > + '*out': 'AudiodevPipewirePerDirectionOptions' } } > + > ## > # @AudiodevSdlPerDirectionOptions: > # > @@ -416,6 +457,7 @@ > { 'name': 'jack', 'if': 'CONFIG_AUDIO_JACK' }, > { 'name': 'oss', 'if': 'CONFIG_AUDIO_OSS' }, > { 'name': 'pa', 'if': 'CONFIG_AUDIO_PA' }, > + { 'name': 'pipewire', 'if': 'CONFIG_AUDIO_PIPEWIRE' }, > { 'name': 'sdl', 'if': 'CONFIG_AUDIO_SDL' }, > { 'name': 'sndio', 'if': 'CONFIG_AUDIO_SNDIO' }, > { 'name': 'spice', 'if': 'CONFIG_SPICE' }, > @@ -456,6 +498,8 @@ > 'if': 'CONFIG_AUDIO_OSS' }, > 'pa': { 'type': 'AudiodevPaOptions', > 'if': 'CONFIG_AUDIO_PA' }, > + 'pipewire': { 'type': 'AudiodevPipewireOptions', > + 'if': 'CONFIG_AUDIO_PIPEWIRE' }, > 'sdl': { 'type': 'AudiodevSdlOptions', > 'if': 'CONFIG_AUDIO_SDL' }, > 'sndio': { 'type': 'AudiodevSndioOptions', > diff --git a/qemu-options.hx b/qemu-options.hx > index 59bdf67a2c..2d908717bd 100644 > --- a/qemu-options.hx > +++ b/qemu-options.hx > @@ -779,6 +779,12 @@ DEF("audiodev", HAS_ARG, QEMU_OPTION_audiodev, > " in|out.name= source/sink device name\n" > " in|out.latency= desired latency in microseconds\n" > #endif > +#ifdef CONFIG_AUDIO_PIPEWIRE > + "-audiodev pipewire,id=id[,prop[=value][,...]]\n" > + " in|out.name= source/sink device name\n" > + " in|out.stream-name= name of pipewire stream\n" > + " in|out.latency= desired latency in microseconds\n" > +#endif > #ifdef CONFIG_AUDIO_SDL > "-audiodev sdl,id=id[,prop[=value][,...]]\n" > " in|out.buffer-count= number of buffers\n" > @@ -942,6 +948,21 @@ SRST > Desired latency in microseconds. The PulseAudio server will try > to honor this value but actual latencies may be lower or higher. > > +``-audiodev pipewire,id=id[,prop[=value][,...]]`` > + Creates a backend using Pipewire. This backend is available on > + most systems. > + > + Pipewire specific options are: > + > + ``in|out.latency=usecs`` > + Desired latency in microseconds. > + > + ``in|out.name=sink`` > + Use the specified source/sink for recording/playback. > + > + ``in|out.stream-name`` > + Specify the name of pipewire stream. > + > ``-audiodev sdl,id=id[,prop[=value][,...]]`` > Creates a backend using SDL. This backend is available on most > systems, but you should use your platform's native backend if > diff --git a/scripts/meson-buildoptions.sh b/scripts/meson-buildoptions.sh > index 009fab1515..ba1057b62c 100644 > --- a/scripts/meson-buildoptions.sh > +++ b/scripts/meson-buildoptions.sh > @@ -1,7 +1,8 @@ > # This file is generated by meson-buildoptions.py, do not edit! > meson_options_help() { > - printf "%s\n" ' --audio-drv-list=CHOICES Set audio driver list [default] (choices: alsa/co' > - printf "%s\n" ' reaudio/default/dsound/jack/oss/pa/sdl/sndio)' > + printf "%s\n" ' --audio-drv-list=CHOICES Set audio driver list [default] (choices: al' > + printf "%s\n" ' sa/coreaudio/default/dsound/jack/oss/pa/' > + printf "%s\n" ' pipewire/sdl/sndio)' > printf "%s\n" ' --block-drv-ro-whitelist=VALUE' > printf "%s\n" ' set block driver read-only whitelist (by default' > printf "%s\n" ' affects only QEMU, not tools like qemu-img)' > @@ -136,6 +137,7 @@ meson_options_help() { > printf "%s\n" ' oss OSS sound support' > printf "%s\n" ' pa PulseAudio sound support' > printf "%s\n" ' parallels parallels image format support' > + printf "%s\n" ' pipewire Pipewire sound support' > printf "%s\n" ' png PNG support with libpng' > printf "%s\n" ' pvrdma Enable PVRDMA support' > printf "%s\n" ' qcow1 qcow1 image format support' > @@ -370,6 +372,8 @@ _meson_option_parse() { > --disable-pa) printf "%s" -Dpa=disabled ;; > --enable-parallels) printf "%s" -Dparallels=enabled ;; > --disable-parallels) printf "%s" -Dparallels=disabled ;; > + --enable-pipewire) printf "%s" -Dpipewire=enabled ;; > + --disable-pipewire) printf "%s" -Dpipewire=disabled ;; > --with-pkgversion=*) quote_sh "-Dpkgversion=$2" ;; > --enable-png) printf "%s" -Dpng=enabled ;; > --disable-png) printf "%s" -Dpng=disabled ;;
Thank you for the clarification, I will look into it. Regards, Dorinda. On Sat, Apr 15, 2023 at 9:39 AM Volker Rümelin <vr_qemu@t-online.de> wrote: > Hi Dorinda, > > > This commit adds a new audiodev backend to allow QEMU to use Pipewire as > > both an audio sink and source. This backend is available on most systems > > > > Add Pipewire entry points for QEMU Pipewire audio backend > > Add wrappers for QEMU Pipewire audio backend in qpw_pcm_ops() > > qpw_write function returns the current state of the stream to pwaudio > > and Writes some data to the server for playback streams using pipewire > > spa_ringbuffer implementation. > > qpw_read function returns the current state of the stream to pwaudio and > > reads some data from the server for capture streams using pipewire > > spa_ringbuffer implementation. These functions qpw_write and qpw_read > > are called during playback and capture. > > Added some functions that convert pw audio formats to QEMU audio format > > and vice versa which would be needed in the pipewire audio sink and > > source functions qpw_init_in() & qpw_init_out(). > > These methods that implement playback and recording will create streams > > for playback and capture that will start processing and will result in > > the on_process callbacks to be called. > > Built a connection to the Pipewire sound system server in the > > qpw_audio_init() method. > > > > Signed-off-by: Dorinda Bassey <dbassey@redhat.com> > > --- > > v11: > > handle buffer underruns in qpw_write > > use local variable > > change param name frame_size > > fix format specifier > > change trace value to trace quantum > > > > audio/audio.c | 3 + > > audio/audio_template.h | 4 + > > audio/meson.build | 1 + > > audio/pwaudio.c | 913 ++++++++++++++++++++++++++++++++++ > > audio/trace-events | 8 + > > meson.build | 8 + > > meson_options.txt | 4 +- > > qapi/audio.json | 44 ++ > > qemu-options.hx | 21 + > > scripts/meson-buildoptions.sh | 8 +- > > 10 files changed, 1011 insertions(+), 3 deletions(-) > > create mode 100644 audio/pwaudio.c > > > > diff --git a/audio/audio.c b/audio/audio.c > > index 70b096713c..90c7c49d11 100644 > > --- a/audio/audio.c > > +++ b/audio/audio.c > > @@ -2061,6 +2061,9 @@ void audio_create_pdos(Audiodev *dev) > > #ifdef CONFIG_AUDIO_PA > > CASE(PA, pa, Pa); > > #endif > > +#ifdef CONFIG_AUDIO_PIPEWIRE > > + CASE(PIPEWIRE, pipewire, Pipewire); > > +#endif > > #ifdef CONFIG_AUDIO_SDL > > CASE(SDL, sdl, Sdl); > > #endif > > diff --git a/audio/audio_template.h b/audio/audio_template.h > > index e42326c20d..dc0c74aa74 100644 > > --- a/audio/audio_template.h > > +++ b/audio/audio_template.h > > @@ -362,6 +362,10 @@ AudiodevPerDirectionOptions *glue(audio_get_pdo_, > TYPE)(Audiodev *dev) > > case AUDIODEV_DRIVER_PA: > > return qapi_AudiodevPaPerDirectionOptions_base(dev->u.pa.TYPE); > > #endif > > +#ifdef CONFIG_AUDIO_PIPEWIRE > > + case AUDIODEV_DRIVER_PIPEWIRE: > > + return > qapi_AudiodevPipewirePerDirectionOptions_base(dev->u.pipewire.TYPE); > > +#endif > > #ifdef CONFIG_AUDIO_SDL > > case AUDIODEV_DRIVER_SDL: > > return > qapi_AudiodevSdlPerDirectionOptions_base(dev->u.sdl.TYPE); > > diff --git a/audio/meson.build b/audio/meson.build > > index 0722224ba9..65a49c1a10 100644 > > --- a/audio/meson.build > > +++ b/audio/meson.build > > @@ -19,6 +19,7 @@ foreach m : [ > > ['sdl', sdl, files('sdlaudio.c')], > > ['jack', jack, files('jackaudio.c')], > > ['sndio', sndio, files('sndioaudio.c')], > > + ['pipewire', pipewire, files('pwaudio.c')], > > ['spice', spice, files('spiceaudio.c')] > > ] > > if m[1].found() > > diff --git a/audio/pwaudio.c b/audio/pwaudio.c > > new file mode 100644 > > index 0000000000..adf1a538c0 > > --- /dev/null > > +++ b/audio/pwaudio.c > > @@ -0,0 +1,913 @@ > > +/* > > + * QEMU Pipewire audio driver > > + * > > + * Copyright (c) 2023 Red Hat Inc. > > + * > > + * Author: Dorinda Bassey <dbassey@redhat.com> > > + * > > + * SPDX-License-Identifier: GPL-2.0-or-later > > + */ > > + > > +#include "qemu/osdep.h" > > +#include "qemu/module.h" > > +#include "audio.h" > > +#include <errno.h> > > +#include "qemu/error-report.h" > > +#include <spa/param/audio/format-utils.h> > > +#include <spa/utils/ringbuffer.h> > > +#include <spa/utils/result.h> > > +#include <spa/param/props.h> > > + > > +#include <pipewire/pipewire.h> > > +#include "trace.h" > > + > > +#define AUDIO_CAP "pipewire" > > +#define RINGBUFFER_SIZE (1u << 22) > > +#define RINGBUFFER_MASK (RINGBUFFER_SIZE - 1) > > + > > +#include "audio_int.h" > > + > > +typedef struct pwvolume { > > + uint32_t channels; > > + float values[SPA_AUDIO_MAX_CHANNELS]; > > +} pwvolume; > > + > > +typedef struct pwaudio { > > + Audiodev *dev; > > + struct pw_thread_loop *thread_loop; > > + struct pw_context *context; > > + > > + struct pw_core *core; > > + struct spa_hook core_listener; > > + int last_seq, pending_seq, error; > > +} pwaudio; > > + > > +typedef struct PWVoice { > > + pwaudio *g; > > + struct pw_stream *stream; > > + struct spa_hook stream_listener; > > + struct spa_audio_info_raw info; > > + uint32_t highwater_mark; > > + uint32_t sample_size, req; > > The older patches used the correct name frame_size instead of > sample_size. Please revert this. Further below I've written an explanation. > > > + struct spa_ringbuffer ring; > > + uint8_t buffer[RINGBUFFER_SIZE]; > > + > > + pwvolume volume; > > + bool muted; > > +} PWVoice; > > + > > +typedef struct PWVoiceOut { > > + HWVoiceOut hw; > > + PWVoice v; > > +} PWVoiceOut; > > + > > +typedef struct PWVoiceIn { > > + HWVoiceIn hw; > > + PWVoice v; > > +} PWVoiceIn; > > + > > +static void > > +stream_destroy(void *data) > > +{ > > + PWVoice *v = (PWVoice *) data; > > + spa_hook_remove(&v->stream_listener); > > + v->stream = NULL; > > +} > > + > > +/* output data processing function to read stuffs from the buffer */ > > +static void > > +playback_on_process(void *data) > > +{ > > + PWVoice *v = (PWVoice *) data; > > + void *p; > > + struct pw_buffer *b; > > + struct spa_buffer *buf; > > + uint32_t req, index, n_bytes; > > + int32_t avail; > > + > > + assert(v->stream); > > + > > + /* obtain a buffer to read from */ > > + b = pw_stream_dequeue_buffer(v->stream); > > + if (b == NULL) { > > + error_report("out of buffers: %s", strerror(errno)); > > + return; > > + } > > + > > + buf = b->buffer; > > + p = buf->datas[0].data; > > + if (p == NULL) { > > + return; > > + } > > + /* calculate the total no of bytes to read data from buffer */ > > + req = b->requested * v->sample_size; > > + if (req == 0) { > > + req = v->req; > > + } > > + n_bytes = SPA_MIN(req, buf->datas[0].maxsize); > > + > > + /* get no of available bytes to read data from buffer */ > > + > > + avail = spa_ringbuffer_get_read_index(&v->ring, &index); > > + > > + if (avail <= 0) { > > + /* underrun, can't really happen but if it does we */ > > + /* do nothing and wait for more data */ > > + error_report("%p: underrun read:%u avail:%d", p, index, avail); > > Please don't do that. The PipeWire audio threads have a higher priority > than the QEMU thread. On a heavily loaded system the callbacks will be > called even if QEMU nearly stalled. So this is the right place to detect > a buffer underflow and continue to write silent audio frames. > > There's no need for an error report. Buffer underflows are expected and > the QEMU users can hear the problem. > > > + } 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->sample_size; > > + buf->datas[0].chunk->size = n_bytes; > > + > > + /* queue the buffer for playback */ > > + pw_stream_queue_buffer(v->stream, b); > > +} > > + > > +/* output data processing function to generate stuffs in the buffer */ > > +static void > > +capture_on_process(void *data) > > +{ > > + PWVoice *v = (PWVoice *) data; > > + void *p; > > + struct pw_buffer *b; > > + struct spa_buffer *buf; > > + int32_t filled; > > + uint32_t index, offs, n_bytes; > > + > > + assert(v->stream); > > + > > + /* obtain a buffer */ > > + b = pw_stream_dequeue_buffer(v->stream); > > + if (b == NULL) { > > + error_report("out of buffers: %s", strerror(errno)); > > + return; > > + } > > + > > + /* Write data into buffer */ > > + buf = b->buffer; > > + p = buf->datas[0].data; > > + if (p == NULL) { > > + return; > > + } > > + offs = SPA_MIN(buf->datas[0].chunk->offset, buf->datas[0].maxsize); > > + n_bytes = SPA_MIN(buf->datas[0].chunk->size, buf->datas[0].maxsize > - offs); > > + > > + filled = spa_ringbuffer_get_write_index(&v->ring, &index); > > + > > + > > + if (filled < 0) { > > + error_report("%p: underrun write:%u filled:%d", p, index, > filled); > > + } else { > > + if ((uint32_t) filled + n_bytes > RINGBUFFER_SIZE) { > > + error_report("%p: overrun write:%u filled:%d + size:%u > > max:%u", > > + p, index, filled, n_bytes, RINGBUFFER_SIZE); > > + } > > + } > > + spa_ringbuffer_write_data(&v->ring, > > + v->buffer, RINGBUFFER_SIZE, > > + index & RINGBUFFER_MASK, > > + SPA_PTROFF(p, offs, void), n_bytes); > > + index += n_bytes; > > + spa_ringbuffer_write_update(&v->ring, index); > > + > > + /* queue the buffer for playback */ > > + pw_stream_queue_buffer(v->stream, b); > > +} > > + > > +static void > > +on_stream_state_changed(void *data, enum pw_stream_state old, > > + enum pw_stream_state state, const char *error) > > +{ > > + PWVoice *v = (PWVoice *) data; > > + > > + trace_pw_state_changed(pw_stream_get_node_id(v->stream), > > + pw_stream_state_as_string(state)); > > + > > + switch (state) { > > + case PW_STREAM_STATE_ERROR: > > + case PW_STREAM_STATE_UNCONNECTED: > > + break; > > + case PW_STREAM_STATE_PAUSED: > > + case PW_STREAM_STATE_CONNECTING: > > + case PW_STREAM_STATE_STREAMING: > > + break; > > + } > > +} > > + > > +static const struct pw_stream_events capture_stream_events = { > > + PW_VERSION_STREAM_EVENTS, > > + .destroy = stream_destroy, > > + .state_changed = on_stream_state_changed, > > + .process = capture_on_process > > +}; > > + > > +static const struct pw_stream_events playback_stream_events = { > > + PW_VERSION_STREAM_EVENTS, > > + .destroy = stream_destroy, > > + .state_changed = on_stream_state_changed, > > + .process = playback_on_process > > +}; > > + > > +static size_t > > +qpw_read(HWVoiceIn *hw, void *data, size_t len) > > +{ > > + PWVoiceIn *pw = (PWVoiceIn *) hw; > > + PWVoice *v = &pw->v; > > + pwaudio *c = v->g; > > + const char *error = NULL; > > + size_t l; > > + int32_t avail; > > + uint32_t index; > > + > > + pw_thread_loop_lock(c->thread_loop); > > + if (pw_stream_get_state(v->stream, &error) != > PW_STREAM_STATE_STREAMING) { > > + /* wait for stream to become ready */ > > + l = 0; > > + goto done_unlock; > > + } > > + /* get no of available bytes to read data from buffer */ > > + avail = spa_ringbuffer_get_read_index(&v->ring, &index); > > + > > + trace_pw_read(avail, index, len); > > + > > + if (avail < (int32_t) len) { > > + len = avail; > > + } > > + > > + spa_ringbuffer_read_data(&v->ring, > > + v->buffer, RINGBUFFER_SIZE, > > + index & RINGBUFFER_MASK, data, len); > > + index += len; > > + spa_ringbuffer_read_update(&v->ring, index); > > + l = len; > > + > > +done_unlock: > > + pw_thread_loop_unlock(c->thread_loop); > > + return l; > > +} > > + > > +static size_t qpw_buffer_get_free(HWVoiceOut *hw) > > +{ > > + PWVoiceOut *pw = (PWVoiceOut *)hw; > > + PWVoice *v = &pw->v; > > + pwaudio *c = v->g; > > + const char *error = NULL; > > + int32_t filled, avail; > > + uint32_t index; > > + > > + pw_thread_loop_lock(c->thread_loop); > > + if (pw_stream_get_state(v->stream, &error) != > PW_STREAM_STATE_STREAMING) { > > + /* wait for stream to become ready */ > > + avail = 0; > > + goto done_unlock; > > + } > > + > > + filled = spa_ringbuffer_get_write_index(&v->ring, &index); > > + avail = v->highwater_mark - filled; > > + > > +done_unlock: > > + pw_thread_loop_unlock(c->thread_loop); > > + return avail; > > +} > > + > > +static size_t > > +qpw_write(HWVoiceOut *hw, void *data, size_t len) > > +{ > > + PWVoiceOut *pw = (PWVoiceOut *) hw; > > + PWVoice *v = &pw->v; > > + pwaudio *c = v->g; > > + const char *error = NULL; > > + int32_t filled, avail; > > + uint32_t index; > > + > > + pw_thread_loop_lock(c->thread_loop); > > + if (pw_stream_get_state(v->stream, &error) != > PW_STREAM_STATE_STREAMING) { > > + /* wait for stream to become ready */ > > + len = 0; > > + goto done_unlock; > > + } > > + filled = spa_ringbuffer_get_write_index(&v->ring, &index); > > + avail = v->highwater_mark - filled; > > + > > + trace_pw_write(filled, avail, index, len); > > + > > + if (len > avail) { > > + len = avail; > > + } > > + > > + if (filled < 0) { > > + audio_pcm_info_clear_buf(&hw->info, data, len / > hw->info.bytes_per_frame); > > - audio_pcm_info_clear_buf(&hw->info, data, len / > hw->info.bytes_per_frame); > > There is no way to reach this code. > > > + error_report("%p: underrun write:%u filled:%d", pw, index, > filled); > > + } else { > > + if ((uint32_t) filled + len > RINGBUFFER_SIZE) { > > + error_report("%p: overrun write:%u filled:%d + size:%zu > > max:%u", > > + pw, index, filled, len, RINGBUFFER_SIZE); > > + } > > + } > > + > > + spa_ringbuffer_write_data(&v->ring, > > + v->buffer, RINGBUFFER_SIZE, > > + index & RINGBUFFER_MASK, data, len); > > + index += len; > > + spa_ringbuffer_write_update(&v->ring, index); > > + > > +done_unlock: > > + pw_thread_loop_unlock(c->thread_loop); > > + return len; > > +} > > + > > +static int > > +audfmt_to_pw(AudioFormat fmt, int endianness) > > +{ > > + int format; > > + > > + switch (fmt) { > > + case AUDIO_FORMAT_S8: > > + format = SPA_AUDIO_FORMAT_S8; > > + break; > > + case AUDIO_FORMAT_U8: > > + format = SPA_AUDIO_FORMAT_U8; > > + break; > > + case AUDIO_FORMAT_S16: > > + format = endianness ? SPA_AUDIO_FORMAT_S16_BE : > SPA_AUDIO_FORMAT_S16_LE; > > + break; > > + case AUDIO_FORMAT_U16: > > + format = endianness ? SPA_AUDIO_FORMAT_U16_BE : > SPA_AUDIO_FORMAT_U16_LE; > > + break; > > + case AUDIO_FORMAT_S32: > > + format = endianness ? SPA_AUDIO_FORMAT_S32_BE : > SPA_AUDIO_FORMAT_S32_LE; > > + break; > > + case AUDIO_FORMAT_U32: > > + format = endianness ? SPA_AUDIO_FORMAT_U32_BE : > SPA_AUDIO_FORMAT_U32_LE; > > + break; > > + case AUDIO_FORMAT_F32: > > + format = endianness ? SPA_AUDIO_FORMAT_F32_BE : > SPA_AUDIO_FORMAT_F32_LE; > > + break; > > + default: > > + dolog("Internal logic error: Bad audio format %d\n", fmt); > > + format = SPA_AUDIO_FORMAT_U8; > > + break; > > + } > > + return format; > > +} > > + > > +static AudioFormat > > +pw_to_audfmt(enum spa_audio_format fmt, int *endianness, > > + uint32_t *sample_size) > > +{ > > + switch (fmt) { > > + case SPA_AUDIO_FORMAT_S8: > > + *sample_size = 1; > > + return AUDIO_FORMAT_S8; > > + case SPA_AUDIO_FORMAT_U8: > > + *sample_size = 1; > > + return AUDIO_FORMAT_U8; > > + case SPA_AUDIO_FORMAT_S16_BE: > > + *sample_size = 2; > > + *endianness = 1; > > + return AUDIO_FORMAT_S16; > > + case SPA_AUDIO_FORMAT_S16_LE: > > + *sample_size = 2; > > + *endianness = 0; > > + return AUDIO_FORMAT_S16; > > + case SPA_AUDIO_FORMAT_U16_BE: > > + *sample_size = 2; > > + *endianness = 1; > > + return AUDIO_FORMAT_U16; > > + case SPA_AUDIO_FORMAT_U16_LE: > > + *sample_size = 2; > > + *endianness = 0; > > + return AUDIO_FORMAT_U16; > > + case SPA_AUDIO_FORMAT_S32_BE: > > + *sample_size = 4; > > + *endianness = 1; > > + return AUDIO_FORMAT_S32; > > + case SPA_AUDIO_FORMAT_S32_LE: > > + *sample_size = 4; > > + *endianness = 0; > > + return AUDIO_FORMAT_S32; > > + case SPA_AUDIO_FORMAT_U32_BE: > > + *sample_size = 4; > > + *endianness = 1; > > + return AUDIO_FORMAT_U32; > > + case SPA_AUDIO_FORMAT_U32_LE: > > + *sample_size = 4; > > + *endianness = 0; > > + return AUDIO_FORMAT_U32; > > + case SPA_AUDIO_FORMAT_F32_BE: > > + *sample_size = 4; > > + *endianness = 1; > > + return AUDIO_FORMAT_F32; > > + case SPA_AUDIO_FORMAT_F32_LE: > > + *sample_size = 4; > > + *endianness = 0; > > + return AUDIO_FORMAT_F32; > > + default: > > + *sample_size = 1; > > + dolog("Internal logic error: Bad spa_audio_format %d\n", fmt); > > + return AUDIO_FORMAT_U8; > > + } > > +} > > + > > +static int > > +create_stream(pwaudio *c, PWVoice *v, const char *stream_name, > > + const char *name, enum spa_direction dir) > > +{ > > + int res; > > + uint32_t n_params; > > + const struct spa_pod *params[2]; > > + uint8_t buffer[1024]; > > + struct spa_pod_builder b; > > + uint64_t buf_samples; > > + struct pw_properties *props; > > + > > + props = pw_properties_new(NULL, NULL); > > + > > + /* 75% of the timer period for faster updates */ > > + buf_samples = (uint64_t)v->g->dev->timer_period * v->info.rate > > + * 3 / 4 / 1000000; > > + trace_pw_timer(v->g->dev->timer_period); > > + pw_properties_setf(props, PW_KEY_NODE_LATENCY, "%" PRIu64 "/%u", > > + buf_samples, v->info.rate); > > + > > + if (name) { > > + pw_properties_set(props, PW_KEY_TARGET_OBJECT, name); > > + } > > + v->stream = pw_stream_new(c->core, stream_name, props); > > + > > + if (v->stream == NULL) { > > + return -1; > > + } > > + > > + if (dir == SPA_DIRECTION_INPUT) { > > + pw_stream_add_listener(v->stream, > > + &v->stream_listener, > &capture_stream_events, v); > > + } else { > > + pw_stream_add_listener(v->stream, > > + &v->stream_listener, > &playback_stream_events, v); > > + } > > + > > + n_params = 0; > > + spa_pod_builder_init(&b, buffer, sizeof(buffer)); > > + params[n_params++] = spa_format_audio_raw_build(&b, > > + SPA_PARAM_EnumFormat, > > + &v->info); > > + > > + /* connect the stream to a sink or source */ > > + res = pw_stream_connect(v->stream, > > + dir == > > + SPA_DIRECTION_INPUT ? PW_DIRECTION_INPUT : > > + PW_DIRECTION_OUTPUT, PW_ID_ANY, > > + PW_STREAM_FLAG_AUTOCONNECT | > > + PW_STREAM_FLAG_INACTIVE | > > + PW_STREAM_FLAG_MAP_BUFFERS | > > + PW_STREAM_FLAG_RT_PROCESS, params, > n_params); > > + if (res < 0) { > > + pw_stream_destroy(v->stream); > > + return -1; > > + } > > + > > + return 0; > > +} > > + > > +static int > > +qpw_stream_new(pwaudio *c, PWVoice *v, const char *stream_name, > > + const char *name, enum spa_direction dir) > > +{ > > + int r; > > + > > + switch (v->info.channels) { > > + case 8: > > + v->info.position[0] = SPA_AUDIO_CHANNEL_FL; > > + v->info.position[1] = SPA_AUDIO_CHANNEL_FR; > > + v->info.position[2] = SPA_AUDIO_CHANNEL_FC; > > + v->info.position[3] = SPA_AUDIO_CHANNEL_LFE; > > + v->info.position[4] = SPA_AUDIO_CHANNEL_RL; > > + v->info.position[5] = SPA_AUDIO_CHANNEL_RR; > > + v->info.position[6] = SPA_AUDIO_CHANNEL_SL; > > + v->info.position[7] = SPA_AUDIO_CHANNEL_SR; > > + break; > > + case 6: > > + v->info.position[0] = SPA_AUDIO_CHANNEL_FL; > > + v->info.position[1] = SPA_AUDIO_CHANNEL_FR; > > + v->info.position[2] = SPA_AUDIO_CHANNEL_FC; > > + v->info.position[3] = SPA_AUDIO_CHANNEL_LFE; > > + v->info.position[4] = SPA_AUDIO_CHANNEL_RL; > > + v->info.position[5] = SPA_AUDIO_CHANNEL_RR; > > + break; > > + case 5: > > + v->info.position[0] = SPA_AUDIO_CHANNEL_FL; > > + v->info.position[1] = SPA_AUDIO_CHANNEL_FR; > > + v->info.position[2] = SPA_AUDIO_CHANNEL_FC; > > + v->info.position[3] = SPA_AUDIO_CHANNEL_LFE; > > + v->info.position[4] = SPA_AUDIO_CHANNEL_RC; > > + break; > > + case 4: > > + v->info.position[0] = SPA_AUDIO_CHANNEL_FL; > > + v->info.position[1] = SPA_AUDIO_CHANNEL_FR; > > + v->info.position[2] = SPA_AUDIO_CHANNEL_FC; > > + v->info.position[3] = SPA_AUDIO_CHANNEL_RC; > > + break; > > + case 3: > > + v->info.position[0] = SPA_AUDIO_CHANNEL_FL; > > + v->info.position[1] = SPA_AUDIO_CHANNEL_FR; > > + v->info.position[2] = SPA_AUDIO_CHANNEL_LFE; > > + break; > > + case 2: > > + v->info.position[0] = SPA_AUDIO_CHANNEL_FL; > > + v->info.position[1] = SPA_AUDIO_CHANNEL_FR; > > + break; > > + case 1: > > + v->info.position[0] = SPA_AUDIO_CHANNEL_MONO; > > + break; > > + default: > > + for (size_t i = 0; i < v->info.channels; i++) { > > + v->info.position[i] = SPA_AUDIO_CHANNEL_UNKNOWN; > > + } > > + break; > > + } > > + > > + /* create a new unconnected pwstream */ > > + r = create_stream(c, v, stream_name, name, dir); > > + if (r < 0) { > > + AUD_log(AUDIO_CAP, "Failed to create stream."); > > + return -1; > > + } > > + > > + return r; > > +} > > + > > +static int > > +qpw_init_out(HWVoiceOut *hw, struct audsettings *as, void *drv_opaque) > > +{ > > + PWVoiceOut *pw = (PWVoiceOut *) hw; > > + PWVoice *v = &pw->v; > > + struct audsettings obt_as = *as; > > + pwaudio *c = v->g = drv_opaque; > > + AudiodevPipewireOptions *popts = &c->dev->u.pipewire; > > + AudiodevPipewirePerDirectionOptions *ppdo = popts->out; > > + int r; > > + > > + pw_thread_loop_lock(c->thread_loop); > > + > > + v->info.format = audfmt_to_pw(as->fmt, as->endianness); > > + v->info.channels = as->nchannels; > > + v->info.rate = as->freq; > > + > > + obt_as.fmt = > > + pw_to_audfmt(v->info.format, &obt_as.endianness, > &v->sample_size); > > The third argument of pw_to_audfmt() returns the sample size. > > > + v->sample_size *= as->nchannels; > > Here you calculate the frame size from the sample size. The correct name > is v->frame_size. I'm aware the rest of QEMU quite often uses samples as > a synonym for frames. But new code should get the variable names right. > > > + > > + v->req = (uint64_t)c->dev->timer_period * v->info.rate > > + * 1 / 2 / 1000000 * v->sample_size; > > + > > + /* call the function that creates a new stream for playback */ > > + r = qpw_stream_new(c, v, ppdo->stream_name ? : c->dev->id, > > + ppdo->name, SPA_DIRECTION_OUTPUT); > > + if (r < 0) { > > + error_report("qpw_stream_new for playback failed"); > > + pw_thread_loop_unlock(c->thread_loop); > > + return -1; > > + } > > + > > + /* report the audio format we support */ > > + audio_pcm_init_info(&hw->info, &obt_as); > > + > > + /* report the buffer size to qemu */ > > + hw->samples = audio_buffer_frames( > > + qapi_AudiodevPipewirePerDirectionOptions_base(ppdo), &obt_as, > 46440); > > + v->highwater_mark = MIN(RINGBUFFER_SIZE, > > + (ppdo->has_latency ? ppdo->latency : 46440) > > + * (uint64_t)v->info.rate / 1000000 * > v->sample_size); > > + > > + pw_thread_loop_unlock(c->thread_loop); > > + return 0; > > +} > > + > > +static int > > +qpw_init_in(HWVoiceIn *hw, struct audsettings *as, void *drv_opaque) > > +{ > > + PWVoiceIn *pw = (PWVoiceIn *) hw; > > + PWVoice *v = &pw->v; > > + struct audsettings obt_as = *as; > > + pwaudio *c = v->g = drv_opaque; > > + AudiodevPipewireOptions *popts = &c->dev->u.pipewire; > > + AudiodevPipewirePerDirectionOptions *ppdo = popts->in; > > + int r; > > + > > + pw_thread_loop_lock(c->thread_loop); > > + > > + v->info.format = audfmt_to_pw(as->fmt, as->endianness); > > + v->info.channels = as->nchannels; > > + v->info.rate = as->freq; > > + > > + obt_as.fmt = > > + pw_to_audfmt(v->info.format, &obt_as.endianness, > &v->sample_size); > > + v->sample_size *= as->nchannels; > > + > > + /* call the function that creates a new stream for recording */ > > + r = qpw_stream_new(c, v, ppdo->stream_name ? : c->dev->id, > > + ppdo->name, SPA_DIRECTION_INPUT); > > + if (r < 0) { > > + error_report("qpw_stream_new for recording failed"); > > + pw_thread_loop_unlock(c->thread_loop); > > + return -1; > > + } > > + > > + /* report the audio format we support */ > > + audio_pcm_init_info(&hw->info, &obt_as); > > + > > + /* report the buffer size to qemu */ > > + hw->samples = audio_buffer_frames( > > + qapi_AudiodevPipewirePerDirectionOptions_base(ppdo), &obt_as, > 46440); > > + > > + pw_thread_loop_unlock(c->thread_loop); > > + return 0; > > +} > > + > > +static void > > +qpw_fini_out(HWVoiceOut *hw) > > +{ > > + PWVoiceOut *pw = (PWVoiceOut *) hw; > > + PWVoice *v = &pw->v; > > + > > + if (v->stream) { > > + pwaudio *c = v->g; > > + pw_thread_loop_lock(c->thread_loop); > > + pw_stream_destroy(v->stream); > > + v->stream = NULL; > > + pw_thread_loop_unlock(c->thread_loop); > > + } > > +} > > + > > +static void > > +qpw_fini_in(HWVoiceIn *hw) > > +{ > > + PWVoiceIn *pw = (PWVoiceIn *) hw; > > + PWVoice *v = &pw->v; > > + > > + if (v->stream) { > > + pwaudio *c = v->g; > > + pw_thread_loop_lock(c->thread_loop); > > + pw_stream_destroy(v->stream); > > + v->stream = NULL; > > + pw_thread_loop_unlock(c->thread_loop); > > + } > > +} > > + > > +static void > > +qpw_enable_out(HWVoiceOut *hw, bool enable) > > +{ > > + PWVoiceOut *po = (PWVoiceOut *) hw; > > + PWVoice *v = &po->v; > > + pwaudio *c = v->g; > > + pw_thread_loop_lock(c->thread_loop); > > + pw_stream_set_active(v->stream, enable); > > + pw_thread_loop_unlock(c->thread_loop); > > +} > > + > > +static void > > +qpw_enable_in(HWVoiceIn *hw, bool enable) > > +{ > > + PWVoiceIn *pi = (PWVoiceIn *) hw; > > + PWVoice *v = &pi->v; > > + pwaudio *c = v->g; > > + pw_thread_loop_lock(c->thread_loop); > > + pw_stream_set_active(v->stream, enable); > > + pw_thread_loop_unlock(c->thread_loop); > > +} > > + > > +static void > > +qpw_volume_out(HWVoiceOut *hw, Volume *vol) > > +{ > > + PWVoiceOut *pw = (PWVoiceOut *) hw; > > + PWVoice *v = &pw->v; > > + pwaudio *c = v->g; > > + int i, ret; > > + > > + pw_thread_loop_lock(c->thread_loop); > > + v->volume.channels = vol->channels; > > + > > + for (i = 0; i < vol->channels; ++i) { > > + v->volume.values[i] = (float)vol->vol[i] / 255; > > + } > > + > > + ret = pw_stream_set_control(v->stream, > > + SPA_PROP_channelVolumes, v->volume.channels, v->volume.values, > 0); > > + trace_pw_vol(ret == 0 ? "success" : "failed"); > > + > > + v->muted = vol->mute; > > + float val = v->muted ? 1.f : 0.f; > > + ret = pw_stream_set_control(v->stream, SPA_PROP_mute, 1, &val, 0); > > + pw_thread_loop_unlock(c->thread_loop); > > +} > > + > > +static void > > +qpw_volume_in(HWVoiceIn *hw, Volume *vol) > > +{ > > + PWVoiceIn *pw = (PWVoiceIn *) hw; > > + PWVoice *v = &pw->v; > > + pwaudio *c = v->g; > > + int i, ret; > > + > > + pw_thread_loop_lock(c->thread_loop); > > + v->volume.channels = vol->channels; > > + > > + for (i = 0; i < vol->channels; ++i) { > > + v->volume.values[i] = (float)vol->vol[i] / 255; > > + } > > + > > + ret = pw_stream_set_control(v->stream, > > + SPA_PROP_channelVolumes, v->volume.channels, v->volume.values, > 0); > > + trace_pw_vol(ret == 0 ? "success" : "failed"); > > + > > + v->muted = vol->mute; > > + float val = v->muted ? 1.f : 0.f; > > + ret = pw_stream_set_control(v->stream, SPA_PROP_mute, 1, &val, 0); > > + pw_thread_loop_unlock(c->thread_loop); > > +} > > + > > +static int wait_resync(pwaudio *pw) > > +{ > > + int res; > > + pw->pending_seq = pw_core_sync(pw->core, PW_ID_CORE, > pw->pending_seq); > > + > > + while (true) { > > + pw_thread_loop_wait(pw->thread_loop); > > + > > + res = pw->error; > > + if (res < 0) { > > + pw->error = 0; > > + return res; > > + } > > + if (pw->pending_seq == pw->last_seq) { > > + break; > > + } > > + } > > + return 0; > > +} > > +static void > > +on_core_error(void *data, uint32_t id, int seq, int res, const char > *message) > > +{ > > + pwaudio *pw = data; > > + > > + error_report("error id:%u seq:%d res:%d (%s): %s", > > + id, seq, res, spa_strerror(res), message); > > + > > + /* stop and exit the thread loop */ > > + pw_thread_loop_signal(pw->thread_loop, FALSE); > > +} > > + > > +static void > > +on_core_done(void *data, uint32_t id, int seq) > > +{ > > + pwaudio *pw = data; > > + assert(id == PW_ID_CORE); > > + pw->last_seq = seq; > > + if (pw->pending_seq == seq) { > > + /* stop and exit the thread loop */ > > + pw_thread_loop_signal(pw->thread_loop, FALSE); > > + } > > +} > > + > > +static const struct pw_core_events core_events = { > > + PW_VERSION_CORE_EVENTS, > > + .done = on_core_done, > > + .error = on_core_error, > > +}; > > + > > +static void * > > +qpw_audio_init(Audiodev *dev) > > +{ > > + g_autofree pwaudio *pw = g_new0(pwaudio, 1); > > + pw_init(NULL, NULL); > > + > > + trace_pw_audio_init(); > > + assert(dev->driver == AUDIODEV_DRIVER_PIPEWIRE); > > + > > + pw->dev = dev; > > + pw->thread_loop = pw_thread_loop_new("Pipewire thread loop", NULL); > > + if (pw->thread_loop == NULL) { > > + error_report("Could not create Pipewire loop"); > > + goto fail; > > + } > > + > > + pw->context = > > + pw_context_new(pw_thread_loop_get_loop(pw->thread_loop), NULL, > 0); > > + if (pw->context == NULL) { > > + error_report("Could not create Pipewire context"); > > + goto fail; > > + } > > + > > + if (pw_thread_loop_start(pw->thread_loop) < 0) { > > + error_report("Could not start Pipewire loop"); > > + goto fail; > > + } > > + > > + pw_thread_loop_lock(pw->thread_loop); > > + > > + pw->core = pw_context_connect(pw->context, NULL, 0); > > + if (pw->core == NULL) { > > + pw_thread_loop_unlock(pw->thread_loop); > > + goto fail; > > + } > > + > > + if (pw_core_add_listener(pw->core, &pw->core_listener, > > + &core_events, pw) < 0) { > > + pw_thread_loop_unlock(pw->thread_loop); > > + goto fail; > > + } > > + if (wait_resync(pw) < 0) { > > + pw_thread_loop_unlock(pw->thread_loop); > > + } > > + > > + pw_thread_loop_unlock(pw->thread_loop); > > + > > + return g_steal_pointer(&pw); > > + > > +fail: > > + AUD_log(AUDIO_CAP, "Failed to initialize PW context"); > > + if (pw->thread_loop) { > > + pw_thread_loop_stop(pw->thread_loop); > > + } > > + if (pw->context) { > > + g_clear_pointer(&pw->context, pw_context_destroy); > > + } > > + if (pw->thread_loop) { > > + g_clear_pointer(&pw->thread_loop, pw_thread_loop_destroy); > > + } > > + return NULL; > > +} > > + > > +static void > > +qpw_audio_fini(void *opaque) > > +{ > > + pwaudio *pw = opaque; > > + > > + if (pw->thread_loop) { > > + pw_thread_loop_stop(pw->thread_loop); > > + } > > + > > + if (pw->core) { > > + spa_hook_remove(&pw->core_listener); > > + spa_zero(pw->core_listener); > > + pw_core_disconnect(pw->core); > > + } > > + > > + if (pw->context) { > > + pw_context_destroy(pw->context); > > + } > > + pw_thread_loop_destroy(pw->thread_loop); > > + > > + g_free(pw); > > +} > > + > > +static struct audio_pcm_ops qpw_pcm_ops = { > > + .init_out = qpw_init_out, > > + .fini_out = qpw_fini_out, > > + .write = qpw_write, > > + .buffer_get_free = qpw_buffer_get_free, > > + .run_buffer_out = audio_generic_run_buffer_out, > > + .enable_out = qpw_enable_out, > > + .volume_out = qpw_volume_out, > > + .volume_in = qpw_volume_in, > > + > > + .init_in = qpw_init_in, > > + .fini_in = qpw_fini_in, > > + .read = qpw_read, > > + .run_buffer_in = audio_generic_run_buffer_in, > > + .enable_in = qpw_enable_in > > +}; > > + > > +static struct audio_driver pw_audio_driver = { > > + .name = "pipewire", > > + .descr = "http://www.pipewire.org/", > > + .init = qpw_audio_init, > > + .fini = qpw_audio_fini, > > + .pcm_ops = &qpw_pcm_ops, > > + .can_be_default = 1, > > + .max_voices_out = INT_MAX, > > + .max_voices_in = INT_MAX, > > + .voice_size_out = sizeof(PWVoiceOut), > > + .voice_size_in = sizeof(PWVoiceIn), > > +}; > > + > > +static void > > +register_audio_pw(void) > > +{ > > + audio_driver_register(&pw_audio_driver); > > +} > > + > > +type_init(register_audio_pw); > > diff --git a/audio/trace-events b/audio/trace-events > > index e1ab643add..c764e5641b 100644 > > --- a/audio/trace-events > > +++ b/audio/trace-events > > @@ -18,6 +18,14 @@ dbus_audio_register(const char *s, const char *dir) > "sender = %s, dir = %s" > > dbus_audio_put_buffer_out(size_t len) "len = %zu" > > dbus_audio_read(size_t len) "len = %zu" > > > > +# pwaudio.c > > +pw_state_changed(int nodeid, const char *s) "node id: %d stream state: > %s" > > +pw_read(int32_t avail, uint32_t index, size_t len) "avail=%d index=%u > len=%zu" > > +pw_write(int32_t filled, int32_t avail, uint32_t index, size_t len) > "filled=%d avail=%d index=%u len=%zu" > > +pw_vol(const char *ret) "set volume: %s" > > +pw_timer(uint64_t buf_samples) "timer period = %" PRIu64 > > Sorry, I was not very clear last time. I wrote 'quantum' but I meant the > PipeWire scheduling period. > > -pw_timer(uint64_t buf_samples) "timer period = %" PRIu64 > +pw_period(uint64_t quant, uint32_t rate) "period=%" PRIu64 "/%u" > > This is the same you see with pw-top. > > S ID QUANT RATE WAIT BUSY W/Q B/Q ERR FORMAT NAME > S 28 0 0 --- --- --- --- 0 > Dummy-Driver > S 29 0 0 --- --- --- --- 0 > Freewheel-Driver > S 37 0 0 --- --- --- --- 0 > Midi-Bridge > S 46 0 0 --- --- --- --- 0 > alsa_output.pci-0000_00_03.0.hdmi-stereo > R 47 256 48000 102,2us 22,1us 0,02 0,00 1 S32LE 2 48000 > alsa_output.pci-0000_00_1b.0.analog-stereo > R 63 240 32000 65,4us 4,9us 0,01 0,00 1 U8 1 32000 > + qemu-system-x86_64 > R 67 330 44100 32,8us 33,4us 0,01 0,01 1 S16LE 2 44100 > + qemu-system-x86_64 > S 48 0 0 --- --- --- --- 0 > alsa_input.pci-0000_00_1b.0.analog-stereo > S 68 0 0 --- --- --- --- 0 > qemu-system-x86_64 > > With best regards, > Volker > > > +pw_audio_init(void) "Initialize Pipewire context" > > + > > # audio.c > > audio_timer_start(int interval) "interval %d ms" > > audio_timer_stop(void) "" > > diff --git a/meson.build b/meson.build > > index 29f8644d6d..31bf280c0d 100644 > > --- a/meson.build > > +++ b/meson.build > > @@ -730,6 +730,12 @@ if not get_option('jack').auto() or have_system > > jack = dependency('jack', required: get_option('jack'), > > method: 'pkg-config', kwargs: static_kwargs) > > endif > > +pipewire = not_found > > +if not get_option('pipewire').auto() or (targetos == 'linux' and > have_system) > > + pipewire = dependency('libpipewire-0.3', version: '>=0.3.60', > > + required: get_option('pipewire'), > > + method: 'pkg-config', kwargs: static_kwargs) > > +endif > > sndio = not_found > > if not get_option('sndio').auto() or have_system > > sndio = dependency('sndio', required: get_option('sndio'), > > @@ -1667,6 +1673,7 @@ if have_system > > 'jack': jack.found(), > > 'oss': oss.found(), > > 'pa': pulse.found(), > > + 'pipewire': pipewire.found(), > > 'sdl': sdl.found(), > > 'sndio': sndio.found(), > > } > > @@ -3980,6 +3987,7 @@ if targetos == 'linux' > > summary_info += {'ALSA support': alsa} > > summary_info += {'PulseAudio support': pulse} > > endif > > +summary_info += {'Pipewire support': pipewire} > > summary_info += {'JACK support': jack} > > summary_info += {'brlapi support': brlapi} > > summary_info += {'vde support': vde} > > diff --git a/meson_options.txt b/meson_options.txt > > index fc9447d267..9ae1ec7f47 100644 > > --- a/meson_options.txt > > +++ b/meson_options.txt > > @@ -21,7 +21,7 @@ option('tls_priority', type : 'string', value : > 'NORMAL', > > option('default_devices', type : 'boolean', value : true, > > description: 'Include a default selection of devices in > emulators') > > option('audio_drv_list', type: 'array', value: ['default'], > > - choices: ['alsa', 'coreaudio', 'default', 'dsound', 'jack', > 'oss', 'pa', 'sdl', 'sndio'], > > + choices: ['alsa', 'coreaudio', 'default', 'dsound', 'jack', > 'oss', 'pa', 'pipewire', 'sdl', 'sndio'], > > description: 'Set audio driver list') > > option('block_drv_rw_whitelist', type : 'string', value : '', > > description: 'set block driver read-write whitelist (by default > affects only QEMU, not tools like qemu-img)') > > @@ -255,6 +255,8 @@ option('oss', type: 'feature', value: 'auto', > > description: 'OSS sound support') > > option('pa', type: 'feature', value: 'auto', > > description: 'PulseAudio sound support') > > +option('pipewire', type: 'feature', value: 'auto', > > + description: 'Pipewire sound support') > > option('sndio', type: 'feature', value: 'auto', > > description: 'sndio sound support') > > > > diff --git a/qapi/audio.json b/qapi/audio.json > > index 4e54c00f51..e03396a7bc 100644 > > --- a/qapi/audio.json > > +++ b/qapi/audio.json > > @@ -324,6 +324,47 @@ > > '*out': 'AudiodevPaPerDirectionOptions', > > '*server': 'str' } } > > > > +## > > +# @AudiodevPipewirePerDirectionOptions: > > +# > > +# Options of the Pipewire backend that are used for both playback and > > +# recording. > > +# > > +# @name: name of the sink/source to use > > +# > > +# @stream-name: name of the Pipewire stream created by qemu. Can be > > +# used to identify the stream in Pipewire when you > > +# create multiple Pipewire devices or run multiple qemu > > +# instances (default: audiodev's id) > > +# > > +# @latency: latency you want Pipewire to achieve in microseconds > > +# (default 46000) > > +# > > +# Since: 8.1 > > +## > > +{ 'struct': 'AudiodevPipewirePerDirectionOptions', > > + 'base': 'AudiodevPerDirectionOptions', > > + 'data': { > > + '*name': 'str', > > + '*stream-name': 'str', > > + '*latency': 'uint32' } } > > + > > +## > > +# @AudiodevPipewireOptions: > > +# > > +# Options of the Pipewire audio backend. > > +# > > +# @in: options of the capture stream > > +# > > +# @out: options of the playback stream > > +# > > +# Since: 8.1 > > +## > > +{ 'struct': 'AudiodevPipewireOptions', > > + 'data': { > > + '*in': 'AudiodevPipewirePerDirectionOptions', > > + '*out': 'AudiodevPipewirePerDirectionOptions' } } > > + > > ## > > # @AudiodevSdlPerDirectionOptions: > > # > > @@ -416,6 +457,7 @@ > > { 'name': 'jack', 'if': 'CONFIG_AUDIO_JACK' }, > > { 'name': 'oss', 'if': 'CONFIG_AUDIO_OSS' }, > > { 'name': 'pa', 'if': 'CONFIG_AUDIO_PA' }, > > + { 'name': 'pipewire', 'if': 'CONFIG_AUDIO_PIPEWIRE' }, > > { 'name': 'sdl', 'if': 'CONFIG_AUDIO_SDL' }, > > { 'name': 'sndio', 'if': 'CONFIG_AUDIO_SNDIO' }, > > { 'name': 'spice', 'if': 'CONFIG_SPICE' }, > > @@ -456,6 +498,8 @@ > > 'if': 'CONFIG_AUDIO_OSS' }, > > 'pa': { 'type': 'AudiodevPaOptions', > > 'if': 'CONFIG_AUDIO_PA' }, > > + 'pipewire': { 'type': 'AudiodevPipewireOptions', > > + 'if': 'CONFIG_AUDIO_PIPEWIRE' }, > > 'sdl': { 'type': 'AudiodevSdlOptions', > > 'if': 'CONFIG_AUDIO_SDL' }, > > 'sndio': { 'type': 'AudiodevSndioOptions', > > diff --git a/qemu-options.hx b/qemu-options.hx > > index 59bdf67a2c..2d908717bd 100644 > > --- a/qemu-options.hx > > +++ b/qemu-options.hx > > @@ -779,6 +779,12 @@ DEF("audiodev", HAS_ARG, QEMU_OPTION_audiodev, > > " in|out.name= source/sink device name\n" > > " in|out.latency= desired latency in microseconds\n" > > #endif > > +#ifdef CONFIG_AUDIO_PIPEWIRE > > + "-audiodev pipewire,id=id[,prop[=value][,...]]\n" > > + " in|out.name= source/sink device name\n" > > + " in|out.stream-name= name of pipewire stream\n" > > + " in|out.latency= desired latency in microseconds\n" > > +#endif > > #ifdef CONFIG_AUDIO_SDL > > "-audiodev sdl,id=id[,prop[=value][,...]]\n" > > " in|out.buffer-count= number of buffers\n" > > @@ -942,6 +948,21 @@ SRST > > Desired latency in microseconds. The PulseAudio server will try > > to honor this value but actual latencies may be lower or > higher. > > > > +``-audiodev pipewire,id=id[,prop[=value][,...]]`` > > + Creates a backend using Pipewire. This backend is available on > > + most systems. > > + > > + Pipewire specific options are: > > + > > + ``in|out.latency=usecs`` > > + Desired latency in microseconds. > > + > > + ``in|out.name=sink`` > > + Use the specified source/sink for recording/playback. > > + > > + ``in|out.stream-name`` > > + Specify the name of pipewire stream. > > + > > ``-audiodev sdl,id=id[,prop[=value][,...]]`` > > Creates a backend using SDL. This backend is available on most > > systems, but you should use your platform's native backend if > > diff --git a/scripts/meson-buildoptions.sh > b/scripts/meson-buildoptions.sh > > index 009fab1515..ba1057b62c 100644 > > --- a/scripts/meson-buildoptions.sh > > +++ b/scripts/meson-buildoptions.sh > > @@ -1,7 +1,8 @@ > > # This file is generated by meson-buildoptions.py, do not edit! > > meson_options_help() { > > - printf "%s\n" ' --audio-drv-list=CHOICES Set audio driver list > [default] (choices: alsa/co' > > - printf "%s\n" ' > reaudio/default/dsound/jack/oss/pa/sdl/sndio)' > > + printf "%s\n" ' --audio-drv-list=CHOICES Set audio driver list > [default] (choices: al' > > + printf "%s\n" ' > sa/coreaudio/default/dsound/jack/oss/pa/' > > + printf "%s\n" ' pipewire/sdl/sndio)' > > printf "%s\n" ' --block-drv-ro-whitelist=VALUE' > > printf "%s\n" ' set block driver read-only > whitelist (by default' > > printf "%s\n" ' affects only QEMU, not > tools like qemu-img)' > > @@ -136,6 +137,7 @@ meson_options_help() { > > printf "%s\n" ' oss OSS sound support' > > printf "%s\n" ' pa PulseAudio sound support' > > printf "%s\n" ' parallels parallels image format support' > > + printf "%s\n" ' pipewire Pipewire sound support' > > printf "%s\n" ' png PNG support with libpng' > > printf "%s\n" ' pvrdma Enable PVRDMA support' > > printf "%s\n" ' qcow1 qcow1 image format support' > > @@ -370,6 +372,8 @@ _meson_option_parse() { > > --disable-pa) printf "%s" -Dpa=disabled ;; > > --enable-parallels) printf "%s" -Dparallels=enabled ;; > > --disable-parallels) printf "%s" -Dparallels=disabled ;; > > + --enable-pipewire) printf "%s" -Dpipewire=enabled ;; > > + --disable-pipewire) printf "%s" -Dpipewire=disabled ;; > > --with-pkgversion=*) quote_sh "-Dpkgversion=$2" ;; > > --enable-png) printf "%s" -Dpng=enabled ;; > > --disable-png) printf "%s" -Dpng=disabled ;; > >
diff --git a/audio/audio.c b/audio/audio.c index 70b096713c..90c7c49d11 100644 --- a/audio/audio.c +++ b/audio/audio.c @@ -2061,6 +2061,9 @@ void audio_create_pdos(Audiodev *dev) #ifdef CONFIG_AUDIO_PA CASE(PA, pa, Pa); #endif +#ifdef CONFIG_AUDIO_PIPEWIRE + CASE(PIPEWIRE, pipewire, Pipewire); +#endif #ifdef CONFIG_AUDIO_SDL CASE(SDL, sdl, Sdl); #endif diff --git a/audio/audio_template.h b/audio/audio_template.h index e42326c20d..dc0c74aa74 100644 --- a/audio/audio_template.h +++ b/audio/audio_template.h @@ -362,6 +362,10 @@ AudiodevPerDirectionOptions *glue(audio_get_pdo_, TYPE)(Audiodev *dev) case AUDIODEV_DRIVER_PA: return qapi_AudiodevPaPerDirectionOptions_base(dev->u.pa.TYPE); #endif +#ifdef CONFIG_AUDIO_PIPEWIRE + case AUDIODEV_DRIVER_PIPEWIRE: + return qapi_AudiodevPipewirePerDirectionOptions_base(dev->u.pipewire.TYPE); +#endif #ifdef CONFIG_AUDIO_SDL case AUDIODEV_DRIVER_SDL: return qapi_AudiodevSdlPerDirectionOptions_base(dev->u.sdl.TYPE); diff --git a/audio/meson.build b/audio/meson.build index 0722224ba9..65a49c1a10 100644 --- a/audio/meson.build +++ b/audio/meson.build @@ -19,6 +19,7 @@ foreach m : [ ['sdl', sdl, files('sdlaudio.c')], ['jack', jack, files('jackaudio.c')], ['sndio', sndio, files('sndioaudio.c')], + ['pipewire', pipewire, files('pwaudio.c')], ['spice', spice, files('spiceaudio.c')] ] if m[1].found() diff --git a/audio/pwaudio.c b/audio/pwaudio.c new file mode 100644 index 0000000000..adf1a538c0 --- /dev/null +++ b/audio/pwaudio.c @@ -0,0 +1,913 @@ +/* + * QEMU Pipewire audio driver + * + * Copyright (c) 2023 Red Hat Inc. + * + * Author: Dorinda Bassey <dbassey@redhat.com> + * + * SPDX-License-Identifier: GPL-2.0-or-later + */ + +#include "qemu/osdep.h" +#include "qemu/module.h" +#include "audio.h" +#include <errno.h> +#include "qemu/error-report.h" +#include <spa/param/audio/format-utils.h> +#include <spa/utils/ringbuffer.h> +#include <spa/utils/result.h> +#include <spa/param/props.h> + +#include <pipewire/pipewire.h> +#include "trace.h" + +#define AUDIO_CAP "pipewire" +#define RINGBUFFER_SIZE (1u << 22) +#define RINGBUFFER_MASK (RINGBUFFER_SIZE - 1) + +#include "audio_int.h" + +typedef struct pwvolume { + uint32_t channels; + float values[SPA_AUDIO_MAX_CHANNELS]; +} pwvolume; + +typedef struct pwaudio { + Audiodev *dev; + struct pw_thread_loop *thread_loop; + struct pw_context *context; + + struct pw_core *core; + struct spa_hook core_listener; + int last_seq, pending_seq, error; +} pwaudio; + +typedef struct PWVoice { + pwaudio *g; + struct pw_stream *stream; + struct spa_hook stream_listener; + struct spa_audio_info_raw info; + uint32_t highwater_mark; + uint32_t sample_size, req; + struct spa_ringbuffer ring; + uint8_t buffer[RINGBUFFER_SIZE]; + + pwvolume volume; + bool muted; +} PWVoice; + +typedef struct PWVoiceOut { + HWVoiceOut hw; + PWVoice v; +} PWVoiceOut; + +typedef struct PWVoiceIn { + HWVoiceIn hw; + PWVoice v; +} PWVoiceIn; + +static void +stream_destroy(void *data) +{ + PWVoice *v = (PWVoice *) data; + spa_hook_remove(&v->stream_listener); + v->stream = NULL; +} + +/* output data processing function to read stuffs from the buffer */ +static void +playback_on_process(void *data) +{ + PWVoice *v = (PWVoice *) data; + void *p; + struct pw_buffer *b; + struct spa_buffer *buf; + uint32_t req, index, n_bytes; + int32_t avail; + + assert(v->stream); + + /* obtain a buffer to read from */ + b = pw_stream_dequeue_buffer(v->stream); + if (b == NULL) { + error_report("out of buffers: %s", strerror(errno)); + return; + } + + buf = b->buffer; + p = buf->datas[0].data; + if (p == NULL) { + return; + } + /* calculate the total no of bytes to read data from buffer */ + req = b->requested * v->sample_size; + if (req == 0) { + req = v->req; + } + n_bytes = SPA_MIN(req, buf->datas[0].maxsize); + + /* get no of available bytes to read data from buffer */ + + avail = spa_ringbuffer_get_read_index(&v->ring, &index); + + if (avail <= 0) { + /* underrun, can't really happen but if it does we */ + /* do nothing and wait for more data */ + error_report("%p: underrun read:%u avail:%d", p, index, avail); + } 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->sample_size; + buf->datas[0].chunk->size = n_bytes; + + /* queue the buffer for playback */ + pw_stream_queue_buffer(v->stream, b); +} + +/* output data processing function to generate stuffs in the buffer */ +static void +capture_on_process(void *data) +{ + PWVoice *v = (PWVoice *) data; + void *p; + struct pw_buffer *b; + struct spa_buffer *buf; + int32_t filled; + uint32_t index, offs, n_bytes; + + assert(v->stream); + + /* obtain a buffer */ + b = pw_stream_dequeue_buffer(v->stream); + if (b == NULL) { + error_report("out of buffers: %s", strerror(errno)); + return; + } + + /* Write data into buffer */ + buf = b->buffer; + p = buf->datas[0].data; + if (p == NULL) { + return; + } + offs = SPA_MIN(buf->datas[0].chunk->offset, buf->datas[0].maxsize); + n_bytes = SPA_MIN(buf->datas[0].chunk->size, buf->datas[0].maxsize - offs); + + filled = spa_ringbuffer_get_write_index(&v->ring, &index); + + + if (filled < 0) { + error_report("%p: underrun write:%u filled:%d", p, index, filled); + } else { + if ((uint32_t) filled + n_bytes > RINGBUFFER_SIZE) { + error_report("%p: overrun write:%u filled:%d + size:%u > max:%u", + p, index, filled, n_bytes, RINGBUFFER_SIZE); + } + } + spa_ringbuffer_write_data(&v->ring, + v->buffer, RINGBUFFER_SIZE, + index & RINGBUFFER_MASK, + SPA_PTROFF(p, offs, void), n_bytes); + index += n_bytes; + spa_ringbuffer_write_update(&v->ring, index); + + /* queue the buffer for playback */ + pw_stream_queue_buffer(v->stream, b); +} + +static void +on_stream_state_changed(void *data, enum pw_stream_state old, + enum pw_stream_state state, const char *error) +{ + PWVoice *v = (PWVoice *) data; + + trace_pw_state_changed(pw_stream_get_node_id(v->stream), + pw_stream_state_as_string(state)); + + switch (state) { + case PW_STREAM_STATE_ERROR: + case PW_STREAM_STATE_UNCONNECTED: + break; + case PW_STREAM_STATE_PAUSED: + case PW_STREAM_STATE_CONNECTING: + case PW_STREAM_STATE_STREAMING: + break; + } +} + +static const struct pw_stream_events capture_stream_events = { + PW_VERSION_STREAM_EVENTS, + .destroy = stream_destroy, + .state_changed = on_stream_state_changed, + .process = capture_on_process +}; + +static const struct pw_stream_events playback_stream_events = { + PW_VERSION_STREAM_EVENTS, + .destroy = stream_destroy, + .state_changed = on_stream_state_changed, + .process = playback_on_process +}; + +static size_t +qpw_read(HWVoiceIn *hw, void *data, size_t len) +{ + PWVoiceIn *pw = (PWVoiceIn *) hw; + PWVoice *v = &pw->v; + pwaudio *c = v->g; + const char *error = NULL; + size_t l; + int32_t avail; + uint32_t index; + + pw_thread_loop_lock(c->thread_loop); + if (pw_stream_get_state(v->stream, &error) != PW_STREAM_STATE_STREAMING) { + /* wait for stream to become ready */ + l = 0; + goto done_unlock; + } + /* get no of available bytes to read data from buffer */ + avail = spa_ringbuffer_get_read_index(&v->ring, &index); + + trace_pw_read(avail, index, len); + + if (avail < (int32_t) len) { + len = avail; + } + + spa_ringbuffer_read_data(&v->ring, + v->buffer, RINGBUFFER_SIZE, + index & RINGBUFFER_MASK, data, len); + index += len; + spa_ringbuffer_read_update(&v->ring, index); + l = len; + +done_unlock: + pw_thread_loop_unlock(c->thread_loop); + return l; +} + +static size_t qpw_buffer_get_free(HWVoiceOut *hw) +{ + PWVoiceOut *pw = (PWVoiceOut *)hw; + PWVoice *v = &pw->v; + pwaudio *c = v->g; + const char *error = NULL; + int32_t filled, avail; + uint32_t index; + + pw_thread_loop_lock(c->thread_loop); + if (pw_stream_get_state(v->stream, &error) != PW_STREAM_STATE_STREAMING) { + /* wait for stream to become ready */ + avail = 0; + goto done_unlock; + } + + filled = spa_ringbuffer_get_write_index(&v->ring, &index); + avail = v->highwater_mark - filled; + +done_unlock: + pw_thread_loop_unlock(c->thread_loop); + return avail; +} + +static size_t +qpw_write(HWVoiceOut *hw, void *data, size_t len) +{ + PWVoiceOut *pw = (PWVoiceOut *) hw; + PWVoice *v = &pw->v; + pwaudio *c = v->g; + const char *error = NULL; + int32_t filled, avail; + uint32_t index; + + pw_thread_loop_lock(c->thread_loop); + if (pw_stream_get_state(v->stream, &error) != PW_STREAM_STATE_STREAMING) { + /* wait for stream to become ready */ + len = 0; + goto done_unlock; + } + filled = spa_ringbuffer_get_write_index(&v->ring, &index); + avail = v->highwater_mark - filled; + + trace_pw_write(filled, avail, index, len); + + if (len > avail) { + len = avail; + } + + if (filled < 0) { + audio_pcm_info_clear_buf(&hw->info, data, len / hw->info.bytes_per_frame); + error_report("%p: underrun write:%u filled:%d", pw, index, filled); + } else { + if ((uint32_t) filled + len > RINGBUFFER_SIZE) { + error_report("%p: overrun write:%u filled:%d + size:%zu > max:%u", + pw, index, filled, len, RINGBUFFER_SIZE); + } + } + + spa_ringbuffer_write_data(&v->ring, + v->buffer, RINGBUFFER_SIZE, + index & RINGBUFFER_MASK, data, len); + index += len; + spa_ringbuffer_write_update(&v->ring, index); + +done_unlock: + pw_thread_loop_unlock(c->thread_loop); + return len; +} + +static int +audfmt_to_pw(AudioFormat fmt, int endianness) +{ + int format; + + switch (fmt) { + case AUDIO_FORMAT_S8: + format = SPA_AUDIO_FORMAT_S8; + break; + case AUDIO_FORMAT_U8: + format = SPA_AUDIO_FORMAT_U8; + break; + case AUDIO_FORMAT_S16: + format = endianness ? SPA_AUDIO_FORMAT_S16_BE : SPA_AUDIO_FORMAT_S16_LE; + break; + case AUDIO_FORMAT_U16: + format = endianness ? SPA_AUDIO_FORMAT_U16_BE : SPA_AUDIO_FORMAT_U16_LE; + break; + case AUDIO_FORMAT_S32: + format = endianness ? SPA_AUDIO_FORMAT_S32_BE : SPA_AUDIO_FORMAT_S32_LE; + break; + case AUDIO_FORMAT_U32: + format = endianness ? SPA_AUDIO_FORMAT_U32_BE : SPA_AUDIO_FORMAT_U32_LE; + break; + case AUDIO_FORMAT_F32: + format = endianness ? SPA_AUDIO_FORMAT_F32_BE : SPA_AUDIO_FORMAT_F32_LE; + break; + default: + dolog("Internal logic error: Bad audio format %d\n", fmt); + format = SPA_AUDIO_FORMAT_U8; + break; + } + return format; +} + +static AudioFormat +pw_to_audfmt(enum spa_audio_format fmt, int *endianness, + uint32_t *sample_size) +{ + switch (fmt) { + case SPA_AUDIO_FORMAT_S8: + *sample_size = 1; + return AUDIO_FORMAT_S8; + case SPA_AUDIO_FORMAT_U8: + *sample_size = 1; + return AUDIO_FORMAT_U8; + case SPA_AUDIO_FORMAT_S16_BE: + *sample_size = 2; + *endianness = 1; + return AUDIO_FORMAT_S16; + case SPA_AUDIO_FORMAT_S16_LE: + *sample_size = 2; + *endianness = 0; + return AUDIO_FORMAT_S16; + case SPA_AUDIO_FORMAT_U16_BE: + *sample_size = 2; + *endianness = 1; + return AUDIO_FORMAT_U16; + case SPA_AUDIO_FORMAT_U16_LE: + *sample_size = 2; + *endianness = 0; + return AUDIO_FORMAT_U16; + case SPA_AUDIO_FORMAT_S32_BE: + *sample_size = 4; + *endianness = 1; + return AUDIO_FORMAT_S32; + case SPA_AUDIO_FORMAT_S32_LE: + *sample_size = 4; + *endianness = 0; + return AUDIO_FORMAT_S32; + case SPA_AUDIO_FORMAT_U32_BE: + *sample_size = 4; + *endianness = 1; + return AUDIO_FORMAT_U32; + case SPA_AUDIO_FORMAT_U32_LE: + *sample_size = 4; + *endianness = 0; + return AUDIO_FORMAT_U32; + case SPA_AUDIO_FORMAT_F32_BE: + *sample_size = 4; + *endianness = 1; + return AUDIO_FORMAT_F32; + case SPA_AUDIO_FORMAT_F32_LE: + *sample_size = 4; + *endianness = 0; + return AUDIO_FORMAT_F32; + default: + *sample_size = 1; + dolog("Internal logic error: Bad spa_audio_format %d\n", fmt); + return AUDIO_FORMAT_U8; + } +} + +static int +create_stream(pwaudio *c, PWVoice *v, const char *stream_name, + const char *name, enum spa_direction dir) +{ + int res; + uint32_t n_params; + const struct spa_pod *params[2]; + uint8_t buffer[1024]; + struct spa_pod_builder b; + uint64_t buf_samples; + struct pw_properties *props; + + props = pw_properties_new(NULL, NULL); + + /* 75% of the timer period for faster updates */ + buf_samples = (uint64_t)v->g->dev->timer_period * v->info.rate + * 3 / 4 / 1000000; + trace_pw_timer(v->g->dev->timer_period); + pw_properties_setf(props, PW_KEY_NODE_LATENCY, "%" PRIu64 "/%u", + buf_samples, v->info.rate); + + if (name) { + pw_properties_set(props, PW_KEY_TARGET_OBJECT, name); + } + v->stream = pw_stream_new(c->core, stream_name, props); + + if (v->stream == NULL) { + return -1; + } + + if (dir == SPA_DIRECTION_INPUT) { + pw_stream_add_listener(v->stream, + &v->stream_listener, &capture_stream_events, v); + } else { + pw_stream_add_listener(v->stream, + &v->stream_listener, &playback_stream_events, v); + } + + n_params = 0; + spa_pod_builder_init(&b, buffer, sizeof(buffer)); + params[n_params++] = spa_format_audio_raw_build(&b, + SPA_PARAM_EnumFormat, + &v->info); + + /* connect the stream to a sink or source */ + res = pw_stream_connect(v->stream, + dir == + SPA_DIRECTION_INPUT ? PW_DIRECTION_INPUT : + PW_DIRECTION_OUTPUT, PW_ID_ANY, + PW_STREAM_FLAG_AUTOCONNECT | + PW_STREAM_FLAG_INACTIVE | + PW_STREAM_FLAG_MAP_BUFFERS | + PW_STREAM_FLAG_RT_PROCESS, params, n_params); + if (res < 0) { + pw_stream_destroy(v->stream); + return -1; + } + + return 0; +} + +static int +qpw_stream_new(pwaudio *c, PWVoice *v, const char *stream_name, + const char *name, enum spa_direction dir) +{ + int r; + + switch (v->info.channels) { + case 8: + v->info.position[0] = SPA_AUDIO_CHANNEL_FL; + v->info.position[1] = SPA_AUDIO_CHANNEL_FR; + v->info.position[2] = SPA_AUDIO_CHANNEL_FC; + v->info.position[3] = SPA_AUDIO_CHANNEL_LFE; + v->info.position[4] = SPA_AUDIO_CHANNEL_RL; + v->info.position[5] = SPA_AUDIO_CHANNEL_RR; + v->info.position[6] = SPA_AUDIO_CHANNEL_SL; + v->info.position[7] = SPA_AUDIO_CHANNEL_SR; + break; + case 6: + v->info.position[0] = SPA_AUDIO_CHANNEL_FL; + v->info.position[1] = SPA_AUDIO_CHANNEL_FR; + v->info.position[2] = SPA_AUDIO_CHANNEL_FC; + v->info.position[3] = SPA_AUDIO_CHANNEL_LFE; + v->info.position[4] = SPA_AUDIO_CHANNEL_RL; + v->info.position[5] = SPA_AUDIO_CHANNEL_RR; + break; + case 5: + v->info.position[0] = SPA_AUDIO_CHANNEL_FL; + v->info.position[1] = SPA_AUDIO_CHANNEL_FR; + v->info.position[2] = SPA_AUDIO_CHANNEL_FC; + v->info.position[3] = SPA_AUDIO_CHANNEL_LFE; + v->info.position[4] = SPA_AUDIO_CHANNEL_RC; + break; + case 4: + v->info.position[0] = SPA_AUDIO_CHANNEL_FL; + v->info.position[1] = SPA_AUDIO_CHANNEL_FR; + v->info.position[2] = SPA_AUDIO_CHANNEL_FC; + v->info.position[3] = SPA_AUDIO_CHANNEL_RC; + break; + case 3: + v->info.position[0] = SPA_AUDIO_CHANNEL_FL; + v->info.position[1] = SPA_AUDIO_CHANNEL_FR; + v->info.position[2] = SPA_AUDIO_CHANNEL_LFE; + break; + case 2: + v->info.position[0] = SPA_AUDIO_CHANNEL_FL; + v->info.position[1] = SPA_AUDIO_CHANNEL_FR; + break; + case 1: + v->info.position[0] = SPA_AUDIO_CHANNEL_MONO; + break; + default: + for (size_t i = 0; i < v->info.channels; i++) { + v->info.position[i] = SPA_AUDIO_CHANNEL_UNKNOWN; + } + break; + } + + /* create a new unconnected pwstream */ + r = create_stream(c, v, stream_name, name, dir); + if (r < 0) { + AUD_log(AUDIO_CAP, "Failed to create stream."); + return -1; + } + + return r; +} + +static int +qpw_init_out(HWVoiceOut *hw, struct audsettings *as, void *drv_opaque) +{ + PWVoiceOut *pw = (PWVoiceOut *) hw; + PWVoice *v = &pw->v; + struct audsettings obt_as = *as; + pwaudio *c = v->g = drv_opaque; + AudiodevPipewireOptions *popts = &c->dev->u.pipewire; + AudiodevPipewirePerDirectionOptions *ppdo = popts->out; + int r; + + pw_thread_loop_lock(c->thread_loop); + + v->info.format = audfmt_to_pw(as->fmt, as->endianness); + v->info.channels = as->nchannels; + v->info.rate = as->freq; + + obt_as.fmt = + pw_to_audfmt(v->info.format, &obt_as.endianness, &v->sample_size); + v->sample_size *= as->nchannels; + + v->req = (uint64_t)c->dev->timer_period * v->info.rate + * 1 / 2 / 1000000 * v->sample_size; + + /* call the function that creates a new stream for playback */ + r = qpw_stream_new(c, v, ppdo->stream_name ? : c->dev->id, + ppdo->name, SPA_DIRECTION_OUTPUT); + if (r < 0) { + error_report("qpw_stream_new for playback failed"); + pw_thread_loop_unlock(c->thread_loop); + return -1; + } + + /* report the audio format we support */ + audio_pcm_init_info(&hw->info, &obt_as); + + /* report the buffer size to qemu */ + hw->samples = audio_buffer_frames( + qapi_AudiodevPipewirePerDirectionOptions_base(ppdo), &obt_as, 46440); + v->highwater_mark = MIN(RINGBUFFER_SIZE, + (ppdo->has_latency ? ppdo->latency : 46440) + * (uint64_t)v->info.rate / 1000000 * v->sample_size); + + pw_thread_loop_unlock(c->thread_loop); + return 0; +} + +static int +qpw_init_in(HWVoiceIn *hw, struct audsettings *as, void *drv_opaque) +{ + PWVoiceIn *pw = (PWVoiceIn *) hw; + PWVoice *v = &pw->v; + struct audsettings obt_as = *as; + pwaudio *c = v->g = drv_opaque; + AudiodevPipewireOptions *popts = &c->dev->u.pipewire; + AudiodevPipewirePerDirectionOptions *ppdo = popts->in; + int r; + + pw_thread_loop_lock(c->thread_loop); + + v->info.format = audfmt_to_pw(as->fmt, as->endianness); + v->info.channels = as->nchannels; + v->info.rate = as->freq; + + obt_as.fmt = + pw_to_audfmt(v->info.format, &obt_as.endianness, &v->sample_size); + v->sample_size *= as->nchannels; + + /* call the function that creates a new stream for recording */ + r = qpw_stream_new(c, v, ppdo->stream_name ? : c->dev->id, + ppdo->name, SPA_DIRECTION_INPUT); + if (r < 0) { + error_report("qpw_stream_new for recording failed"); + pw_thread_loop_unlock(c->thread_loop); + return -1; + } + + /* report the audio format we support */ + audio_pcm_init_info(&hw->info, &obt_as); + + /* report the buffer size to qemu */ + hw->samples = audio_buffer_frames( + qapi_AudiodevPipewirePerDirectionOptions_base(ppdo), &obt_as, 46440); + + pw_thread_loop_unlock(c->thread_loop); + return 0; +} + +static void +qpw_fini_out(HWVoiceOut *hw) +{ + PWVoiceOut *pw = (PWVoiceOut *) hw; + PWVoice *v = &pw->v; + + if (v->stream) { + pwaudio *c = v->g; + pw_thread_loop_lock(c->thread_loop); + pw_stream_destroy(v->stream); + v->stream = NULL; + pw_thread_loop_unlock(c->thread_loop); + } +} + +static void +qpw_fini_in(HWVoiceIn *hw) +{ + PWVoiceIn *pw = (PWVoiceIn *) hw; + PWVoice *v = &pw->v; + + if (v->stream) { + pwaudio *c = v->g; + pw_thread_loop_lock(c->thread_loop); + pw_stream_destroy(v->stream); + v->stream = NULL; + pw_thread_loop_unlock(c->thread_loop); + } +} + +static void +qpw_enable_out(HWVoiceOut *hw, bool enable) +{ + PWVoiceOut *po = (PWVoiceOut *) hw; + PWVoice *v = &po->v; + pwaudio *c = v->g; + pw_thread_loop_lock(c->thread_loop); + pw_stream_set_active(v->stream, enable); + pw_thread_loop_unlock(c->thread_loop); +} + +static void +qpw_enable_in(HWVoiceIn *hw, bool enable) +{ + PWVoiceIn *pi = (PWVoiceIn *) hw; + PWVoice *v = &pi->v; + pwaudio *c = v->g; + pw_thread_loop_lock(c->thread_loop); + pw_stream_set_active(v->stream, enable); + pw_thread_loop_unlock(c->thread_loop); +} + +static void +qpw_volume_out(HWVoiceOut *hw, Volume *vol) +{ + PWVoiceOut *pw = (PWVoiceOut *) hw; + PWVoice *v = &pw->v; + pwaudio *c = v->g; + int i, ret; + + pw_thread_loop_lock(c->thread_loop); + v->volume.channels = vol->channels; + + for (i = 0; i < vol->channels; ++i) { + v->volume.values[i] = (float)vol->vol[i] / 255; + } + + ret = pw_stream_set_control(v->stream, + SPA_PROP_channelVolumes, v->volume.channels, v->volume.values, 0); + trace_pw_vol(ret == 0 ? "success" : "failed"); + + v->muted = vol->mute; + float val = v->muted ? 1.f : 0.f; + ret = pw_stream_set_control(v->stream, SPA_PROP_mute, 1, &val, 0); + pw_thread_loop_unlock(c->thread_loop); +} + +static void +qpw_volume_in(HWVoiceIn *hw, Volume *vol) +{ + PWVoiceIn *pw = (PWVoiceIn *) hw; + PWVoice *v = &pw->v; + pwaudio *c = v->g; + int i, ret; + + pw_thread_loop_lock(c->thread_loop); + v->volume.channels = vol->channels; + + for (i = 0; i < vol->channels; ++i) { + v->volume.values[i] = (float)vol->vol[i] / 255; + } + + ret = pw_stream_set_control(v->stream, + SPA_PROP_channelVolumes, v->volume.channels, v->volume.values, 0); + trace_pw_vol(ret == 0 ? "success" : "failed"); + + v->muted = vol->mute; + float val = v->muted ? 1.f : 0.f; + ret = pw_stream_set_control(v->stream, SPA_PROP_mute, 1, &val, 0); + pw_thread_loop_unlock(c->thread_loop); +} + +static int wait_resync(pwaudio *pw) +{ + int res; + pw->pending_seq = pw_core_sync(pw->core, PW_ID_CORE, pw->pending_seq); + + while (true) { + pw_thread_loop_wait(pw->thread_loop); + + res = pw->error; + if (res < 0) { + pw->error = 0; + return res; + } + if (pw->pending_seq == pw->last_seq) { + break; + } + } + return 0; +} +static void +on_core_error(void *data, uint32_t id, int seq, int res, const char *message) +{ + pwaudio *pw = data; + + error_report("error id:%u seq:%d res:%d (%s): %s", + id, seq, res, spa_strerror(res), message); + + /* stop and exit the thread loop */ + pw_thread_loop_signal(pw->thread_loop, FALSE); +} + +static void +on_core_done(void *data, uint32_t id, int seq) +{ + pwaudio *pw = data; + assert(id == PW_ID_CORE); + pw->last_seq = seq; + if (pw->pending_seq == seq) { + /* stop and exit the thread loop */ + pw_thread_loop_signal(pw->thread_loop, FALSE); + } +} + +static const struct pw_core_events core_events = { + PW_VERSION_CORE_EVENTS, + .done = on_core_done, + .error = on_core_error, +}; + +static void * +qpw_audio_init(Audiodev *dev) +{ + g_autofree pwaudio *pw = g_new0(pwaudio, 1); + pw_init(NULL, NULL); + + trace_pw_audio_init(); + assert(dev->driver == AUDIODEV_DRIVER_PIPEWIRE); + + pw->dev = dev; + pw->thread_loop = pw_thread_loop_new("Pipewire thread loop", NULL); + if (pw->thread_loop == NULL) { + error_report("Could not create Pipewire loop"); + goto fail; + } + + pw->context = + pw_context_new(pw_thread_loop_get_loop(pw->thread_loop), NULL, 0); + if (pw->context == NULL) { + error_report("Could not create Pipewire context"); + goto fail; + } + + if (pw_thread_loop_start(pw->thread_loop) < 0) { + error_report("Could not start Pipewire loop"); + goto fail; + } + + pw_thread_loop_lock(pw->thread_loop); + + pw->core = pw_context_connect(pw->context, NULL, 0); + if (pw->core == NULL) { + pw_thread_loop_unlock(pw->thread_loop); + goto fail; + } + + if (pw_core_add_listener(pw->core, &pw->core_listener, + &core_events, pw) < 0) { + pw_thread_loop_unlock(pw->thread_loop); + goto fail; + } + if (wait_resync(pw) < 0) { + pw_thread_loop_unlock(pw->thread_loop); + } + + pw_thread_loop_unlock(pw->thread_loop); + + return g_steal_pointer(&pw); + +fail: + AUD_log(AUDIO_CAP, "Failed to initialize PW context"); + if (pw->thread_loop) { + pw_thread_loop_stop(pw->thread_loop); + } + if (pw->context) { + g_clear_pointer(&pw->context, pw_context_destroy); + } + if (pw->thread_loop) { + g_clear_pointer(&pw->thread_loop, pw_thread_loop_destroy); + } + return NULL; +} + +static void +qpw_audio_fini(void *opaque) +{ + pwaudio *pw = opaque; + + if (pw->thread_loop) { + pw_thread_loop_stop(pw->thread_loop); + } + + if (pw->core) { + spa_hook_remove(&pw->core_listener); + spa_zero(pw->core_listener); + pw_core_disconnect(pw->core); + } + + if (pw->context) { + pw_context_destroy(pw->context); + } + pw_thread_loop_destroy(pw->thread_loop); + + g_free(pw); +} + +static struct audio_pcm_ops qpw_pcm_ops = { + .init_out = qpw_init_out, + .fini_out = qpw_fini_out, + .write = qpw_write, + .buffer_get_free = qpw_buffer_get_free, + .run_buffer_out = audio_generic_run_buffer_out, + .enable_out = qpw_enable_out, + .volume_out = qpw_volume_out, + .volume_in = qpw_volume_in, + + .init_in = qpw_init_in, + .fini_in = qpw_fini_in, + .read = qpw_read, + .run_buffer_in = audio_generic_run_buffer_in, + .enable_in = qpw_enable_in +}; + +static struct audio_driver pw_audio_driver = { + .name = "pipewire", + .descr = "http://www.pipewire.org/", + .init = qpw_audio_init, + .fini = qpw_audio_fini, + .pcm_ops = &qpw_pcm_ops, + .can_be_default = 1, + .max_voices_out = INT_MAX, + .max_voices_in = INT_MAX, + .voice_size_out = sizeof(PWVoiceOut), + .voice_size_in = sizeof(PWVoiceIn), +}; + +static void +register_audio_pw(void) +{ + audio_driver_register(&pw_audio_driver); +} + +type_init(register_audio_pw); diff --git a/audio/trace-events b/audio/trace-events index e1ab643add..c764e5641b 100644 --- a/audio/trace-events +++ b/audio/trace-events @@ -18,6 +18,14 @@ dbus_audio_register(const char *s, const char *dir) "sender = %s, dir = %s" dbus_audio_put_buffer_out(size_t len) "len = %zu" dbus_audio_read(size_t len) "len = %zu" +# pwaudio.c +pw_state_changed(int nodeid, const char *s) "node id: %d stream state: %s" +pw_read(int32_t avail, uint32_t index, size_t len) "avail=%d index=%u len=%zu" +pw_write(int32_t filled, int32_t avail, uint32_t index, size_t len) "filled=%d avail=%d index=%u len=%zu" +pw_vol(const char *ret) "set volume: %s" +pw_timer(uint64_t buf_samples) "timer period = %" PRIu64 +pw_audio_init(void) "Initialize Pipewire context" + # audio.c audio_timer_start(int interval) "interval %d ms" audio_timer_stop(void) "" diff --git a/meson.build b/meson.build index 29f8644d6d..31bf280c0d 100644 --- a/meson.build +++ b/meson.build @@ -730,6 +730,12 @@ if not get_option('jack').auto() or have_system jack = dependency('jack', required: get_option('jack'), method: 'pkg-config', kwargs: static_kwargs) endif +pipewire = not_found +if not get_option('pipewire').auto() or (targetos == 'linux' and have_system) + pipewire = dependency('libpipewire-0.3', version: '>=0.3.60', + required: get_option('pipewire'), + method: 'pkg-config', kwargs: static_kwargs) +endif sndio = not_found if not get_option('sndio').auto() or have_system sndio = dependency('sndio', required: get_option('sndio'), @@ -1667,6 +1673,7 @@ if have_system 'jack': jack.found(), 'oss': oss.found(), 'pa': pulse.found(), + 'pipewire': pipewire.found(), 'sdl': sdl.found(), 'sndio': sndio.found(), } @@ -3980,6 +3987,7 @@ if targetos == 'linux' summary_info += {'ALSA support': alsa} summary_info += {'PulseAudio support': pulse} endif +summary_info += {'Pipewire support': pipewire} summary_info += {'JACK support': jack} summary_info += {'brlapi support': brlapi} summary_info += {'vde support': vde} diff --git a/meson_options.txt b/meson_options.txt index fc9447d267..9ae1ec7f47 100644 --- a/meson_options.txt +++ b/meson_options.txt @@ -21,7 +21,7 @@ option('tls_priority', type : 'string', value : 'NORMAL', option('default_devices', type : 'boolean', value : true, description: 'Include a default selection of devices in emulators') option('audio_drv_list', type: 'array', value: ['default'], - choices: ['alsa', 'coreaudio', 'default', 'dsound', 'jack', 'oss', 'pa', 'sdl', 'sndio'], + choices: ['alsa', 'coreaudio', 'default', 'dsound', 'jack', 'oss', 'pa', 'pipewire', 'sdl', 'sndio'], description: 'Set audio driver list') option('block_drv_rw_whitelist', type : 'string', value : '', description: 'set block driver read-write whitelist (by default affects only QEMU, not tools like qemu-img)') @@ -255,6 +255,8 @@ option('oss', type: 'feature', value: 'auto', description: 'OSS sound support') option('pa', type: 'feature', value: 'auto', description: 'PulseAudio sound support') +option('pipewire', type: 'feature', value: 'auto', + description: 'Pipewire sound support') option('sndio', type: 'feature', value: 'auto', description: 'sndio sound support') diff --git a/qapi/audio.json b/qapi/audio.json index 4e54c00f51..e03396a7bc 100644 --- a/qapi/audio.json +++ b/qapi/audio.json @@ -324,6 +324,47 @@ '*out': 'AudiodevPaPerDirectionOptions', '*server': 'str' } } +## +# @AudiodevPipewirePerDirectionOptions: +# +# Options of the Pipewire backend that are used for both playback and +# recording. +# +# @name: name of the sink/source to use +# +# @stream-name: name of the Pipewire stream created by qemu. Can be +# used to identify the stream in Pipewire when you +# create multiple Pipewire devices or run multiple qemu +# instances (default: audiodev's id) +# +# @latency: latency you want Pipewire to achieve in microseconds +# (default 46000) +# +# Since: 8.1 +## +{ 'struct': 'AudiodevPipewirePerDirectionOptions', + 'base': 'AudiodevPerDirectionOptions', + 'data': { + '*name': 'str', + '*stream-name': 'str', + '*latency': 'uint32' } } + +## +# @AudiodevPipewireOptions: +# +# Options of the Pipewire audio backend. +# +# @in: options of the capture stream +# +# @out: options of the playback stream +# +# Since: 8.1 +## +{ 'struct': 'AudiodevPipewireOptions', + 'data': { + '*in': 'AudiodevPipewirePerDirectionOptions', + '*out': 'AudiodevPipewirePerDirectionOptions' } } + ## # @AudiodevSdlPerDirectionOptions: # @@ -416,6 +457,7 @@ { 'name': 'jack', 'if': 'CONFIG_AUDIO_JACK' }, { 'name': 'oss', 'if': 'CONFIG_AUDIO_OSS' }, { 'name': 'pa', 'if': 'CONFIG_AUDIO_PA' }, + { 'name': 'pipewire', 'if': 'CONFIG_AUDIO_PIPEWIRE' }, { 'name': 'sdl', 'if': 'CONFIG_AUDIO_SDL' }, { 'name': 'sndio', 'if': 'CONFIG_AUDIO_SNDIO' }, { 'name': 'spice', 'if': 'CONFIG_SPICE' }, @@ -456,6 +498,8 @@ 'if': 'CONFIG_AUDIO_OSS' }, 'pa': { 'type': 'AudiodevPaOptions', 'if': 'CONFIG_AUDIO_PA' }, + 'pipewire': { 'type': 'AudiodevPipewireOptions', + 'if': 'CONFIG_AUDIO_PIPEWIRE' }, 'sdl': { 'type': 'AudiodevSdlOptions', 'if': 'CONFIG_AUDIO_SDL' }, 'sndio': { 'type': 'AudiodevSndioOptions', diff --git a/qemu-options.hx b/qemu-options.hx index 59bdf67a2c..2d908717bd 100644 --- a/qemu-options.hx +++ b/qemu-options.hx @@ -779,6 +779,12 @@ DEF("audiodev", HAS_ARG, QEMU_OPTION_audiodev, " in|out.name= source/sink device name\n" " in|out.latency= desired latency in microseconds\n" #endif +#ifdef CONFIG_AUDIO_PIPEWIRE + "-audiodev pipewire,id=id[,prop[=value][,...]]\n" + " in|out.name= source/sink device name\n" + " in|out.stream-name= name of pipewire stream\n" + " in|out.latency= desired latency in microseconds\n" +#endif #ifdef CONFIG_AUDIO_SDL "-audiodev sdl,id=id[,prop[=value][,...]]\n" " in|out.buffer-count= number of buffers\n" @@ -942,6 +948,21 @@ SRST Desired latency in microseconds. The PulseAudio server will try to honor this value but actual latencies may be lower or higher. +``-audiodev pipewire,id=id[,prop[=value][,...]]`` + Creates a backend using Pipewire. This backend is available on + most systems. + + Pipewire specific options are: + + ``in|out.latency=usecs`` + Desired latency in microseconds. + + ``in|out.name=sink`` + Use the specified source/sink for recording/playback. + + ``in|out.stream-name`` + Specify the name of pipewire stream. + ``-audiodev sdl,id=id[,prop[=value][,...]]`` Creates a backend using SDL. This backend is available on most systems, but you should use your platform's native backend if diff --git a/scripts/meson-buildoptions.sh b/scripts/meson-buildoptions.sh index 009fab1515..ba1057b62c 100644 --- a/scripts/meson-buildoptions.sh +++ b/scripts/meson-buildoptions.sh @@ -1,7 +1,8 @@ # This file is generated by meson-buildoptions.py, do not edit! meson_options_help() { - printf "%s\n" ' --audio-drv-list=CHOICES Set audio driver list [default] (choices: alsa/co' - printf "%s\n" ' reaudio/default/dsound/jack/oss/pa/sdl/sndio)' + printf "%s\n" ' --audio-drv-list=CHOICES Set audio driver list [default] (choices: al' + printf "%s\n" ' sa/coreaudio/default/dsound/jack/oss/pa/' + printf "%s\n" ' pipewire/sdl/sndio)' printf "%s\n" ' --block-drv-ro-whitelist=VALUE' printf "%s\n" ' set block driver read-only whitelist (by default' printf "%s\n" ' affects only QEMU, not tools like qemu-img)' @@ -136,6 +137,7 @@ meson_options_help() { printf "%s\n" ' oss OSS sound support' printf "%s\n" ' pa PulseAudio sound support' printf "%s\n" ' parallels parallels image format support' + printf "%s\n" ' pipewire Pipewire sound support' printf "%s\n" ' png PNG support with libpng' printf "%s\n" ' pvrdma Enable PVRDMA support' printf "%s\n" ' qcow1 qcow1 image format support' @@ -370,6 +372,8 @@ _meson_option_parse() { --disable-pa) printf "%s" -Dpa=disabled ;; --enable-parallels) printf "%s" -Dparallels=enabled ;; --disable-parallels) printf "%s" -Dparallels=disabled ;; + --enable-pipewire) printf "%s" -Dpipewire=enabled ;; + --disable-pipewire) printf "%s" -Dpipewire=disabled ;; --with-pkgversion=*) quote_sh "-Dpkgversion=$2" ;; --enable-png) printf "%s" -Dpng=enabled ;; --disable-png) printf "%s" -Dpng=disabled ;;
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> --- v11: handle buffer underruns in qpw_write use local variable change param name frame_size fix format specifier change trace value to trace quantum audio/audio.c | 3 + audio/audio_template.h | 4 + audio/meson.build | 1 + audio/pwaudio.c | 913 ++++++++++++++++++++++++++++++++++ audio/trace-events | 8 + meson.build | 8 + meson_options.txt | 4 +- qapi/audio.json | 44 ++ qemu-options.hx | 21 + scripts/meson-buildoptions.sh | 8 +- 10 files changed, 1011 insertions(+), 3 deletions(-) create mode 100644 audio/pwaudio.c