| Message ID | 20230216082544.119313-1-dbassey@redhat.com (mailing list archive) |
|---|---|
| State | New, archived |
| Headers | show |
| Series | [v2] audio/pwaudio.c: Add Pipewire audio backend for QEMU | expand |
On Thursday, February 16, 2023 9:25:44 AM CET Dorinda Bassey wrote: > This commit adds a new audiodev backend to allow QEMU to use Pipewire as both an audio sink and source. > Please wrap commit log. > Signed-off-by: Dorinda Bassey <dbassey@redhat.com> > --- > v2: > * Shorten commit message > * fix copyright ownership and authour > * use QEMU standard of 4 space indentation > * verbose use of pipewire instead pf pw > > audio/audio.c | 3 + > audio/audio_template.h | 4 + > audio/meson.build | 1 + > audio/pwaudio.c | 818 ++++++++++++++++++++++++++++++++++ > meson.build | 7 + > meson_options.txt | 4 +- > qapi/audio.json | 45 ++ > qemu-options.hx | 17 + > scripts/meson-buildoptions.sh | 8 +- > 9 files changed, 904 insertions(+), 3 deletions(-) > create mode 100644 audio/pwaudio.c > > diff --git a/audio/audio.c b/audio/audio.c > index 4290309d18..aa55e41ad8 100644 > --- a/audio/audio.c > +++ b/audio/audio.c > @@ -2069,6 +2069,9 @@ void audio_create_pdos(Audiodev *dev) > #ifdef CONFIG_AUDIO_PA > CASE(PA, pa, Pa); > #endif > +#ifdef CONFIG_AUDIO_PIPEWIRE > + CASE(PIPEWIRE, pipewire, Pipewire); > +#endif > #ifdef CONFIG_AUDIO_SDL > CASE(SDL, sdl, Sdl); > #endif > diff --git a/audio/audio_template.h b/audio/audio_template.h > index 42b4712acb..0f02afb921 100644 > --- a/audio/audio_template.h > +++ b/audio/audio_template.h > @@ -355,6 +355,10 @@ AudiodevPerDirectionOptions *glue(audio_get_pdo_, TYPE)(Audiodev *dev) > case AUDIODEV_DRIVER_PA: > return qapi_AudiodevPaPerDirectionOptions_base(dev->u.pa.TYPE); > #endif > +#ifdef CONFIG_AUDIO_PIPEWIRE > + case AUDIODEV_DRIVER_PIPEWIRE: > + return qapi_AudiodevPipewirePerDirectionOptions_base(dev->u.pipewire.TYPE); > +#endif > #ifdef CONFIG_AUDIO_SDL > case AUDIODEV_DRIVER_SDL: > return qapi_AudiodevSdlPerDirectionOptions_base(dev->u.sdl.TYPE); > diff --git a/audio/meson.build b/audio/meson.build > index 0722224ba9..65a49c1a10 100644 > --- a/audio/meson.build > +++ b/audio/meson.build > @@ -19,6 +19,7 @@ foreach m : [ > ['sdl', sdl, files('sdlaudio.c')], > ['jack', jack, files('jackaudio.c')], > ['sndio', sndio, files('sndioaudio.c')], > + ['pipewire', pipewire, files('pwaudio.c')], > ['spice', spice, files('spiceaudio.c')] > ] > if m[1].found() > diff --git a/audio/pwaudio.c b/audio/pwaudio.c > new file mode 100644 > index 0000000000..bb25133414 > --- /dev/null > +++ b/audio/pwaudio.c > @@ -0,0 +1,818 @@ > +/* > + * QEMU Pipewire audio driver > + * > + * Copyright (c) 2023 Red Hat Inc. > + * > + * Author: Dorinda Bassey <dbassey@redhat.com> > + * > + * Permission is hereby granted, free of charge, to any person obtaining a copy > + * of this software and associated documentation files (the "Software"), to deal > + * in the Software without restriction, including without limitation the rights > + * to use, copy, modify, merge, publish, distribute, sublicense, and/or sell > + * copies of the Software, and to permit persons to whom the Software is > + * furnished to do so, subject to the following conditions: > + * > + * The above copyright notice and this permission notice shall be included in > + * all copies or substantial portions of the Software. > + * > + * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR > + * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY, > + * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL > + * THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER > + * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM, > + * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN > + * THE SOFTWARE. > + */ > + > +#include "qemu/osdep.h" > +#include "qemu/module.h" > +#include "audio.h" > +#include <errno.h> > +#include <spa/param/audio/format-utils.h> > +#include <spa/utils/ringbuffer.h> > +#include <spa/utils/result.h> > + > +#include <pipewire/pipewire.h> > + > +#define AUDIO_CAP "pipewire" > +#define RINGBUFFER_SIZE (1u << 22) > +#define RINGBUFFER_MASK (RINGBUFFER_SIZE - 1) > +#define BUFFER_SAMPLES 128 BUFFER_SAMPLES is not used anywhere, and in code you are using 512 as literals instead. > + > +#include "audio_int.h" > + > +enum { > + MODE_SINK, > + MODE_SOURCE > +}; > + > +typedef struct pwaudio { > + Audiodev *dev; > + struct pw_thread_loop *thread_loop; > + struct pw_context *context; > + > + struct pw_core *core; > + struct spa_hook core_listener; > + int seq; > +} pwaudio; > + > +typedef struct PWVoice { > + pwaudio *g; > + bool enabled; > + struct pw_stream *stream; > + struct spa_hook stream_listener; > + struct spa_audio_info_raw info; > + uint32_t frame_size; > + struct spa_ringbuffer ring; > + uint8_t buffer[RINGBUFFER_SIZE]; s/buffer/ringbuffer/ maybe? > + > + uint32_t mode; > + struct pw_properties *props; > +} PWVoice; > + > +typedef struct PWVoiceOut { > + HWVoiceOut hw; > + PWVoice v; > +} PWVoiceOut; > + > +typedef struct PWVoiceIn { > + HWVoiceIn hw; > + PWVoice v; > +} PWVoiceIn; > + > +static void > +stream_destroy(void *data) > +{ > + PWVoice *v = (PWVoice *) data; > + spa_hook_remove(&v->stream_listener); > + v->stream = NULL; > +} > + > +/* output data processing function to read stuffs from the buffer */ > +static void > +playback_on_process(void *data) > +{ > + PWVoice *v = (PWVoice *) data; > + void *p; > + struct pw_buffer *b; > + struct spa_buffer *buf; > + uint32_t n_frames, req, index, n_bytes; > + int32_t avail; > + if (!v->stream) { return; } As pw_stream_dequeue_buffer() apparently can't cope with NULL. > + /* obtain a buffer to read from */ > + b = pw_stream_dequeue_buffer(v->stream); > + if (b == NULL) { > + pw_log_warn("out of buffers: %m"); > + return; > + } > + > + buf = b->buffer; > + p = buf->datas[0].data; > + if (p == NULL) { > + return; > + } > + req = b->requested * v->frame_size; > + if (req == 0) { > + req = 4096 * v->frame_size; > + } Why exactly 4k? > + n_frames = SPA_MIN(req, buf->datas[0].maxsize); > + n_bytes = n_frames * v->frame_size; > + > + /* get no of available bytes to read data from buffer */ > + > + avail = spa_ringbuffer_get_read_index(&v->ring, &index); > + > + if (!v->enabled) { > + avail = 0; > + } > + > + if (avail == 0) { > + memset(p, 0, n_bytes); > + } else { > + if (avail < (int32_t) n_bytes) { > + n_bytes = avail; > + } > + > + spa_ringbuffer_read_data(&v->ring, > + v->buffer, RINGBUFFER_SIZE, > + index & RINGBUFFER_MASK, p, n_bytes); > + > + index += n_bytes; > + spa_ringbuffer_read_update(&v->ring, index); > + } > + > + buf->datas[0].chunk->offset = 0; > + buf->datas[0].chunk->stride = v->frame_size; > + buf->datas[0].chunk->size = n_bytes; > + > + /* queue the buffer for playback */ > + pw_stream_queue_buffer(v->stream, b); > +} > + > +/* output data processing function to generate stuffs in the buffer */ > +static void > +capture_on_process(void *data) > +{ > + PWVoice *v = (PWVoice *) data; > + void *p; > + struct pw_buffer *b; > + struct spa_buffer *buf; > + int32_t filled; > + uint32_t index, offs, n_bytes; > + if (!v->stream) { return; } > + /* obtain a buffer */ > + b = pw_stream_dequeue_buffer(v->stream); > + if (b == NULL) { > + pw_log_warn("out of buffers: %m"); > + return; > + } > + > + /* Write data into buffer */ > + buf = b->buffer; > + p = buf->datas[0].data; > + if (p == NULL) { > + return; > + } > + offs = SPA_MIN(buf->datas[0].chunk->offset, buf->datas[0].maxsize); > + n_bytes = SPA_MIN(buf->datas[0].chunk->size, buf->datas[0].maxsize - offs); > + > + filled = spa_ringbuffer_get_write_index(&v->ring, &index); > + > + if (!v->enabled) { > + n_bytes = 0; > + } > + > + if (filled < 0) { > + pw_log_warn("%p: underrun write:%u filled:%d", p, index, filled); > + } else { > + if ((uint32_t) filled + n_bytes > RINGBUFFER_SIZE) { > + pw_log_warn("%p: overrun write:%u filled:%d + size:%u > max:%u", > + p, index, filled, n_bytes, RINGBUFFER_SIZE); > + } > + } > + spa_ringbuffer_write_data(&v->ring, > + v->buffer, RINGBUFFER_SIZE, > + index & RINGBUFFER_MASK, > + SPA_PTROFF(p, offs, void), n_bytes); > + index += n_bytes; > + spa_ringbuffer_write_update(&v->ring, index); > + > + /* queue the buffer for playback */ > + pw_stream_queue_buffer(v->stream, b); > +} > + > +static void > +on_stream_state_changed(void *_data, enum pw_stream_state old, > + enum pw_stream_state state, const char *error) > +{ > + PWVoice *v = (PWVoice *) _data; > + > + printf("stream state: \"%s\"\n", pw_stream_state_as_string(state)); > + > + switch (state) { > + case PW_STREAM_STATE_ERROR: > + case PW_STREAM_STATE_UNCONNECTED: > + { > + break; > + } > + case PW_STREAM_STATE_PAUSED: > + printf("node id: %d\n", pw_stream_get_node_id(v->stream)); > + break; > + case PW_STREAM_STATE_CONNECTING: > + case PW_STREAM_STATE_STREAMING: > + break; > + } > +} > + > +static const struct pw_stream_events capture_stream_events = { > + PW_VERSION_STREAM_EVENTS, > + .destroy = stream_destroy, > + .state_changed = on_stream_state_changed, > + .process = capture_on_process > +}; > + > +static const struct pw_stream_events playback_stream_events = { > + PW_VERSION_STREAM_EVENTS, > + .destroy = stream_destroy, > + .state_changed = on_stream_state_changed, > + .process = playback_on_process > +}; > + > +static size_t > +qpw_read(HWVoiceIn *hw, void *data, size_t len) > +{ > + PWVoiceIn *pw = (PWVoiceIn *) hw; > + PWVoice *v = &pw->v; > + pwaudio *c = v->g; > + const char *error = NULL; > + size_t l; > + int32_t avail; > + uint32_t index; > + > + pw_thread_loop_lock(c->thread_loop); > + if (pw_stream_get_state(v->stream, &error) != PW_STREAM_STATE_STREAMING) { > + /* wait for stream to become ready */ > + l = 0; > + goto done_unlock; > + } > + /* get no of available bytes to read data from buffer */ > + avail = spa_ringbuffer_get_read_index(&v->ring, &index); > + > + if (avail < (int32_t) len) { > + len = avail; > + } > + > + spa_ringbuffer_read_data(&v->ring, > + v->buffer, RINGBUFFER_SIZE, > + index & RINGBUFFER_MASK, data, len); > + index += len; > + spa_ringbuffer_read_update(&v->ring, index); > + l = len; > + > +done_unlock: > + pw_thread_loop_unlock(c->thread_loop); > + return l; > +} > + > +static size_t > +qpw_write(HWVoiceOut *hw, void *data, size_t len) > +{ > + PWVoiceOut *pw = (PWVoiceOut *) hw; > + PWVoice *v = &pw->v; > + pwaudio *c = v->g; > + const char *error = NULL; > + size_t l; > + int32_t filled, avail; > + uint32_t index; > + > + pw_thread_loop_lock(c->thread_loop); > + if (pw_stream_get_state(v->stream, &error) != PW_STREAM_STATE_STREAMING) { > + /* wait for stream to become ready */ > + l = 0; > + goto done_unlock; > + } > + filled = spa_ringbuffer_get_write_index(&v->ring, &index); > + > + avail = 512 * v->frame_size * 3 - filled; I would not use literals for period size and number of periods directly in code. Better use macros or constants instead. > + > + pw_log_debug("%u %u %u %zu", filled, avail, index, len); > + > + if (len > avail) { > + len = avail; > + } > + > + if (filled < 0) { > + pw_log_warn("%p: underrun write:%u filled:%d", pw, index, filled); > + } else { > + if ((uint32_t) filled + len > RINGBUFFER_SIZE) { > + pw_log_warn("%p: overrun write:%u filled:%d + size:%zu > max:%u", > + pw, index, filled, len, RINGBUFFER_SIZE); > + } > + } > + > + spa_ringbuffer_write_data(&v->ring, > + v->buffer, RINGBUFFER_SIZE, > + index & RINGBUFFER_MASK, data, len); > + index += len; > + spa_ringbuffer_write_update(&v->ring, index); > + l = len; > + > +done_unlock: > + pw_thread_loop_unlock(c->thread_loop); > + return l; > +} > + > +static int > +audfmt_to_pw(AudioFormat fmt, int endianness) > +{ > + int format; > + > + switch (fmt) { > + case AUDIO_FORMAT_S8: > + format = SPA_AUDIO_FORMAT_S8; > + break; > + case AUDIO_FORMAT_U8: > + format = SPA_AUDIO_FORMAT_U8; > + break; > + case AUDIO_FORMAT_S16: > + format = endianness ? SPA_AUDIO_FORMAT_S16_BE : SPA_AUDIO_FORMAT_S16_LE; > + break; > + case AUDIO_FORMAT_U16: > + format = endianness ? SPA_AUDIO_FORMAT_U16_BE : SPA_AUDIO_FORMAT_U16_LE; > + break; > + case AUDIO_FORMAT_S32: > + format = endianness ? SPA_AUDIO_FORMAT_S32_BE : SPA_AUDIO_FORMAT_S32_LE; > + break; > + case AUDIO_FORMAT_U32: > + format = endianness ? SPA_AUDIO_FORMAT_U32_BE : SPA_AUDIO_FORMAT_U32_LE; > + break; > + case AUDIO_FORMAT_F32: > + format = endianness ? SPA_AUDIO_FORMAT_F32_BE : SPA_AUDIO_FORMAT_F32_LE; > + break; > + default: > + dolog("Internal logic error: Bad audio format %d\n", fmt); > + format = SPA_AUDIO_FORMAT_U8; > + break; > + } > + return format; > +} > + > +static AudioFormat > +pw_to_audfmt(enum spa_audio_format fmt, int *endianness, > + uint32_t *frame_size) > +{ > + switch (fmt) { > + case SPA_AUDIO_FORMAT_S8: > + *frame_size = 1; > + return AUDIO_FORMAT_S8; > + case SPA_AUDIO_FORMAT_U8: > + *frame_size = 1; > + return AUDIO_FORMAT_U8; > + case SPA_AUDIO_FORMAT_S16_BE: > + *frame_size = 2; > + *endianness = 1; > + return AUDIO_FORMAT_S16; > + case SPA_AUDIO_FORMAT_S16_LE: > + *frame_size = 2; > + *endianness = 0; > + return AUDIO_FORMAT_S16; > + case SPA_AUDIO_FORMAT_U16_BE: > + *frame_size = 2; > + *endianness = 1; > + return AUDIO_FORMAT_U16; > + case SPA_AUDIO_FORMAT_U16_LE: > + *frame_size = 2; > + *endianness = 0; > + return AUDIO_FORMAT_U16; > + case SPA_AUDIO_FORMAT_S32_BE: > + *frame_size = 4; > + *endianness = 1; > + return AUDIO_FORMAT_S32; > + case SPA_AUDIO_FORMAT_S32_LE: > + *frame_size = 4; > + *endianness = 0; > + return AUDIO_FORMAT_S32; > + case SPA_AUDIO_FORMAT_U32_BE: > + *frame_size = 4; > + *endianness = 1; > + return AUDIO_FORMAT_U32; > + case SPA_AUDIO_FORMAT_U32_LE: > + *frame_size = 4; > + *endianness = 0; > + return AUDIO_FORMAT_U32; > + case SPA_AUDIO_FORMAT_F32_BE: > + *frame_size = 4; > + *endianness = 1; > + return AUDIO_FORMAT_F32; > + case SPA_AUDIO_FORMAT_F32_LE: > + *frame_size = 4; > + *endianness = 0; > + return AUDIO_FORMAT_F32; > + default: > + *frame_size = 1; > + dolog("Internal logic error: Bad spa_audio_format %d\n", fmt); > + return AUDIO_FORMAT_U8; > + } > +} > + > +static int > +create_stream(pwaudio *c, PWVoice *v, const char *name) > +{ > + int res; > + uint32_t n_params; > + const struct spa_pod *params[2]; > + uint8_t buffer[1024]; > + struct spa_pod_builder b; > + > + v->stream = pw_stream_new(c->core, name, NULL); > + > + if (v->stream == NULL) { > + res = -errno; > + goto error; > + } > + > + if (v->mode == MODE_SOURCE) { > + pw_stream_add_listener(v->stream, > + &v->stream_listener, &capture_stream_events, v); > + } else { > + pw_stream_add_listener(v->stream, > + &v->stream_listener, &playback_stream_events, v); > + } > + > + n_params = 0; > + spa_pod_builder_init(&b, buffer, sizeof(buffer)); > + params[n_params++] = spa_format_audio_raw_build(&b, > + SPA_PARAM_EnumFormat, > + &v->info); > + > + /* connect the stream to a sink or source */ > + res = pw_stream_connect(v->stream, > + v->mode == > + MODE_SOURCE ? PW_DIRECTION_INPUT : > + PW_DIRECTION_OUTPUT, PW_ID_ANY, > + PW_STREAM_FLAG_AUTOCONNECT | > + PW_STREAM_FLAG_MAP_BUFFERS | > + PW_STREAM_FLAG_RT_PROCESS, params, n_params); > + if (res < 0) { > + goto error; > + } > + > + return 0; > +error: > + return res; > +} > + > +static void > +pw_destroy(pwaudio *c) > +{ > + if (c->thread_loop) { > + pw_thread_loop_stop(c->thread_loop); > + } > + if (c->core) { > + pw_core_disconnect(c->core); > + } > + > + free(c); g_free(c); > +} > + > +static int > +qpw_stream_new(pwaudio *c, PWVoice *v, const char *name) > +{ > + int r; > + > + pw_thread_loop_lock(c->thread_loop); > + > + switch (v->info.channels) { > + case 8: > + v->info.position[0] = SPA_AUDIO_CHANNEL_FL; > + v->info.position[1] = SPA_AUDIO_CHANNEL_FR; > + v->info.position[2] = SPA_AUDIO_CHANNEL_FC; > + v->info.position[3] = SPA_AUDIO_CHANNEL_LFE; > + v->info.position[4] = SPA_AUDIO_CHANNEL_RL; > + v->info.position[5] = SPA_AUDIO_CHANNEL_RR; > + v->info.position[6] = SPA_AUDIO_CHANNEL_SL; > + v->info.position[7] = SPA_AUDIO_CHANNEL_SR; > + break; > + case 6: > + v->info.position[0] = SPA_AUDIO_CHANNEL_FL; > + v->info.position[1] = SPA_AUDIO_CHANNEL_FR; > + v->info.position[2] = SPA_AUDIO_CHANNEL_FC; > + v->info.position[3] = SPA_AUDIO_CHANNEL_LFE; > + v->info.position[4] = SPA_AUDIO_CHANNEL_RL; > + v->info.position[5] = SPA_AUDIO_CHANNEL_RR; > + break; > + case 5: > + v->info.position[0] = SPA_AUDIO_CHANNEL_FL; > + v->info.position[1] = SPA_AUDIO_CHANNEL_FR; > + v->info.position[2] = SPA_AUDIO_CHANNEL_FC; > + v->info.position[3] = SPA_AUDIO_CHANNEL_LFE; > + v->info.position[4] = SPA_AUDIO_CHANNEL_RC; > + break; > + case 4: > + v->info.position[0] = SPA_AUDIO_CHANNEL_FL; > + v->info.position[1] = SPA_AUDIO_CHANNEL_FR; > + v->info.position[2] = SPA_AUDIO_CHANNEL_FC; > + v->info.position[3] = SPA_AUDIO_CHANNEL_RC; > + break; > + case 3: > + v->info.position[0] = SPA_AUDIO_CHANNEL_FL; > + v->info.position[1] = SPA_AUDIO_CHANNEL_FR; > + v->info.position[2] = SPA_AUDIO_CHANNEL_LFE; > + break; > + case 2: > + v->info.position[0] = SPA_AUDIO_CHANNEL_FL; > + v->info.position[1] = SPA_AUDIO_CHANNEL_FR; > + break; > + case 1: > + v->info.position[0] = SPA_AUDIO_CHANNEL_MONO; > + break; > + default: > + for (size_t i = 0; i < v->info.channels; i++) { > + v->info.position[i] = SPA_AUDIO_CHANNEL_UNKNOWN; > + } > + break; > + } > + > + /* create a new unconnected pwstream */ > + r = create_stream(c, v, name); > + if (r < 0) { > + goto error; > + } > + > + pw_thread_loop_unlock(c->thread_loop); > + return r; > + > +error: > + AUD_log(AUDIO_CAP, "Failed to create stream."); > + pw_thread_loop_unlock(c->thread_loop); > + pw_destroy(c); > + return -1; > +} > + > +static int > +qpw_init_out(HWVoiceOut *hw, struct audsettings *as, void *drv_opaque) > +{ > + PWVoiceOut *pw = (PWVoiceOut *) hw; > + PWVoice *v = &pw->v; > + struct audsettings obt_as = *as; > + pwaudio *c = v->g = drv_opaque; > + AudiodevPipewireOptions *popts = &c->dev->u.pipewire; > + AudiodevPipewirePerDirectionOptions *ppdo = popts->out; > + int r; > + v->enabled = false; > + > + v->mode = MODE_SINK; > + > + pw_thread_loop_lock(c->thread_loop); > + > + v->info.format = audfmt_to_pw(as->fmt, as->endianness); > + v->info.channels = as->nchannels; > + v->info.rate = as->freq; > + > + obt_as.fmt = > + pw_to_audfmt(v->info.format, &obt_as.endianness, &v->frame_size); > + v->frame_size *= as->nchannels; > + > + /* call the function that creates a new stream for playback */ > + r = qpw_stream_new(c, v, ppdo->stream_name ? : c->dev->id); > + if (r < 0) { > + pw_log_error("qpw_stream_new for playback failed\n "); > + goto fail; > + } > + > + /* report the audio format we support */ > + audio_pcm_init_info(&hw->info, &obt_as); > + > + /* report the buffer size to qemu */ > + hw->samples = 512; > + > + pw_thread_loop_unlock(c->thread_loop); > + return 0; > +fail: > + pw_thread_loop_unlock(c->thread_loop); > + return -1; > +} > + > +static int > +qpw_init_in(HWVoiceIn *hw, struct audsettings *as, void *drv_opaque) > +{ > + PWVoiceIn *pw = (PWVoiceIn *) hw; > + PWVoice *v = &pw->v; > + struct audsettings obt_as = *as; > + pwaudio *c = v->g = drv_opaque; > + AudiodevPipewireOptions *popts = &c->dev->u.pipewire; > + AudiodevPipewirePerDirectionOptions *ppdo = popts->in; > + int r; > + v->enabled = false; > + > + v->mode = MODE_SOURCE; > + pw_thread_loop_lock(c->thread_loop); > + > + v->info.format = audfmt_to_pw(as->fmt, as->endianness); > + v->info.channels = as->nchannels; > + v->info.rate = as->freq; > + > + obt_as.fmt = > + pw_to_audfmt(v->info.format, &obt_as.endianness, &v->frame_size); > + v->frame_size *= as->nchannels; > + > + /* call the function that creates a new stream for recording */ > + r = qpw_stream_new(c, v, ppdo->stream_name ? : c->dev->id); > + if (r < 0) { > + pw_log_error("qpw_stream_new for recording failed\n "); > + goto fail; > + } > + > + /* report the audio format we support */ > + audio_pcm_init_info(&hw->info, &obt_as); > + > + /* report the buffer size to qemu */ > + hw->samples = 512; > + > + pw_thread_loop_unlock(c->thread_loop); > + return 0; > +fail: > + pw_thread_loop_unlock(c->thread_loop); > + return -1; > +} > + > +static void > +qpw_fini_out(HWVoiceOut *hw) > +{ > + PWVoiceOut *pw = (PWVoiceOut *) hw; > + PWVoice *v = &pw->v; > + > + if (v->stream) { > + pwaudio *c = v->g; > + pw_thread_loop_lock(c->thread_loop); > + pw_stream_destroy(v->stream); > + v->stream = NULL; > + pw_thread_loop_unlock(c->thread_loop); > + } > +} > + > +static void > +qpw_fini_in(HWVoiceIn *hw) > +{ > + PWVoiceIn *pw = (PWVoiceIn *) hw; > + PWVoice *v = &pw->v; > + > + if (v->stream) { > + pwaudio *c = v->g; > + pw_thread_loop_lock(c->thread_loop); > + pw_stream_destroy(v->stream); > + v->stream = NULL; > + pw_thread_loop_unlock(c->thread_loop); > + } > +} > + > +static void > +qpw_enable_out(HWVoiceOut *hw, bool enable) > +{ > + PWVoiceOut *po = (PWVoiceOut *) hw; > + PWVoice *v = &po->v; > + v->enabled = enable; > +} > + > +static void > +qpw_enable_in(HWVoiceIn *hw, bool enable) > +{ > + PWVoiceIn *pi = (PWVoiceIn *) hw; > + PWVoice *v = &pi->v; > + v->enabled = enable; > +} > + > +static void > +on_core_error(void *data, uint32_t id, int seq, int res, const char *message) > +{ > + pwaudio *pw = data; > + > + pw_log_warn("error id:%u seq:%d res:%d (%s): %s", > + id, seq, res, spa_strerror(res), message); > + > + pw_thread_loop_signal(pw->thread_loop, FALSE); > +} > + > +static void > +on_core_done(void *data, uint32_t id, int seq) > +{ > + pwaudio *pw = data; > + if (id == PW_ID_CORE) { > + pw->seq = seq; > + pw_thread_loop_signal(pw->thread_loop, FALSE); > + } > +} > + > +static const struct pw_core_events core_events = { > + PW_VERSION_CORE_EVENTS, > + .done = on_core_done, > + .error = on_core_error, > +}; > + > +static void * > +qpw_audio_init(Audiodev *dev) > +{ > + pwaudio *pw; > + pw = g_new0(pwaudio, 1); > + pw_init(NULL, NULL); > + > + AudiodevPipewireOptions *popts; > + AUD_log(AUDIO_CAP, "Initialize PW context\n"); > + assert(dev->driver == AUDIODEV_DRIVER_PIPEWIRE); > + popts = &dev->u.pipewire; > + > + if (!popts->has_latency) { > + popts->has_latency = true; > + popts->latency = 44100; > + } Why 44ms? > + > + pw->dev = dev; > + pw->thread_loop = pw_thread_loop_new("Pipewire thread loop", NULL); > + if (pw->thread_loop == NULL) { > + goto fail; > + } > + pw->context = > + pw_context_new(pw_thread_loop_get_loop(pw->thread_loop), NULL, 0); > + > + if (pw_thread_loop_start(pw->thread_loop) < 0) { > + goto fail; > + } > + > + pw_thread_loop_lock(pw->thread_loop); > + > + pw->core = pw_context_connect(pw->context, NULL, 0); > + if (pw->core == NULL) { > + goto fail; > + } > + > + pw_core_add_listener(pw->core, &pw->core_listener, &core_events, pw); > + > + pw_thread_loop_unlock(pw->thread_loop); > + > + return pw; > + > +fail: > + AUD_log(AUDIO_CAP, "Failed to initialize PW context"); > + pw_thread_loop_unlock(pw->thread_loop); > + pw_context_destroy(pw->context); > + pw_thread_loop_destroy(pw->thread_loop); > + g_free(pw); > + return NULL; > +} > + > +static void > +qpw_audio_fini(void *opaque) > +{ > + pwaudio *pw = opaque; > + > + pw_thread_loop_stop(pw->thread_loop); > + > + if (pw->core) { > + spa_hook_remove(&pw->core_listener); > + spa_zero(pw->core_listener); > + pw_core_disconnect(pw->core); > + } > + > + if (pw->context) { > + pw_context_destroy(pw->context); > + } > + pw_thread_loop_destroy(pw->thread_loop); > + > + g_free(pw); > +} > + > +static struct audio_pcm_ops qpw_pcm_ops = { > + .init_out = qpw_init_out, > + .fini_out = qpw_fini_out, > + .write = qpw_write, > + .buffer_get_free = audio_generic_buffer_get_free, > + .run_buffer_out = audio_generic_run_buffer_out, > + .enable_out = qpw_enable_out, > + > + .init_in = qpw_init_in, > + .fini_in = qpw_fini_in, > + .read = qpw_read, > + .run_buffer_in = audio_generic_run_buffer_in, > + .enable_in = qpw_enable_in > +}; > + > +static struct audio_driver pw_audio_driver = { > + .name = "pipewire", > + .descr = "http://www.pipewire.org/", > + .init = qpw_audio_init, > + .fini = qpw_audio_fini, > + .pcm_ops = &qpw_pcm_ops, > + .can_be_default = 1, > + .max_voices_out = INT_MAX, > + .max_voices_in = INT_MAX, > + .voice_size_out = sizeof(PWVoiceOut), > + .voice_size_in = sizeof(PWVoiceIn), > +}; > + > +static void > +register_audio_pw(void) > +{ > + audio_driver_register(&pw_audio_driver); > +} > + > +type_init(register_audio_pw); > diff --git a/meson.build b/meson.build > index a76c855312..686fdd5b81 100644 > --- a/meson.build > +++ b/meson.build > @@ -734,6 +734,11 @@ 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', required: get_option('pipewire'), > + method: 'pkg-config', kwargs: static_kwargs) > +endif > sndio = not_found > if not get_option('sndio').auto() or have_system > sndio = dependency('sndio', required: get_option('sndio'), > @@ -1671,6 +1676,7 @@ if have_system > 'jack': jack.found(), > 'oss': oss.found(), > 'pa': pulse.found(), > + 'pipewire': pipewire.found(), > 'sdl': sdl.found(), > 'sndio': sndio.found(), > } > @@ -3949,6 +3955,7 @@ endif > if targetos == 'linux' > summary_info += {'ALSA support': alsa} > summary_info += {'PulseAudio support': pulse} > + summary_info += {'Pipewire support': pipewire} > endif > summary_info += {'JACK support': jack} > summary_info += {'brlapi support': brlapi} > diff --git a/meson_options.txt b/meson_options.txt > index 7e5801db90..1b7847250d 100644 > --- a/meson_options.txt > +++ b/meson_options.txt > @@ -21,7 +21,7 @@ option('tls_priority', type : 'string', value : 'NORMAL', > option('default_devices', type : 'boolean', value : true, > description: 'Include a default selection of devices in emulators') > option('audio_drv_list', type: 'array', value: ['default'], > - choices: ['alsa', 'coreaudio', 'default', 'dsound', 'jack', 'oss', 'pa', 'sdl', 'sndio'], > + choices: ['alsa', 'coreaudio', 'default', 'dsound', 'jack', 'oss', 'pa', 'pipewire', 'sdl', 'sndio'], > description: 'Set audio driver list') > option('block_drv_rw_whitelist', type : 'string', value : '', > description: 'set block driver read-write whitelist (by default affects only QEMU, not tools like qemu-img)') > @@ -255,6 +255,8 @@ option('oss', type: 'feature', value: 'auto', > description: 'OSS sound support') > option('pa', type: 'feature', value: 'auto', > description: 'PulseAudio sound support') > +option('pipewire', type: 'feature', value: 'auto', > + description: 'Pipewire sound support') > option('sndio', type: 'feature', value: 'auto', > description: 'sndio sound support') > > diff --git a/qapi/audio.json b/qapi/audio.json > index 4e54c00f51..b872e9f10d 100644 > --- a/qapi/audio.json > +++ b/qapi/audio.json > @@ -324,6 +324,48 @@ > '*out': 'AudiodevPaPerDirectionOptions', > '*server': 'str' } } > > +## > +# @AudiodevPipewirePerDirectionOptions: > +# > +# Options of the Pipewire backend that are used for both playback and > +# recording. > +# > +# @name: name of the sink/source to use > +# > +# @stream-name: name of the Pipewire stream created by qemu. Can be > +# used to identify the stream in Pipewire when you > +# create multiple Pipewire devices or run multiple qemu > +# instances (default: audiodev's id, since 7.1) > +# > +# > +# Since: 7.2 > +## > +{ 'struct': 'AudiodevPipewirePerDirectionOptions', > + 'base': 'AudiodevPerDirectionOptions', > + 'data': { > + '*name': 'str', > + '*stream-name': 'str' } } > + > +## > +# @AudiodevPipewireOptions: > +# > +# Options of the Pipewire audio backend. > +# > +# @in: options of the capture stream > +# > +# @out: options of the playback stream > +# > +# @latency: add latency to playback in microseconds > +# (default 44100) > +# > +# Since: 7.2 > +## > +{ 'struct': 'AudiodevPipewireOptions', > + 'data': { > + '*in': 'AudiodevPipewirePerDirectionOptions', > + '*out': 'AudiodevPipewirePerDirectionOptions', > + '*latency': 'uint32' } } > + > ## > # @AudiodevSdlPerDirectionOptions: > # > @@ -416,6 +458,7 @@ > { 'name': 'jack', 'if': 'CONFIG_AUDIO_JACK' }, > { 'name': 'oss', 'if': 'CONFIG_AUDIO_OSS' }, > { 'name': 'pa', 'if': 'CONFIG_AUDIO_PA' }, > + { 'name': 'pipewire', 'if': 'CONFIG_AUDIO_PIPEWIRE' }, > { 'name': 'sdl', 'if': 'CONFIG_AUDIO_SDL' }, > { 'name': 'sndio', 'if': 'CONFIG_AUDIO_SNDIO' }, > { 'name': 'spice', 'if': 'CONFIG_SPICE' }, > @@ -456,6 +499,8 @@ > 'if': 'CONFIG_AUDIO_OSS' }, > 'pa': { 'type': 'AudiodevPaOptions', > 'if': 'CONFIG_AUDIO_PA' }, > + 'pipewire': { 'type': 'AudiodevPipewireOptions', > + 'if': 'CONFIG_AUDIO_PIPEWIRE' }, > 'sdl': { 'type': 'AudiodevSdlOptions', > 'if': 'CONFIG_AUDIO_SDL' }, > 'sndio': { 'type': 'AudiodevSndioOptions', > diff --git a/qemu-options.hx b/qemu-options.hx > index 88e93c6103..bde4830fab 100644 > --- a/qemu-options.hx > +++ b/qemu-options.hx > @@ -779,6 +779,11 @@ DEF("audiodev", HAS_ARG, QEMU_OPTION_audiodev, > " in|out.name= source/sink device name\n" > " in|out.latency= desired latency in microseconds\n" > #endif > +#ifdef CONFIG_AUDIO_PIPEWIRE > + "-audiodev pipewire,id=id[,prop[=value][,...]]\n" > + " in|out.name= source/sink device name\n" > + " latency= desired latency in microseconds\n" > +#endif > #ifdef CONFIG_AUDIO_SDL > "-audiodev sdl,id=id[,prop[=value][,...]]\n" > " in|out.buffer-count= number of buffers\n" > @@ -942,6 +947,18 @@ SRST > Desired latency in microseconds. The PulseAudio server will try > to honor this value but actual latencies may be lower or higher. > > +``-audiodev pipewire,id=id[,prop[=value][,...]]`` > + Creates a backend using Pipewire. This backend is available on > + most systems. > + > + Pipewire specific options are: > + > + ``latency=latency`` > + Add extra latency to playback in microseconds > + > + ``in|out.name=sink`` > + Use the specified source/sink for recording/playback. > + > ``-audiodev sdl,id=id[,prop[=value][,...]]`` > Creates a backend using SDL. This backend is available on most > systems, but you should use your platform's native backend if > diff --git a/scripts/meson-buildoptions.sh b/scripts/meson-buildoptions.sh > index 180c11665a..d9f6525346 100644 > --- a/scripts/meson-buildoptions.sh > +++ b/scripts/meson-buildoptions.sh > @@ -1,7 +1,8 @@ > # This file is generated by meson-buildoptions.py, do not edit! > meson_options_help() { > - printf "%s\n" ' --audio-drv-list=CHOICES Set audio driver list [default] (choices: alsa/co' > - printf "%s\n" ' reaudio/default/dsound/jack/oss/pa/sdl/sndio)' > + printf "%s\n" ' --audio-drv-list=CHOICES Set audio driver list [default] (choices: al' > + printf "%s\n" ' sa/coreaudio/default/dsound/jack/oss/pa/' > + printf "%s\n" ' pipewire/sdl/sndio)' > printf "%s\n" ' --block-drv-ro-whitelist=VALUE' > printf "%s\n" ' set block driver read-only whitelist (by default' > printf "%s\n" ' affects only QEMU, not tools like qemu-img)' > @@ -135,6 +136,7 @@ meson_options_help() { > printf "%s\n" ' oss OSS sound support' > printf "%s\n" ' pa PulseAudio sound support' > printf "%s\n" ' parallels parallels image format support' > + printf "%s\n" ' pipewire Pipewire sound support' > printf "%s\n" ' png PNG support with libpng' > printf "%s\n" ' pvrdma Enable PVRDMA support' > printf "%s\n" ' qcow1 qcow1 image format support' > @@ -370,6 +372,8 @@ _meson_option_parse() { > --disable-pa) printf "%s" -Dpa=disabled ;; > --enable-parallels) printf "%s" -Dparallels=enabled ;; > --disable-parallels) printf "%s" -Dparallels=disabled ;; > + --enable-pipewire) printf "%s" -Dpipewire=enabled ;; > + --disable-pipewire) printf "%s" -Dpipewire=disabled ;; > --with-pkgversion=*) quote_sh "-Dpkgversion=$2" ;; > --enable-png) printf "%s" -Dpng=enabled ;; > --disable-png) printf "%s" -Dpng=disabled ;; >
> > BUFFER_SAMPLES is not used anywhere, and in code you are using 512 as > literals > instead. That was an oversight indeed, It's intended use was removed. s/buffer/ringbuffer/ maybe? > I think the naming convention 'buffer' is good. I would not use literals for period size and number of periods directly in > code. Better use macros or constants instead. > Noted, thanks. Why exactly 4k? > For playback streams, this size allows for more efficient streaming of audio data, as smaller chunks can lead to inaccuracies in sound quality. Why 44ms? > Thanks for spotting that, I had set its calculation to be Hz, because the default rate is between 44kHz to 48kHz, when actually the latency should be low as ~10ms latency (256 /48000 Hz). I would change that to 15ms which is fair for what a generic hardware can handle. BTW there's also the parameter to set the latency to desired value. On Thu, Feb 16, 2023 at 12:41 PM Christian Schoenebeck < qemu_oss@crudebyte.com> wrote: > On Thursday, February 16, 2023 9:25:44 AM CET Dorinda Bassey wrote: > > This commit adds a new audiodev backend to allow QEMU to use Pipewire as > both an audio sink and source. > > > > Please wrap commit log. > > > Signed-off-by: Dorinda Bassey <dbassey@redhat.com> > > --- > > v2: > > * Shorten commit message > > * fix copyright ownership and authour > > * use QEMU standard of 4 space indentation > > * verbose use of pipewire instead pf pw > > > > audio/audio.c | 3 + > > audio/audio_template.h | 4 + > > audio/meson.build | 1 + > > audio/pwaudio.c | 818 ++++++++++++++++++++++++++++++++++ > > meson.build | 7 + > > meson_options.txt | 4 +- > > qapi/audio.json | 45 ++ > > qemu-options.hx | 17 + > > scripts/meson-buildoptions.sh | 8 +- > > 9 files changed, 904 insertions(+), 3 deletions(-) > > create mode 100644 audio/pwaudio.c > > > > diff --git a/audio/audio.c b/audio/audio.c > > index 4290309d18..aa55e41ad8 100644 > > --- a/audio/audio.c > > +++ b/audio/audio.c > > @@ -2069,6 +2069,9 @@ void audio_create_pdos(Audiodev *dev) > > #ifdef CONFIG_AUDIO_PA > > CASE(PA, pa, Pa); > > #endif > > +#ifdef CONFIG_AUDIO_PIPEWIRE > > + CASE(PIPEWIRE, pipewire, Pipewire); > > +#endif > > #ifdef CONFIG_AUDIO_SDL > > CASE(SDL, sdl, Sdl); > > #endif > > diff --git a/audio/audio_template.h b/audio/audio_template.h > > index 42b4712acb..0f02afb921 100644 > > --- a/audio/audio_template.h > > +++ b/audio/audio_template.h > > @@ -355,6 +355,10 @@ AudiodevPerDirectionOptions *glue(audio_get_pdo_, > TYPE)(Audiodev *dev) > > case AUDIODEV_DRIVER_PA: > > return qapi_AudiodevPaPerDirectionOptions_base(dev->u.pa.TYPE); > > #endif > > +#ifdef CONFIG_AUDIO_PIPEWIRE > > + case AUDIODEV_DRIVER_PIPEWIRE: > > + return > qapi_AudiodevPipewirePerDirectionOptions_base(dev->u.pipewire.TYPE); > > +#endif > > #ifdef CONFIG_AUDIO_SDL > > case AUDIODEV_DRIVER_SDL: > > return > qapi_AudiodevSdlPerDirectionOptions_base(dev->u.sdl.TYPE); > > diff --git a/audio/meson.build b/audio/meson.build > > index 0722224ba9..65a49c1a10 100644 > > --- a/audio/meson.build > > +++ b/audio/meson.build > > @@ -19,6 +19,7 @@ foreach m : [ > > ['sdl', sdl, files('sdlaudio.c')], > > ['jack', jack, files('jackaudio.c')], > > ['sndio', sndio, files('sndioaudio.c')], > > + ['pipewire', pipewire, files('pwaudio.c')], > > ['spice', spice, files('spiceaudio.c')] > > ] > > if m[1].found() > > diff --git a/audio/pwaudio.c b/audio/pwaudio.c > > new file mode 100644 > > index 0000000000..bb25133414 > > --- /dev/null > > +++ b/audio/pwaudio.c > > @@ -0,0 +1,818 @@ > > +/* > > + * QEMU Pipewire audio driver > > + * > > + * Copyright (c) 2023 Red Hat Inc. > > + * > > + * Author: Dorinda Bassey <dbassey@redhat.com> > > + * > > + * Permission is hereby granted, free of charge, to any person > obtaining a copy > > + * of this software and associated documentation files (the > "Software"), to deal > > + * in the Software without restriction, including without limitation > the rights > > + * to use, copy, modify, merge, publish, distribute, sublicense, and/or > sell > > + * copies of the Software, and to permit persons to whom the Software is > > + * furnished to do so, subject to the following conditions: > > + * > > + * The above copyright notice and this permission notice shall be > included in > > + * all copies or substantial portions of the Software. > > + * > > + * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, > EXPRESS OR > > + * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF > MERCHANTABILITY, > > + * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT > SHALL > > + * THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR > OTHER > > + * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, > ARISING FROM, > > + * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER > DEALINGS IN > > + * THE SOFTWARE. > > + */ > > + > > +#include "qemu/osdep.h" > > +#include "qemu/module.h" > > +#include "audio.h" > > +#include <errno.h> > > +#include <spa/param/audio/format-utils.h> > > +#include <spa/utils/ringbuffer.h> > > +#include <spa/utils/result.h> > > + > > +#include <pipewire/pipewire.h> > > + > > +#define AUDIO_CAP "pipewire" > > +#define RINGBUFFER_SIZE (1u << 22) > > +#define RINGBUFFER_MASK (RINGBUFFER_SIZE - 1) > > +#define BUFFER_SAMPLES 128 > > BUFFER_SAMPLES is not used anywhere, and in code you are using 512 as > literals > instead. > > > + > > +#include "audio_int.h" > > + > > +enum { > > + MODE_SINK, > > + MODE_SOURCE > > +}; > > + > > +typedef struct pwaudio { > > + Audiodev *dev; > > + struct pw_thread_loop *thread_loop; > > + struct pw_context *context; > > + > > + struct pw_core *core; > > + struct spa_hook core_listener; > > + int seq; > > +} pwaudio; > > + > > +typedef struct PWVoice { > > + pwaudio *g; > > + bool enabled; > > + struct pw_stream *stream; > > + struct spa_hook stream_listener; > > + struct spa_audio_info_raw info; > > + uint32_t frame_size; > > + struct spa_ringbuffer ring; > > + uint8_t buffer[RINGBUFFER_SIZE]; > > s/buffer/ringbuffer/ maybe? > > > + > > + uint32_t mode; > > + struct pw_properties *props; > > +} PWVoice; > > + > > +typedef struct PWVoiceOut { > > + HWVoiceOut hw; > > + PWVoice v; > > +} PWVoiceOut; > > + > > +typedef struct PWVoiceIn { > > + HWVoiceIn hw; > > + PWVoice v; > > +} PWVoiceIn; > > + > > +static void > > +stream_destroy(void *data) > > +{ > > + PWVoice *v = (PWVoice *) data; > > + spa_hook_remove(&v->stream_listener); > > + v->stream = NULL; > > +} > > + > > +/* output data processing function to read stuffs from the buffer */ > > +static void > > +playback_on_process(void *data) > > +{ > > + PWVoice *v = (PWVoice *) data; > > + void *p; > > + struct pw_buffer *b; > > + struct spa_buffer *buf; > > + uint32_t n_frames, req, index, n_bytes; > > + int32_t avail; > > + > > if (!v->stream) { > return; > } > > As pw_stream_dequeue_buffer() apparently can't cope with NULL. > > > + /* obtain a buffer to read from */ > > + b = pw_stream_dequeue_buffer(v->stream); > > + if (b == NULL) { > > + pw_log_warn("out of buffers: %m"); > > + return; > > + } > > + > > + buf = b->buffer; > > + p = buf->datas[0].data; > > + if (p == NULL) { > > + return; > > + } > > + req = b->requested * v->frame_size; > > + if (req == 0) { > > + req = 4096 * v->frame_size; > > + } > > Why exactly 4k? > > > + n_frames = SPA_MIN(req, buf->datas[0].maxsize); > > + n_bytes = n_frames * v->frame_size; > > + > > + /* get no of available bytes to read data from buffer */ > > + > > + avail = spa_ringbuffer_get_read_index(&v->ring, &index); > > + > > + if (!v->enabled) { > > + avail = 0; > > + } > > + > > + if (avail == 0) { > > + memset(p, 0, n_bytes); > > + } else { > > + if (avail < (int32_t) n_bytes) { > > + n_bytes = avail; > > + } > > + > > + spa_ringbuffer_read_data(&v->ring, > > + v->buffer, RINGBUFFER_SIZE, > > + index & RINGBUFFER_MASK, p, > n_bytes); > > + > > + index += n_bytes; > > + spa_ringbuffer_read_update(&v->ring, index); > > + } > > + > > + buf->datas[0].chunk->offset = 0; > > + buf->datas[0].chunk->stride = v->frame_size; > > + buf->datas[0].chunk->size = n_bytes; > > + > > + /* queue the buffer for playback */ > > + pw_stream_queue_buffer(v->stream, b); > > +} > > + > > +/* output data processing function to generate stuffs in the buffer */ > > +static void > > +capture_on_process(void *data) > > +{ > > + PWVoice *v = (PWVoice *) data; > > + void *p; > > + struct pw_buffer *b; > > + struct spa_buffer *buf; > > + int32_t filled; > > + uint32_t index, offs, n_bytes; > > + > > if (!v->stream) { > return; > } > > > + /* obtain a buffer */ > > + b = pw_stream_dequeue_buffer(v->stream); > > + if (b == NULL) { > > + pw_log_warn("out of buffers: %m"); > > + return; > > + } > > + > > + /* Write data into buffer */ > > + buf = b->buffer; > > + p = buf->datas[0].data; > > + if (p == NULL) { > > + return; > > + } > > + offs = SPA_MIN(buf->datas[0].chunk->offset, buf->datas[0].maxsize); > > + n_bytes = SPA_MIN(buf->datas[0].chunk->size, buf->datas[0].maxsize > - offs); > > + > > + filled = spa_ringbuffer_get_write_index(&v->ring, &index); > > + > > + if (!v->enabled) { > > + n_bytes = 0; > > + } > > + > > + if (filled < 0) { > > + pw_log_warn("%p: underrun write:%u filled:%d", p, index, > filled); > > + } else { > > + if ((uint32_t) filled + n_bytes > RINGBUFFER_SIZE) { > > + pw_log_warn("%p: overrun write:%u filled:%d + size:%u > > max:%u", > > + p, index, filled, n_bytes, RINGBUFFER_SIZE); > > + } > > + } > > + spa_ringbuffer_write_data(&v->ring, > > + v->buffer, RINGBUFFER_SIZE, > > + index & RINGBUFFER_MASK, > > + SPA_PTROFF(p, offs, void), n_bytes); > > + index += n_bytes; > > + spa_ringbuffer_write_update(&v->ring, index); > > + > > + /* queue the buffer for playback */ > > + pw_stream_queue_buffer(v->stream, b); > > +} > > + > > +static void > > +on_stream_state_changed(void *_data, enum pw_stream_state old, > > + enum pw_stream_state state, const char *error) > > +{ > > + PWVoice *v = (PWVoice *) _data; > > + > > + printf("stream state: \"%s\"\n", pw_stream_state_as_string(state)); > > + > > + switch (state) { > > + case PW_STREAM_STATE_ERROR: > > + case PW_STREAM_STATE_UNCONNECTED: > > + { > > + break; > > + } > > + case PW_STREAM_STATE_PAUSED: > > + printf("node id: %d\n", pw_stream_get_node_id(v->stream)); > > + break; > > + case PW_STREAM_STATE_CONNECTING: > > + case PW_STREAM_STATE_STREAMING: > > + break; > > + } > > +} > > + > > +static const struct pw_stream_events capture_stream_events = { > > + PW_VERSION_STREAM_EVENTS, > > + .destroy = stream_destroy, > > + .state_changed = on_stream_state_changed, > > + .process = capture_on_process > > +}; > > + > > +static const struct pw_stream_events playback_stream_events = { > > + PW_VERSION_STREAM_EVENTS, > > + .destroy = stream_destroy, > > + .state_changed = on_stream_state_changed, > > + .process = playback_on_process > > +}; > > + > > +static size_t > > +qpw_read(HWVoiceIn *hw, void *data, size_t len) > > +{ > > + PWVoiceIn *pw = (PWVoiceIn *) hw; > > + PWVoice *v = &pw->v; > > + pwaudio *c = v->g; > > + const char *error = NULL; > > + size_t l; > > + int32_t avail; > > + uint32_t index; > > + > > + pw_thread_loop_lock(c->thread_loop); > > + if (pw_stream_get_state(v->stream, &error) != > PW_STREAM_STATE_STREAMING) { > > + /* wait for stream to become ready */ > > + l = 0; > > + goto done_unlock; > > + } > > + /* get no of available bytes to read data from buffer */ > > + avail = spa_ringbuffer_get_read_index(&v->ring, &index); > > + > > + if (avail < (int32_t) len) { > > + len = avail; > > + } > > + > > + spa_ringbuffer_read_data(&v->ring, > > + v->buffer, RINGBUFFER_SIZE, > > + index & RINGBUFFER_MASK, data, len); > > + index += len; > > + spa_ringbuffer_read_update(&v->ring, index); > > + l = len; > > + > > +done_unlock: > > + pw_thread_loop_unlock(c->thread_loop); > > + return l; > > +} > > + > > +static size_t > > +qpw_write(HWVoiceOut *hw, void *data, size_t len) > > +{ > > + PWVoiceOut *pw = (PWVoiceOut *) hw; > > + PWVoice *v = &pw->v; > > + pwaudio *c = v->g; > > + const char *error = NULL; > > + size_t l; > > + int32_t filled, avail; > > + uint32_t index; > > + > > + pw_thread_loop_lock(c->thread_loop); > > + if (pw_stream_get_state(v->stream, &error) != > PW_STREAM_STATE_STREAMING) { > > + /* wait for stream to become ready */ > > + l = 0; > > + goto done_unlock; > > + } > > + filled = spa_ringbuffer_get_write_index(&v->ring, &index); > > + > > + avail = 512 * v->frame_size * 3 - filled; > > I would not use literals for period size and number of periods directly in > code. Better use macros or constants instead. > > > + > > + pw_log_debug("%u %u %u %zu", filled, avail, index, len); > > + > > + if (len > avail) { > > + len = avail; > > + } > > + > > + if (filled < 0) { > > + pw_log_warn("%p: underrun write:%u filled:%d", pw, index, > filled); > > + } else { > > + if ((uint32_t) filled + len > RINGBUFFER_SIZE) { > > + pw_log_warn("%p: overrun write:%u filled:%d + size:%zu > > max:%u", > > + pw, index, filled, len, RINGBUFFER_SIZE); > > + } > > + } > > + > > + spa_ringbuffer_write_data(&v->ring, > > + v->buffer, RINGBUFFER_SIZE, > > + index & RINGBUFFER_MASK, data, len); > > + index += len; > > + spa_ringbuffer_write_update(&v->ring, index); > > + l = len; > > + > > +done_unlock: > > + pw_thread_loop_unlock(c->thread_loop); > > + return l; > > +} > > + > > +static int > > +audfmt_to_pw(AudioFormat fmt, int endianness) > > +{ > > + int format; > > + > > + switch (fmt) { > > + case AUDIO_FORMAT_S8: > > + format = SPA_AUDIO_FORMAT_S8; > > + break; > > + case AUDIO_FORMAT_U8: > > + format = SPA_AUDIO_FORMAT_U8; > > + break; > > + case AUDIO_FORMAT_S16: > > + format = endianness ? SPA_AUDIO_FORMAT_S16_BE : > SPA_AUDIO_FORMAT_S16_LE; > > + break; > > + case AUDIO_FORMAT_U16: > > + format = endianness ? SPA_AUDIO_FORMAT_U16_BE : > SPA_AUDIO_FORMAT_U16_LE; > > + break; > > + case AUDIO_FORMAT_S32: > > + format = endianness ? SPA_AUDIO_FORMAT_S32_BE : > SPA_AUDIO_FORMAT_S32_LE; > > + break; > > + case AUDIO_FORMAT_U32: > > + format = endianness ? SPA_AUDIO_FORMAT_U32_BE : > SPA_AUDIO_FORMAT_U32_LE; > > + break; > > + case AUDIO_FORMAT_F32: > > + format = endianness ? SPA_AUDIO_FORMAT_F32_BE : > SPA_AUDIO_FORMAT_F32_LE; > > + break; > > + default: > > + dolog("Internal logic error: Bad audio format %d\n", fmt); > > + format = SPA_AUDIO_FORMAT_U8; > > + break; > > + } > > + return format; > > +} > > + > > +static AudioFormat > > +pw_to_audfmt(enum spa_audio_format fmt, int *endianness, > > + uint32_t *frame_size) > > +{ > > + switch (fmt) { > > + case SPA_AUDIO_FORMAT_S8: > > + *frame_size = 1; > > + return AUDIO_FORMAT_S8; > > + case SPA_AUDIO_FORMAT_U8: > > + *frame_size = 1; > > + return AUDIO_FORMAT_U8; > > + case SPA_AUDIO_FORMAT_S16_BE: > > + *frame_size = 2; > > + *endianness = 1; > > + return AUDIO_FORMAT_S16; > > + case SPA_AUDIO_FORMAT_S16_LE: > > + *frame_size = 2; > > + *endianness = 0; > > + return AUDIO_FORMAT_S16; > > + case SPA_AUDIO_FORMAT_U16_BE: > > + *frame_size = 2; > > + *endianness = 1; > > + return AUDIO_FORMAT_U16; > > + case SPA_AUDIO_FORMAT_U16_LE: > > + *frame_size = 2; > > + *endianness = 0; > > + return AUDIO_FORMAT_U16; > > + case SPA_AUDIO_FORMAT_S32_BE: > > + *frame_size = 4; > > + *endianness = 1; > > + return AUDIO_FORMAT_S32; > > + case SPA_AUDIO_FORMAT_S32_LE: > > + *frame_size = 4; > > + *endianness = 0; > > + return AUDIO_FORMAT_S32; > > + case SPA_AUDIO_FORMAT_U32_BE: > > + *frame_size = 4; > > + *endianness = 1; > > + return AUDIO_FORMAT_U32; > > + case SPA_AUDIO_FORMAT_U32_LE: > > + *frame_size = 4; > > + *endianness = 0; > > + return AUDIO_FORMAT_U32; > > + case SPA_AUDIO_FORMAT_F32_BE: > > + *frame_size = 4; > > + *endianness = 1; > > + return AUDIO_FORMAT_F32; > > + case SPA_AUDIO_FORMAT_F32_LE: > > + *frame_size = 4; > > + *endianness = 0; > > + return AUDIO_FORMAT_F32; > > + default: > > + *frame_size = 1; > > + dolog("Internal logic error: Bad spa_audio_format %d\n", fmt); > > + return AUDIO_FORMAT_U8; > > + } > > +} > > + > > +static int > > +create_stream(pwaudio *c, PWVoice *v, const char *name) > > +{ > > + int res; > > + uint32_t n_params; > > + const struct spa_pod *params[2]; > > + uint8_t buffer[1024]; > > + struct spa_pod_builder b; > > + > > + v->stream = pw_stream_new(c->core, name, NULL); > > + > > + if (v->stream == NULL) { > > + res = -errno; > > + goto error; > > + } > > + > > + if (v->mode == MODE_SOURCE) { > > + pw_stream_add_listener(v->stream, > > + &v->stream_listener, > &capture_stream_events, v); > > + } else { > > + pw_stream_add_listener(v->stream, > > + &v->stream_listener, > &playback_stream_events, v); > > + } > > + > > + n_params = 0; > > + spa_pod_builder_init(&b, buffer, sizeof(buffer)); > > + params[n_params++] = spa_format_audio_raw_build(&b, > > + SPA_PARAM_EnumFormat, > > + &v->info); > > + > > + /* connect the stream to a sink or source */ > > + res = pw_stream_connect(v->stream, > > + v->mode == > > + MODE_SOURCE ? PW_DIRECTION_INPUT : > > + PW_DIRECTION_OUTPUT, PW_ID_ANY, > > + PW_STREAM_FLAG_AUTOCONNECT | > > + PW_STREAM_FLAG_MAP_BUFFERS | > > + PW_STREAM_FLAG_RT_PROCESS, params, > n_params); > > + if (res < 0) { > > + goto error; > > + } > > + > > + return 0; > > +error: > > + return res; > > +} > > + > > +static void > > +pw_destroy(pwaudio *c) > > +{ > > + if (c->thread_loop) { > > + pw_thread_loop_stop(c->thread_loop); > > + } > > + if (c->core) { > > + pw_core_disconnect(c->core); > > + } > > + > > + free(c); > > g_free(c); > > > +} > > + > > +static int > > +qpw_stream_new(pwaudio *c, PWVoice *v, const char *name) > > +{ > > + int r; > > + > > + pw_thread_loop_lock(c->thread_loop); > > + > > + switch (v->info.channels) { > > + case 8: > > + v->info.position[0] = SPA_AUDIO_CHANNEL_FL; > > + v->info.position[1] = SPA_AUDIO_CHANNEL_FR; > > + v->info.position[2] = SPA_AUDIO_CHANNEL_FC; > > + v->info.position[3] = SPA_AUDIO_CHANNEL_LFE; > > + v->info.position[4] = SPA_AUDIO_CHANNEL_RL; > > + v->info.position[5] = SPA_AUDIO_CHANNEL_RR; > > + v->info.position[6] = SPA_AUDIO_CHANNEL_SL; > > + v->info.position[7] = SPA_AUDIO_CHANNEL_SR; > > + break; > > + case 6: > > + v->info.position[0] = SPA_AUDIO_CHANNEL_FL; > > + v->info.position[1] = SPA_AUDIO_CHANNEL_FR; > > + v->info.position[2] = SPA_AUDIO_CHANNEL_FC; > > + v->info.position[3] = SPA_AUDIO_CHANNEL_LFE; > > + v->info.position[4] = SPA_AUDIO_CHANNEL_RL; > > + v->info.position[5] = SPA_AUDIO_CHANNEL_RR; > > + break; > > + case 5: > > + v->info.position[0] = SPA_AUDIO_CHANNEL_FL; > > + v->info.position[1] = SPA_AUDIO_CHANNEL_FR; > > + v->info.position[2] = SPA_AUDIO_CHANNEL_FC; > > + v->info.position[3] = SPA_AUDIO_CHANNEL_LFE; > > + v->info.position[4] = SPA_AUDIO_CHANNEL_RC; > > + break; > > + case 4: > > + v->info.position[0] = SPA_AUDIO_CHANNEL_FL; > > + v->info.position[1] = SPA_AUDIO_CHANNEL_FR; > > + v->info.position[2] = SPA_AUDIO_CHANNEL_FC; > > + v->info.position[3] = SPA_AUDIO_CHANNEL_RC; > > + break; > > + case 3: > > + v->info.position[0] = SPA_AUDIO_CHANNEL_FL; > > + v->info.position[1] = SPA_AUDIO_CHANNEL_FR; > > + v->info.position[2] = SPA_AUDIO_CHANNEL_LFE; > > + break; > > + case 2: > > + v->info.position[0] = SPA_AUDIO_CHANNEL_FL; > > + v->info.position[1] = SPA_AUDIO_CHANNEL_FR; > > + break; > > + case 1: > > + v->info.position[0] = SPA_AUDIO_CHANNEL_MONO; > > + break; > > + default: > > + for (size_t i = 0; i < v->info.channels; i++) { > > + v->info.position[i] = SPA_AUDIO_CHANNEL_UNKNOWN; > > + } > > + break; > > + } > > + > > + /* create a new unconnected pwstream */ > > + r = create_stream(c, v, name); > > + if (r < 0) { > > + goto error; > > + } > > + > > + pw_thread_loop_unlock(c->thread_loop); > > + return r; > > + > > +error: > > + AUD_log(AUDIO_CAP, "Failed to create stream."); > > + pw_thread_loop_unlock(c->thread_loop); > > + pw_destroy(c); > > + return -1; > > +} > > + > > +static int > > +qpw_init_out(HWVoiceOut *hw, struct audsettings *as, void *drv_opaque) > > +{ > > + PWVoiceOut *pw = (PWVoiceOut *) hw; > > + PWVoice *v = &pw->v; > > + struct audsettings obt_as = *as; > > + pwaudio *c = v->g = drv_opaque; > > + AudiodevPipewireOptions *popts = &c->dev->u.pipewire; > > + AudiodevPipewirePerDirectionOptions *ppdo = popts->out; > > + int r; > > + v->enabled = false; > > + > > + v->mode = MODE_SINK; > > + > > + pw_thread_loop_lock(c->thread_loop); > > + > > + v->info.format = audfmt_to_pw(as->fmt, as->endianness); > > + v->info.channels = as->nchannels; > > + v->info.rate = as->freq; > > + > > + obt_as.fmt = > > + pw_to_audfmt(v->info.format, &obt_as.endianness, > &v->frame_size); > > + v->frame_size *= as->nchannels; > > + > > + /* call the function that creates a new stream for playback */ > > + r = qpw_stream_new(c, v, ppdo->stream_name ? : c->dev->id); > > + if (r < 0) { > > + pw_log_error("qpw_stream_new for playback failed\n "); > > + goto fail; > > + } > > + > > + /* report the audio format we support */ > > + audio_pcm_init_info(&hw->info, &obt_as); > > + > > + /* report the buffer size to qemu */ > > + hw->samples = 512; > > + > > + pw_thread_loop_unlock(c->thread_loop); > > + return 0; > > +fail: > > + pw_thread_loop_unlock(c->thread_loop); > > + return -1; > > +} > > + > > +static int > > +qpw_init_in(HWVoiceIn *hw, struct audsettings *as, void *drv_opaque) > > +{ > > + PWVoiceIn *pw = (PWVoiceIn *) hw; > > + PWVoice *v = &pw->v; > > + struct audsettings obt_as = *as; > > + pwaudio *c = v->g = drv_opaque; > > + AudiodevPipewireOptions *popts = &c->dev->u.pipewire; > > + AudiodevPipewirePerDirectionOptions *ppdo = popts->in; > > + int r; > > + v->enabled = false; > > + > > + v->mode = MODE_SOURCE; > > + pw_thread_loop_lock(c->thread_loop); > > + > > + v->info.format = audfmt_to_pw(as->fmt, as->endianness); > > + v->info.channels = as->nchannels; > > + v->info.rate = as->freq; > > + > > + obt_as.fmt = > > + pw_to_audfmt(v->info.format, &obt_as.endianness, > &v->frame_size); > > + v->frame_size *= as->nchannels; > > + > > + /* call the function that creates a new stream for recording */ > > + r = qpw_stream_new(c, v, ppdo->stream_name ? : c->dev->id); > > + if (r < 0) { > > + pw_log_error("qpw_stream_new for recording failed\n "); > > + goto fail; > > + } > > + > > + /* report the audio format we support */ > > + audio_pcm_init_info(&hw->info, &obt_as); > > + > > + /* report the buffer size to qemu */ > > + hw->samples = 512; > > + > > + pw_thread_loop_unlock(c->thread_loop); > > + return 0; > > +fail: > > + pw_thread_loop_unlock(c->thread_loop); > > + return -1; > > +} > > + > > +static void > > +qpw_fini_out(HWVoiceOut *hw) > > +{ > > + PWVoiceOut *pw = (PWVoiceOut *) hw; > > + PWVoice *v = &pw->v; > > + > > + if (v->stream) { > > + pwaudio *c = v->g; > > + pw_thread_loop_lock(c->thread_loop); > > + pw_stream_destroy(v->stream); > > + v->stream = NULL; > > + pw_thread_loop_unlock(c->thread_loop); > > + } > > +} > > + > > +static void > > +qpw_fini_in(HWVoiceIn *hw) > > +{ > > + PWVoiceIn *pw = (PWVoiceIn *) hw; > > + PWVoice *v = &pw->v; > > + > > + if (v->stream) { > > + pwaudio *c = v->g; > > + pw_thread_loop_lock(c->thread_loop); > > + pw_stream_destroy(v->stream); > > + v->stream = NULL; > > + pw_thread_loop_unlock(c->thread_loop); > > + } > > +} > > + > > +static void > > +qpw_enable_out(HWVoiceOut *hw, bool enable) > > +{ > > + PWVoiceOut *po = (PWVoiceOut *) hw; > > + PWVoice *v = &po->v; > > + v->enabled = enable; > > +} > > + > > +static void > > +qpw_enable_in(HWVoiceIn *hw, bool enable) > > +{ > > + PWVoiceIn *pi = (PWVoiceIn *) hw; > > + PWVoice *v = &pi->v; > > + v->enabled = enable; > > +} > > + > > +static void > > +on_core_error(void *data, uint32_t id, int seq, int res, const char > *message) > > +{ > > + pwaudio *pw = data; > > + > > + pw_log_warn("error id:%u seq:%d res:%d (%s): %s", > > + id, seq, res, spa_strerror(res), message); > > + > > + pw_thread_loop_signal(pw->thread_loop, FALSE); > > +} > > + > > +static void > > +on_core_done(void *data, uint32_t id, int seq) > > +{ > > + pwaudio *pw = data; > > + if (id == PW_ID_CORE) { > > + pw->seq = seq; > > + pw_thread_loop_signal(pw->thread_loop, FALSE); > > + } > > +} > > + > > +static const struct pw_core_events core_events = { > > + PW_VERSION_CORE_EVENTS, > > + .done = on_core_done, > > + .error = on_core_error, > > +}; > > + > > +static void * > > +qpw_audio_init(Audiodev *dev) > > +{ > > + pwaudio *pw; > > + pw = g_new0(pwaudio, 1); > > + pw_init(NULL, NULL); > > + > > + AudiodevPipewireOptions *popts; > > + AUD_log(AUDIO_CAP, "Initialize PW context\n"); > > + assert(dev->driver == AUDIODEV_DRIVER_PIPEWIRE); > > + popts = &dev->u.pipewire; > > + > > + if (!popts->has_latency) { > > + popts->has_latency = true; > > + popts->latency = 44100; > > + } > > Why 44ms? > > > + > > + pw->dev = dev; > > + pw->thread_loop = pw_thread_loop_new("Pipewire thread loop", NULL); > > + if (pw->thread_loop == NULL) { > > + goto fail; > > + } > > + pw->context = > > + pw_context_new(pw_thread_loop_get_loop(pw->thread_loop), NULL, > 0); > > + > > + if (pw_thread_loop_start(pw->thread_loop) < 0) { > > + goto fail; > > + } > > + > > + pw_thread_loop_lock(pw->thread_loop); > > + > > + pw->core = pw_context_connect(pw->context, NULL, 0); > > + if (pw->core == NULL) { > > + goto fail; > > + } > > + > > + pw_core_add_listener(pw->core, &pw->core_listener, &core_events, > pw); > > + > > + pw_thread_loop_unlock(pw->thread_loop); > > + > > + return pw; > > + > > +fail: > > + AUD_log(AUDIO_CAP, "Failed to initialize PW context"); > > + pw_thread_loop_unlock(pw->thread_loop); > > + pw_context_destroy(pw->context); > > + pw_thread_loop_destroy(pw->thread_loop); > > + g_free(pw); > > + return NULL; > > +} > > + > > +static void > > +qpw_audio_fini(void *opaque) > > +{ > > + pwaudio *pw = opaque; > > + > > + pw_thread_loop_stop(pw->thread_loop); > > + > > + if (pw->core) { > > + spa_hook_remove(&pw->core_listener); > > + spa_zero(pw->core_listener); > > + pw_core_disconnect(pw->core); > > + } > > + > > + if (pw->context) { > > + pw_context_destroy(pw->context); > > + } > > + pw_thread_loop_destroy(pw->thread_loop); > > + > > + g_free(pw); > > +} > > + > > +static struct audio_pcm_ops qpw_pcm_ops = { > > + .init_out = qpw_init_out, > > + .fini_out = qpw_fini_out, > > + .write = qpw_write, > > + .buffer_get_free = audio_generic_buffer_get_free, > > + .run_buffer_out = audio_generic_run_buffer_out, > > + .enable_out = qpw_enable_out, > > + > > + .init_in = qpw_init_in, > > + .fini_in = qpw_fini_in, > > + .read = qpw_read, > > + .run_buffer_in = audio_generic_run_buffer_in, > > + .enable_in = qpw_enable_in > > +}; > > + > > +static struct audio_driver pw_audio_driver = { > > + .name = "pipewire", > > + .descr = "http://www.pipewire.org/", > > + .init = qpw_audio_init, > > + .fini = qpw_audio_fini, > > + .pcm_ops = &qpw_pcm_ops, > > + .can_be_default = 1, > > + .max_voices_out = INT_MAX, > > + .max_voices_in = INT_MAX, > > + .voice_size_out = sizeof(PWVoiceOut), > > + .voice_size_in = sizeof(PWVoiceIn), > > +}; > > + > > +static void > > +register_audio_pw(void) > > +{ > > + audio_driver_register(&pw_audio_driver); > > +} > > + > > +type_init(register_audio_pw); > > diff --git a/meson.build b/meson.build > > index a76c855312..686fdd5b81 100644 > > --- a/meson.build > > +++ b/meson.build > > @@ -734,6 +734,11 @@ 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', required: > get_option('pipewire'), > > + method: 'pkg-config', kwargs: static_kwargs) > > +endif > > sndio = not_found > > if not get_option('sndio').auto() or have_system > > sndio = dependency('sndio', required: get_option('sndio'), > > @@ -1671,6 +1676,7 @@ if have_system > > 'jack': jack.found(), > > 'oss': oss.found(), > > 'pa': pulse.found(), > > + 'pipewire': pipewire.found(), > > 'sdl': sdl.found(), > > 'sndio': sndio.found(), > > } > > @@ -3949,6 +3955,7 @@ endif > > if targetos == 'linux' > > summary_info += {'ALSA support': alsa} > > summary_info += {'PulseAudio support': pulse} > > + summary_info += {'Pipewire support': pipewire} > > endif > > summary_info += {'JACK support': jack} > > summary_info += {'brlapi support': brlapi} > > diff --git a/meson_options.txt b/meson_options.txt > > index 7e5801db90..1b7847250d 100644 > > --- a/meson_options.txt > > +++ b/meson_options.txt > > @@ -21,7 +21,7 @@ option('tls_priority', type : 'string', value : > 'NORMAL', > > option('default_devices', type : 'boolean', value : true, > > description: 'Include a default selection of devices in > emulators') > > option('audio_drv_list', type: 'array', value: ['default'], > > - choices: ['alsa', 'coreaudio', 'default', 'dsound', 'jack', > 'oss', 'pa', 'sdl', 'sndio'], > > + choices: ['alsa', 'coreaudio', 'default', 'dsound', 'jack', > 'oss', 'pa', 'pipewire', 'sdl', 'sndio'], > > description: 'Set audio driver list') > > option('block_drv_rw_whitelist', type : 'string', value : '', > > description: 'set block driver read-write whitelist (by default > affects only QEMU, not tools like qemu-img)') > > @@ -255,6 +255,8 @@ option('oss', type: 'feature', value: 'auto', > > description: 'OSS sound support') > > option('pa', type: 'feature', value: 'auto', > > description: 'PulseAudio sound support') > > +option('pipewire', type: 'feature', value: 'auto', > > + description: 'Pipewire sound support') > > option('sndio', type: 'feature', value: 'auto', > > description: 'sndio sound support') > > > > diff --git a/qapi/audio.json b/qapi/audio.json > > index 4e54c00f51..b872e9f10d 100644 > > --- a/qapi/audio.json > > +++ b/qapi/audio.json > > @@ -324,6 +324,48 @@ > > '*out': 'AudiodevPaPerDirectionOptions', > > '*server': 'str' } } > > > > +## > > +# @AudiodevPipewirePerDirectionOptions: > > +# > > +# Options of the Pipewire backend that are used for both playback and > > +# recording. > > +# > > +# @name: name of the sink/source to use > > +# > > +# @stream-name: name of the Pipewire stream created by qemu. Can be > > +# used to identify the stream in Pipewire when you > > +# create multiple Pipewire devices or run multiple qemu > > +# instances (default: audiodev's id, since 7.1) > > +# > > +# > > +# Since: 7.2 > > +## > > +{ 'struct': 'AudiodevPipewirePerDirectionOptions', > > + 'base': 'AudiodevPerDirectionOptions', > > + 'data': { > > + '*name': 'str', > > + '*stream-name': 'str' } } > > + > > +## > > +# @AudiodevPipewireOptions: > > +# > > +# Options of the Pipewire audio backend. > > +# > > +# @in: options of the capture stream > > +# > > +# @out: options of the playback stream > > +# > > +# @latency: add latency to playback in microseconds > > +# (default 44100) > > +# > > +# Since: 7.2 > > +## > > +{ 'struct': 'AudiodevPipewireOptions', > > + 'data': { > > + '*in': 'AudiodevPipewirePerDirectionOptions', > > + '*out': 'AudiodevPipewirePerDirectionOptions', > > + '*latency': 'uint32' } } > > + > > ## > > # @AudiodevSdlPerDirectionOptions: > > # > > @@ -416,6 +458,7 @@ > > { 'name': 'jack', 'if': 'CONFIG_AUDIO_JACK' }, > > { 'name': 'oss', 'if': 'CONFIG_AUDIO_OSS' }, > > { 'name': 'pa', 'if': 'CONFIG_AUDIO_PA' }, > > + { 'name': 'pipewire', 'if': 'CONFIG_AUDIO_PIPEWIRE' }, > > { 'name': 'sdl', 'if': 'CONFIG_AUDIO_SDL' }, > > { 'name': 'sndio', 'if': 'CONFIG_AUDIO_SNDIO' }, > > { 'name': 'spice', 'if': 'CONFIG_SPICE' }, > > @@ -456,6 +499,8 @@ > > 'if': 'CONFIG_AUDIO_OSS' }, > > 'pa': { 'type': 'AudiodevPaOptions', > > 'if': 'CONFIG_AUDIO_PA' }, > > + 'pipewire': { 'type': 'AudiodevPipewireOptions', > > + 'if': 'CONFIG_AUDIO_PIPEWIRE' }, > > 'sdl': { 'type': 'AudiodevSdlOptions', > > 'if': 'CONFIG_AUDIO_SDL' }, > > 'sndio': { 'type': 'AudiodevSndioOptions', > > diff --git a/qemu-options.hx b/qemu-options.hx > > index 88e93c6103..bde4830fab 100644 > > --- a/qemu-options.hx > > +++ b/qemu-options.hx > > @@ -779,6 +779,11 @@ DEF("audiodev", HAS_ARG, QEMU_OPTION_audiodev, > > " in|out.name= source/sink device name\n" > > " in|out.latency= desired latency in microseconds\n" > > #endif > > +#ifdef CONFIG_AUDIO_PIPEWIRE > > + "-audiodev pipewire,id=id[,prop[=value][,...]]\n" > > + " in|out.name= source/sink device name\n" > > + " latency= desired latency in microseconds\n" > > +#endif > > #ifdef CONFIG_AUDIO_SDL > > "-audiodev sdl,id=id[,prop[=value][,...]]\n" > > " in|out.buffer-count= number of buffers\n" > > @@ -942,6 +947,18 @@ SRST > > Desired latency in microseconds. The PulseAudio server will try > > to honor this value but actual latencies may be lower or higher. > > > > +``-audiodev pipewire,id=id[,prop[=value][,...]]`` > > + Creates a backend using Pipewire. This backend is available on > > + most systems. > > + > > + Pipewire specific options are: > > + > > + ``latency=latency`` > > + Add extra latency to playback in microseconds > > + > > + ``in|out.name=sink`` > > + Use the specified source/sink for recording/playback. > > + > > ``-audiodev sdl,id=id[,prop[=value][,...]]`` > > Creates a backend using SDL. This backend is available on most > > systems, but you should use your platform's native backend if > > diff --git a/scripts/meson-buildoptions.sh > b/scripts/meson-buildoptions.sh > > index 180c11665a..d9f6525346 100644 > > --- a/scripts/meson-buildoptions.sh > > +++ b/scripts/meson-buildoptions.sh > > @@ -1,7 +1,8 @@ > > # This file is generated by meson-buildoptions.py, do not edit! > > meson_options_help() { > > - printf "%s\n" ' --audio-drv-list=CHOICES Set audio driver list > [default] (choices: alsa/co' > > - printf "%s\n" ' > reaudio/default/dsound/jack/oss/pa/sdl/sndio)' > > + printf "%s\n" ' --audio-drv-list=CHOICES Set audio driver list > [default] (choices: al' > > + printf "%s\n" ' > sa/coreaudio/default/dsound/jack/oss/pa/' > > + printf "%s\n" ' pipewire/sdl/sndio)' > > printf "%s\n" ' --block-drv-ro-whitelist=VALUE' > > printf "%s\n" ' set block driver read-only > whitelist (by default' > > printf "%s\n" ' affects only QEMU, not > tools like qemu-img)' > > @@ -135,6 +136,7 @@ meson_options_help() { > > printf "%s\n" ' oss OSS sound support' > > printf "%s\n" ' pa PulseAudio sound support' > > printf "%s\n" ' parallels parallels image format support' > > + printf "%s\n" ' pipewire Pipewire sound support' > > printf "%s\n" ' png PNG support with libpng' > > printf "%s\n" ' pvrdma Enable PVRDMA support' > > printf "%s\n" ' qcow1 qcow1 image format support' > > @@ -370,6 +372,8 @@ _meson_option_parse() { > > --disable-pa) printf "%s" -Dpa=disabled ;; > > --enable-parallels) printf "%s" -Dparallels=enabled ;; > > --disable-parallels) printf "%s" -Dparallels=disabled ;; > > + --enable-pipewire) printf "%s" -Dpipewire=enabled ;; > > + --disable-pipewire) printf "%s" -Dpipewire=disabled ;; > > --with-pkgversion=*) quote_sh "-Dpkgversion=$2" ;; > > --enable-png) printf "%s" -Dpng=enabled ;; > > --disable-png) printf "%s" -Dpng=disabled ;; > > > > >
diff --git a/audio/audio.c b/audio/audio.c index 4290309d18..aa55e41ad8 100644 --- a/audio/audio.c +++ b/audio/audio.c @@ -2069,6 +2069,9 @@ void audio_create_pdos(Audiodev *dev) #ifdef CONFIG_AUDIO_PA CASE(PA, pa, Pa); #endif +#ifdef CONFIG_AUDIO_PIPEWIRE + CASE(PIPEWIRE, pipewire, Pipewire); +#endif #ifdef CONFIG_AUDIO_SDL CASE(SDL, sdl, Sdl); #endif diff --git a/audio/audio_template.h b/audio/audio_template.h index 42b4712acb..0f02afb921 100644 --- a/audio/audio_template.h +++ b/audio/audio_template.h @@ -355,6 +355,10 @@ AudiodevPerDirectionOptions *glue(audio_get_pdo_, TYPE)(Audiodev *dev) case AUDIODEV_DRIVER_PA: return qapi_AudiodevPaPerDirectionOptions_base(dev->u.pa.TYPE); #endif +#ifdef CONFIG_AUDIO_PIPEWIRE + case AUDIODEV_DRIVER_PIPEWIRE: + return qapi_AudiodevPipewirePerDirectionOptions_base(dev->u.pipewire.TYPE); +#endif #ifdef CONFIG_AUDIO_SDL case AUDIODEV_DRIVER_SDL: return qapi_AudiodevSdlPerDirectionOptions_base(dev->u.sdl.TYPE); diff --git a/audio/meson.build b/audio/meson.build index 0722224ba9..65a49c1a10 100644 --- a/audio/meson.build +++ b/audio/meson.build @@ -19,6 +19,7 @@ foreach m : [ ['sdl', sdl, files('sdlaudio.c')], ['jack', jack, files('jackaudio.c')], ['sndio', sndio, files('sndioaudio.c')], + ['pipewire', pipewire, files('pwaudio.c')], ['spice', spice, files('spiceaudio.c')] ] if m[1].found() diff --git a/audio/pwaudio.c b/audio/pwaudio.c new file mode 100644 index 0000000000..bb25133414 --- /dev/null +++ b/audio/pwaudio.c @@ -0,0 +1,818 @@ +/* + * QEMU Pipewire audio driver + * + * Copyright (c) 2023 Red Hat Inc. + * + * Author: Dorinda Bassey <dbassey@redhat.com> + * + * Permission is hereby granted, free of charge, to any person obtaining a copy + * of this software and associated documentation files (the "Software"), to deal + * in the Software without restriction, including without limitation the rights + * to use, copy, modify, merge, publish, distribute, sublicense, and/or sell + * copies of the Software, and to permit persons to whom the Software is + * furnished to do so, subject to the following conditions: + * + * The above copyright notice and this permission notice shall be included in + * all copies or substantial portions of the Software. + * + * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR + * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY, + * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL + * THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER + * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM, + * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN + * THE SOFTWARE. + */ + +#include "qemu/osdep.h" +#include "qemu/module.h" +#include "audio.h" +#include <errno.h> +#include <spa/param/audio/format-utils.h> +#include <spa/utils/ringbuffer.h> +#include <spa/utils/result.h> + +#include <pipewire/pipewire.h> + +#define AUDIO_CAP "pipewire" +#define RINGBUFFER_SIZE (1u << 22) +#define RINGBUFFER_MASK (RINGBUFFER_SIZE - 1) +#define BUFFER_SAMPLES 128 + +#include "audio_int.h" + +enum { + MODE_SINK, + MODE_SOURCE +}; + +typedef struct pwaudio { + Audiodev *dev; + struct pw_thread_loop *thread_loop; + struct pw_context *context; + + struct pw_core *core; + struct spa_hook core_listener; + int seq; +} pwaudio; + +typedef struct PWVoice { + pwaudio *g; + bool enabled; + struct pw_stream *stream; + struct spa_hook stream_listener; + struct spa_audio_info_raw info; + uint32_t frame_size; + struct spa_ringbuffer ring; + uint8_t buffer[RINGBUFFER_SIZE]; + + uint32_t mode; + struct pw_properties *props; +} PWVoice; + +typedef struct PWVoiceOut { + HWVoiceOut hw; + PWVoice v; +} PWVoiceOut; + +typedef struct PWVoiceIn { + HWVoiceIn hw; + PWVoice v; +} PWVoiceIn; + +static void +stream_destroy(void *data) +{ + PWVoice *v = (PWVoice *) data; + spa_hook_remove(&v->stream_listener); + v->stream = NULL; +} + +/* output data processing function to read stuffs from the buffer */ +static void +playback_on_process(void *data) +{ + PWVoice *v = (PWVoice *) data; + void *p; + struct pw_buffer *b; + struct spa_buffer *buf; + uint32_t n_frames, req, index, n_bytes; + int32_t avail; + + /* obtain a buffer to read from */ + b = pw_stream_dequeue_buffer(v->stream); + if (b == NULL) { + pw_log_warn("out of buffers: %m"); + return; + } + + buf = b->buffer; + p = buf->datas[0].data; + if (p == NULL) { + return; + } + req = b->requested * v->frame_size; + if (req == 0) { + req = 4096 * v->frame_size; + } + n_frames = SPA_MIN(req, buf->datas[0].maxsize); + n_bytes = n_frames * v->frame_size; + + /* get no of available bytes to read data from buffer */ + + avail = spa_ringbuffer_get_read_index(&v->ring, &index); + + if (!v->enabled) { + avail = 0; + } + + if (avail == 0) { + memset(p, 0, n_bytes); + } else { + if (avail < (int32_t) n_bytes) { + n_bytes = avail; + } + + spa_ringbuffer_read_data(&v->ring, + v->buffer, RINGBUFFER_SIZE, + index & RINGBUFFER_MASK, p, n_bytes); + + index += n_bytes; + spa_ringbuffer_read_update(&v->ring, index); + } + + buf->datas[0].chunk->offset = 0; + buf->datas[0].chunk->stride = v->frame_size; + buf->datas[0].chunk->size = n_bytes; + + /* queue the buffer for playback */ + pw_stream_queue_buffer(v->stream, b); +} + +/* output data processing function to generate stuffs in the buffer */ +static void +capture_on_process(void *data) +{ + PWVoice *v = (PWVoice *) data; + void *p; + struct pw_buffer *b; + struct spa_buffer *buf; + int32_t filled; + uint32_t index, offs, n_bytes; + + /* obtain a buffer */ + b = pw_stream_dequeue_buffer(v->stream); + if (b == NULL) { + pw_log_warn("out of buffers: %m"); + return; + } + + /* Write data into buffer */ + buf = b->buffer; + p = buf->datas[0].data; + if (p == NULL) { + return; + } + offs = SPA_MIN(buf->datas[0].chunk->offset, buf->datas[0].maxsize); + n_bytes = SPA_MIN(buf->datas[0].chunk->size, buf->datas[0].maxsize - offs); + + filled = spa_ringbuffer_get_write_index(&v->ring, &index); + + if (!v->enabled) { + n_bytes = 0; + } + + if (filled < 0) { + pw_log_warn("%p: underrun write:%u filled:%d", p, index, filled); + } else { + if ((uint32_t) filled + n_bytes > RINGBUFFER_SIZE) { + pw_log_warn("%p: overrun write:%u filled:%d + size:%u > max:%u", + p, index, filled, n_bytes, RINGBUFFER_SIZE); + } + } + spa_ringbuffer_write_data(&v->ring, + v->buffer, RINGBUFFER_SIZE, + index & RINGBUFFER_MASK, + SPA_PTROFF(p, offs, void), n_bytes); + index += n_bytes; + spa_ringbuffer_write_update(&v->ring, index); + + /* queue the buffer for playback */ + pw_stream_queue_buffer(v->stream, b); +} + +static void +on_stream_state_changed(void *_data, enum pw_stream_state old, + enum pw_stream_state state, const char *error) +{ + PWVoice *v = (PWVoice *) _data; + + printf("stream state: \"%s\"\n", pw_stream_state_as_string(state)); + + switch (state) { + case PW_STREAM_STATE_ERROR: + case PW_STREAM_STATE_UNCONNECTED: + { + break; + } + case PW_STREAM_STATE_PAUSED: + printf("node id: %d\n", pw_stream_get_node_id(v->stream)); + break; + case PW_STREAM_STATE_CONNECTING: + case PW_STREAM_STATE_STREAMING: + break; + } +} + +static const struct pw_stream_events capture_stream_events = { + PW_VERSION_STREAM_EVENTS, + .destroy = stream_destroy, + .state_changed = on_stream_state_changed, + .process = capture_on_process +}; + +static const struct pw_stream_events playback_stream_events = { + PW_VERSION_STREAM_EVENTS, + .destroy = stream_destroy, + .state_changed = on_stream_state_changed, + .process = playback_on_process +}; + +static size_t +qpw_read(HWVoiceIn *hw, void *data, size_t len) +{ + PWVoiceIn *pw = (PWVoiceIn *) hw; + PWVoice *v = &pw->v; + pwaudio *c = v->g; + const char *error = NULL; + size_t l; + int32_t avail; + uint32_t index; + + pw_thread_loop_lock(c->thread_loop); + if (pw_stream_get_state(v->stream, &error) != PW_STREAM_STATE_STREAMING) { + /* wait for stream to become ready */ + l = 0; + goto done_unlock; + } + /* get no of available bytes to read data from buffer */ + avail = spa_ringbuffer_get_read_index(&v->ring, &index); + + if (avail < (int32_t) len) { + len = avail; + } + + spa_ringbuffer_read_data(&v->ring, + v->buffer, RINGBUFFER_SIZE, + index & RINGBUFFER_MASK, data, len); + index += len; + spa_ringbuffer_read_update(&v->ring, index); + l = len; + +done_unlock: + pw_thread_loop_unlock(c->thread_loop); + return l; +} + +static size_t +qpw_write(HWVoiceOut *hw, void *data, size_t len) +{ + PWVoiceOut *pw = (PWVoiceOut *) hw; + PWVoice *v = &pw->v; + pwaudio *c = v->g; + const char *error = NULL; + size_t l; + int32_t filled, avail; + uint32_t index; + + pw_thread_loop_lock(c->thread_loop); + if (pw_stream_get_state(v->stream, &error) != PW_STREAM_STATE_STREAMING) { + /* wait for stream to become ready */ + l = 0; + goto done_unlock; + } + filled = spa_ringbuffer_get_write_index(&v->ring, &index); + + avail = 512 * v->frame_size * 3 - filled; + + pw_log_debug("%u %u %u %zu", filled, avail, index, len); + + if (len > avail) { + len = avail; + } + + if (filled < 0) { + pw_log_warn("%p: underrun write:%u filled:%d", pw, index, filled); + } else { + if ((uint32_t) filled + len > RINGBUFFER_SIZE) { + pw_log_warn("%p: overrun write:%u filled:%d + size:%zu > max:%u", + pw, index, filled, len, RINGBUFFER_SIZE); + } + } + + spa_ringbuffer_write_data(&v->ring, + v->buffer, RINGBUFFER_SIZE, + index & RINGBUFFER_MASK, data, len); + index += len; + spa_ringbuffer_write_update(&v->ring, index); + l = len; + +done_unlock: + pw_thread_loop_unlock(c->thread_loop); + return l; +} + +static int +audfmt_to_pw(AudioFormat fmt, int endianness) +{ + int format; + + switch (fmt) { + case AUDIO_FORMAT_S8: + format = SPA_AUDIO_FORMAT_S8; + break; + case AUDIO_FORMAT_U8: + format = SPA_AUDIO_FORMAT_U8; + break; + case AUDIO_FORMAT_S16: + format = endianness ? SPA_AUDIO_FORMAT_S16_BE : SPA_AUDIO_FORMAT_S16_LE; + break; + case AUDIO_FORMAT_U16: + format = endianness ? SPA_AUDIO_FORMAT_U16_BE : SPA_AUDIO_FORMAT_U16_LE; + break; + case AUDIO_FORMAT_S32: + format = endianness ? SPA_AUDIO_FORMAT_S32_BE : SPA_AUDIO_FORMAT_S32_LE; + break; + case AUDIO_FORMAT_U32: + format = endianness ? SPA_AUDIO_FORMAT_U32_BE : SPA_AUDIO_FORMAT_U32_LE; + break; + case AUDIO_FORMAT_F32: + format = endianness ? SPA_AUDIO_FORMAT_F32_BE : SPA_AUDIO_FORMAT_F32_LE; + break; + default: + dolog("Internal logic error: Bad audio format %d\n", fmt); + format = SPA_AUDIO_FORMAT_U8; + break; + } + return format; +} + +static AudioFormat +pw_to_audfmt(enum spa_audio_format fmt, int *endianness, + uint32_t *frame_size) +{ + switch (fmt) { + case SPA_AUDIO_FORMAT_S8: + *frame_size = 1; + return AUDIO_FORMAT_S8; + case SPA_AUDIO_FORMAT_U8: + *frame_size = 1; + return AUDIO_FORMAT_U8; + case SPA_AUDIO_FORMAT_S16_BE: + *frame_size = 2; + *endianness = 1; + return AUDIO_FORMAT_S16; + case SPA_AUDIO_FORMAT_S16_LE: + *frame_size = 2; + *endianness = 0; + return AUDIO_FORMAT_S16; + case SPA_AUDIO_FORMAT_U16_BE: + *frame_size = 2; + *endianness = 1; + return AUDIO_FORMAT_U16; + case SPA_AUDIO_FORMAT_U16_LE: + *frame_size = 2; + *endianness = 0; + return AUDIO_FORMAT_U16; + case SPA_AUDIO_FORMAT_S32_BE: + *frame_size = 4; + *endianness = 1; + return AUDIO_FORMAT_S32; + case SPA_AUDIO_FORMAT_S32_LE: + *frame_size = 4; + *endianness = 0; + return AUDIO_FORMAT_S32; + case SPA_AUDIO_FORMAT_U32_BE: + *frame_size = 4; + *endianness = 1; + return AUDIO_FORMAT_U32; + case SPA_AUDIO_FORMAT_U32_LE: + *frame_size = 4; + *endianness = 0; + return AUDIO_FORMAT_U32; + case SPA_AUDIO_FORMAT_F32_BE: + *frame_size = 4; + *endianness = 1; + return AUDIO_FORMAT_F32; + case SPA_AUDIO_FORMAT_F32_LE: + *frame_size = 4; + *endianness = 0; + return AUDIO_FORMAT_F32; + default: + *frame_size = 1; + dolog("Internal logic error: Bad spa_audio_format %d\n", fmt); + return AUDIO_FORMAT_U8; + } +} + +static int +create_stream(pwaudio *c, PWVoice *v, const char *name) +{ + int res; + uint32_t n_params; + const struct spa_pod *params[2]; + uint8_t buffer[1024]; + struct spa_pod_builder b; + + v->stream = pw_stream_new(c->core, name, NULL); + + if (v->stream == NULL) { + res = -errno; + goto error; + } + + if (v->mode == MODE_SOURCE) { + pw_stream_add_listener(v->stream, + &v->stream_listener, &capture_stream_events, v); + } else { + pw_stream_add_listener(v->stream, + &v->stream_listener, &playback_stream_events, v); + } + + n_params = 0; + spa_pod_builder_init(&b, buffer, sizeof(buffer)); + params[n_params++] = spa_format_audio_raw_build(&b, + SPA_PARAM_EnumFormat, + &v->info); + + /* connect the stream to a sink or source */ + res = pw_stream_connect(v->stream, + v->mode == + MODE_SOURCE ? PW_DIRECTION_INPUT : + PW_DIRECTION_OUTPUT, PW_ID_ANY, + PW_STREAM_FLAG_AUTOCONNECT | + PW_STREAM_FLAG_MAP_BUFFERS | + PW_STREAM_FLAG_RT_PROCESS, params, n_params); + if (res < 0) { + goto error; + } + + return 0; +error: + return res; +} + +static void +pw_destroy(pwaudio *c) +{ + if (c->thread_loop) { + pw_thread_loop_stop(c->thread_loop); + } + if (c->core) { + pw_core_disconnect(c->core); + } + + free(c); +} + +static int +qpw_stream_new(pwaudio *c, PWVoice *v, const char *name) +{ + int r; + + pw_thread_loop_lock(c->thread_loop); + + switch (v->info.channels) { + case 8: + v->info.position[0] = SPA_AUDIO_CHANNEL_FL; + v->info.position[1] = SPA_AUDIO_CHANNEL_FR; + v->info.position[2] = SPA_AUDIO_CHANNEL_FC; + v->info.position[3] = SPA_AUDIO_CHANNEL_LFE; + v->info.position[4] = SPA_AUDIO_CHANNEL_RL; + v->info.position[5] = SPA_AUDIO_CHANNEL_RR; + v->info.position[6] = SPA_AUDIO_CHANNEL_SL; + v->info.position[7] = SPA_AUDIO_CHANNEL_SR; + break; + case 6: + v->info.position[0] = SPA_AUDIO_CHANNEL_FL; + v->info.position[1] = SPA_AUDIO_CHANNEL_FR; + v->info.position[2] = SPA_AUDIO_CHANNEL_FC; + v->info.position[3] = SPA_AUDIO_CHANNEL_LFE; + v->info.position[4] = SPA_AUDIO_CHANNEL_RL; + v->info.position[5] = SPA_AUDIO_CHANNEL_RR; + break; + case 5: + v->info.position[0] = SPA_AUDIO_CHANNEL_FL; + v->info.position[1] = SPA_AUDIO_CHANNEL_FR; + v->info.position[2] = SPA_AUDIO_CHANNEL_FC; + v->info.position[3] = SPA_AUDIO_CHANNEL_LFE; + v->info.position[4] = SPA_AUDIO_CHANNEL_RC; + break; + case 4: + v->info.position[0] = SPA_AUDIO_CHANNEL_FL; + v->info.position[1] = SPA_AUDIO_CHANNEL_FR; + v->info.position[2] = SPA_AUDIO_CHANNEL_FC; + v->info.position[3] = SPA_AUDIO_CHANNEL_RC; + break; + case 3: + v->info.position[0] = SPA_AUDIO_CHANNEL_FL; + v->info.position[1] = SPA_AUDIO_CHANNEL_FR; + v->info.position[2] = SPA_AUDIO_CHANNEL_LFE; + break; + case 2: + v->info.position[0] = SPA_AUDIO_CHANNEL_FL; + v->info.position[1] = SPA_AUDIO_CHANNEL_FR; + break; + case 1: + v->info.position[0] = SPA_AUDIO_CHANNEL_MONO; + break; + default: + for (size_t i = 0; i < v->info.channels; i++) { + v->info.position[i] = SPA_AUDIO_CHANNEL_UNKNOWN; + } + break; + } + + /* create a new unconnected pwstream */ + r = create_stream(c, v, name); + if (r < 0) { + goto error; + } + + pw_thread_loop_unlock(c->thread_loop); + return r; + +error: + AUD_log(AUDIO_CAP, "Failed to create stream."); + pw_thread_loop_unlock(c->thread_loop); + pw_destroy(c); + return -1; +} + +static int +qpw_init_out(HWVoiceOut *hw, struct audsettings *as, void *drv_opaque) +{ + PWVoiceOut *pw = (PWVoiceOut *) hw; + PWVoice *v = &pw->v; + struct audsettings obt_as = *as; + pwaudio *c = v->g = drv_opaque; + AudiodevPipewireOptions *popts = &c->dev->u.pipewire; + AudiodevPipewirePerDirectionOptions *ppdo = popts->out; + int r; + v->enabled = false; + + v->mode = MODE_SINK; + + pw_thread_loop_lock(c->thread_loop); + + v->info.format = audfmt_to_pw(as->fmt, as->endianness); + v->info.channels = as->nchannels; + v->info.rate = as->freq; + + obt_as.fmt = + pw_to_audfmt(v->info.format, &obt_as.endianness, &v->frame_size); + v->frame_size *= as->nchannels; + + /* call the function that creates a new stream for playback */ + r = qpw_stream_new(c, v, ppdo->stream_name ? : c->dev->id); + if (r < 0) { + pw_log_error("qpw_stream_new for playback failed\n "); + goto fail; + } + + /* report the audio format we support */ + audio_pcm_init_info(&hw->info, &obt_as); + + /* report the buffer size to qemu */ + hw->samples = 512; + + pw_thread_loop_unlock(c->thread_loop); + return 0; +fail: + pw_thread_loop_unlock(c->thread_loop); + return -1; +} + +static int +qpw_init_in(HWVoiceIn *hw, struct audsettings *as, void *drv_opaque) +{ + PWVoiceIn *pw = (PWVoiceIn *) hw; + PWVoice *v = &pw->v; + struct audsettings obt_as = *as; + pwaudio *c = v->g = drv_opaque; + AudiodevPipewireOptions *popts = &c->dev->u.pipewire; + AudiodevPipewirePerDirectionOptions *ppdo = popts->in; + int r; + v->enabled = false; + + v->mode = MODE_SOURCE; + pw_thread_loop_lock(c->thread_loop); + + v->info.format = audfmt_to_pw(as->fmt, as->endianness); + v->info.channels = as->nchannels; + v->info.rate = as->freq; + + obt_as.fmt = + pw_to_audfmt(v->info.format, &obt_as.endianness, &v->frame_size); + v->frame_size *= as->nchannels; + + /* call the function that creates a new stream for recording */ + r = qpw_stream_new(c, v, ppdo->stream_name ? : c->dev->id); + if (r < 0) { + pw_log_error("qpw_stream_new for recording failed\n "); + goto fail; + } + + /* report the audio format we support */ + audio_pcm_init_info(&hw->info, &obt_as); + + /* report the buffer size to qemu */ + hw->samples = 512; + + pw_thread_loop_unlock(c->thread_loop); + return 0; +fail: + pw_thread_loop_unlock(c->thread_loop); + return -1; +} + +static void +qpw_fini_out(HWVoiceOut *hw) +{ + PWVoiceOut *pw = (PWVoiceOut *) hw; + PWVoice *v = &pw->v; + + if (v->stream) { + pwaudio *c = v->g; + pw_thread_loop_lock(c->thread_loop); + pw_stream_destroy(v->stream); + v->stream = NULL; + pw_thread_loop_unlock(c->thread_loop); + } +} + +static void +qpw_fini_in(HWVoiceIn *hw) +{ + PWVoiceIn *pw = (PWVoiceIn *) hw; + PWVoice *v = &pw->v; + + if (v->stream) { + pwaudio *c = v->g; + pw_thread_loop_lock(c->thread_loop); + pw_stream_destroy(v->stream); + v->stream = NULL; + pw_thread_loop_unlock(c->thread_loop); + } +} + +static void +qpw_enable_out(HWVoiceOut *hw, bool enable) +{ + PWVoiceOut *po = (PWVoiceOut *) hw; + PWVoice *v = &po->v; + v->enabled = enable; +} + +static void +qpw_enable_in(HWVoiceIn *hw, bool enable) +{ + PWVoiceIn *pi = (PWVoiceIn *) hw; + PWVoice *v = &pi->v; + v->enabled = enable; +} + +static void +on_core_error(void *data, uint32_t id, int seq, int res, const char *message) +{ + pwaudio *pw = data; + + pw_log_warn("error id:%u seq:%d res:%d (%s): %s", + id, seq, res, spa_strerror(res), message); + + pw_thread_loop_signal(pw->thread_loop, FALSE); +} + +static void +on_core_done(void *data, uint32_t id, int seq) +{ + pwaudio *pw = data; + if (id == PW_ID_CORE) { + pw->seq = seq; + pw_thread_loop_signal(pw->thread_loop, FALSE); + } +} + +static const struct pw_core_events core_events = { + PW_VERSION_CORE_EVENTS, + .done = on_core_done, + .error = on_core_error, +}; + +static void * +qpw_audio_init(Audiodev *dev) +{ + pwaudio *pw; + pw = g_new0(pwaudio, 1); + pw_init(NULL, NULL); + + AudiodevPipewireOptions *popts; + AUD_log(AUDIO_CAP, "Initialize PW context\n"); + assert(dev->driver == AUDIODEV_DRIVER_PIPEWIRE); + popts = &dev->u.pipewire; + + if (!popts->has_latency) { + popts->has_latency = true; + popts->latency = 44100; + } + + pw->dev = dev; + pw->thread_loop = pw_thread_loop_new("Pipewire thread loop", NULL); + if (pw->thread_loop == NULL) { + goto fail; + } + pw->context = + pw_context_new(pw_thread_loop_get_loop(pw->thread_loop), NULL, 0); + + if (pw_thread_loop_start(pw->thread_loop) < 0) { + goto fail; + } + + pw_thread_loop_lock(pw->thread_loop); + + pw->core = pw_context_connect(pw->context, NULL, 0); + if (pw->core == NULL) { + goto fail; + } + + pw_core_add_listener(pw->core, &pw->core_listener, &core_events, pw); + + pw_thread_loop_unlock(pw->thread_loop); + + return pw; + +fail: + AUD_log(AUDIO_CAP, "Failed to initialize PW context"); + pw_thread_loop_unlock(pw->thread_loop); + pw_context_destroy(pw->context); + pw_thread_loop_destroy(pw->thread_loop); + g_free(pw); + return NULL; +} + +static void +qpw_audio_fini(void *opaque) +{ + pwaudio *pw = opaque; + + pw_thread_loop_stop(pw->thread_loop); + + if (pw->core) { + spa_hook_remove(&pw->core_listener); + spa_zero(pw->core_listener); + pw_core_disconnect(pw->core); + } + + if (pw->context) { + pw_context_destroy(pw->context); + } + pw_thread_loop_destroy(pw->thread_loop); + + g_free(pw); +} + +static struct audio_pcm_ops qpw_pcm_ops = { + .init_out = qpw_init_out, + .fini_out = qpw_fini_out, + .write = qpw_write, + .buffer_get_free = audio_generic_buffer_get_free, + .run_buffer_out = audio_generic_run_buffer_out, + .enable_out = qpw_enable_out, + + .init_in = qpw_init_in, + .fini_in = qpw_fini_in, + .read = qpw_read, + .run_buffer_in = audio_generic_run_buffer_in, + .enable_in = qpw_enable_in +}; + +static struct audio_driver pw_audio_driver = { + .name = "pipewire", + .descr = "http://www.pipewire.org/", + .init = qpw_audio_init, + .fini = qpw_audio_fini, + .pcm_ops = &qpw_pcm_ops, + .can_be_default = 1, + .max_voices_out = INT_MAX, + .max_voices_in = INT_MAX, + .voice_size_out = sizeof(PWVoiceOut), + .voice_size_in = sizeof(PWVoiceIn), +}; + +static void +register_audio_pw(void) +{ + audio_driver_register(&pw_audio_driver); +} + +type_init(register_audio_pw); diff --git a/meson.build b/meson.build index a76c855312..686fdd5b81 100644 --- a/meson.build +++ b/meson.build @@ -734,6 +734,11 @@ 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', required: get_option('pipewire'), + method: 'pkg-config', kwargs: static_kwargs) +endif sndio = not_found if not get_option('sndio').auto() or have_system sndio = dependency('sndio', required: get_option('sndio'), @@ -1671,6 +1676,7 @@ if have_system 'jack': jack.found(), 'oss': oss.found(), 'pa': pulse.found(), + 'pipewire': pipewire.found(), 'sdl': sdl.found(), 'sndio': sndio.found(), } @@ -3949,6 +3955,7 @@ endif if targetos == 'linux' summary_info += {'ALSA support': alsa} summary_info += {'PulseAudio support': pulse} + summary_info += {'Pipewire support': pipewire} endif summary_info += {'JACK support': jack} summary_info += {'brlapi support': brlapi} diff --git a/meson_options.txt b/meson_options.txt index 7e5801db90..1b7847250d 100644 --- a/meson_options.txt +++ b/meson_options.txt @@ -21,7 +21,7 @@ option('tls_priority', type : 'string', value : 'NORMAL', option('default_devices', type : 'boolean', value : true, description: 'Include a default selection of devices in emulators') option('audio_drv_list', type: 'array', value: ['default'], - choices: ['alsa', 'coreaudio', 'default', 'dsound', 'jack', 'oss', 'pa', 'sdl', 'sndio'], + choices: ['alsa', 'coreaudio', 'default', 'dsound', 'jack', 'oss', 'pa', 'pipewire', 'sdl', 'sndio'], description: 'Set audio driver list') option('block_drv_rw_whitelist', type : 'string', value : '', description: 'set block driver read-write whitelist (by default affects only QEMU, not tools like qemu-img)') @@ -255,6 +255,8 @@ option('oss', type: 'feature', value: 'auto', description: 'OSS sound support') option('pa', type: 'feature', value: 'auto', description: 'PulseAudio sound support') +option('pipewire', type: 'feature', value: 'auto', + description: 'Pipewire sound support') option('sndio', type: 'feature', value: 'auto', description: 'sndio sound support') diff --git a/qapi/audio.json b/qapi/audio.json index 4e54c00f51..b872e9f10d 100644 --- a/qapi/audio.json +++ b/qapi/audio.json @@ -324,6 +324,48 @@ '*out': 'AudiodevPaPerDirectionOptions', '*server': 'str' } } +## +# @AudiodevPipewirePerDirectionOptions: +# +# Options of the Pipewire backend that are used for both playback and +# recording. +# +# @name: name of the sink/source to use +# +# @stream-name: name of the Pipewire stream created by qemu. Can be +# used to identify the stream in Pipewire when you +# create multiple Pipewire devices or run multiple qemu +# instances (default: audiodev's id, since 7.1) +# +# +# Since: 7.2 +## +{ 'struct': 'AudiodevPipewirePerDirectionOptions', + 'base': 'AudiodevPerDirectionOptions', + 'data': { + '*name': 'str', + '*stream-name': 'str' } } + +## +# @AudiodevPipewireOptions: +# +# Options of the Pipewire audio backend. +# +# @in: options of the capture stream +# +# @out: options of the playback stream +# +# @latency: add latency to playback in microseconds +# (default 44100) +# +# Since: 7.2 +## +{ 'struct': 'AudiodevPipewireOptions', + 'data': { + '*in': 'AudiodevPipewirePerDirectionOptions', + '*out': 'AudiodevPipewirePerDirectionOptions', + '*latency': 'uint32' } } + ## # @AudiodevSdlPerDirectionOptions: # @@ -416,6 +458,7 @@ { 'name': 'jack', 'if': 'CONFIG_AUDIO_JACK' }, { 'name': 'oss', 'if': 'CONFIG_AUDIO_OSS' }, { 'name': 'pa', 'if': 'CONFIG_AUDIO_PA' }, + { 'name': 'pipewire', 'if': 'CONFIG_AUDIO_PIPEWIRE' }, { 'name': 'sdl', 'if': 'CONFIG_AUDIO_SDL' }, { 'name': 'sndio', 'if': 'CONFIG_AUDIO_SNDIO' }, { 'name': 'spice', 'if': 'CONFIG_SPICE' }, @@ -456,6 +499,8 @@ 'if': 'CONFIG_AUDIO_OSS' }, 'pa': { 'type': 'AudiodevPaOptions', 'if': 'CONFIG_AUDIO_PA' }, + 'pipewire': { 'type': 'AudiodevPipewireOptions', + 'if': 'CONFIG_AUDIO_PIPEWIRE' }, 'sdl': { 'type': 'AudiodevSdlOptions', 'if': 'CONFIG_AUDIO_SDL' }, 'sndio': { 'type': 'AudiodevSndioOptions', diff --git a/qemu-options.hx b/qemu-options.hx index 88e93c6103..bde4830fab 100644 --- a/qemu-options.hx +++ b/qemu-options.hx @@ -779,6 +779,11 @@ DEF("audiodev", HAS_ARG, QEMU_OPTION_audiodev, " in|out.name= source/sink device name\n" " in|out.latency= desired latency in microseconds\n" #endif +#ifdef CONFIG_AUDIO_PIPEWIRE + "-audiodev pipewire,id=id[,prop[=value][,...]]\n" + " in|out.name= source/sink device name\n" + " latency= desired latency in microseconds\n" +#endif #ifdef CONFIG_AUDIO_SDL "-audiodev sdl,id=id[,prop[=value][,...]]\n" " in|out.buffer-count= number of buffers\n" @@ -942,6 +947,18 @@ SRST Desired latency in microseconds. The PulseAudio server will try to honor this value but actual latencies may be lower or higher. +``-audiodev pipewire,id=id[,prop[=value][,...]]`` + Creates a backend using Pipewire. This backend is available on + most systems. + + Pipewire specific options are: + + ``latency=latency`` + Add extra latency to playback in microseconds + + ``in|out.name=sink`` + Use the specified source/sink for recording/playback. + ``-audiodev sdl,id=id[,prop[=value][,...]]`` Creates a backend using SDL. This backend is available on most systems, but you should use your platform's native backend if diff --git a/scripts/meson-buildoptions.sh b/scripts/meson-buildoptions.sh index 180c11665a..d9f6525346 100644 --- a/scripts/meson-buildoptions.sh +++ b/scripts/meson-buildoptions.sh @@ -1,7 +1,8 @@ # This file is generated by meson-buildoptions.py, do not edit! meson_options_help() { - printf "%s\n" ' --audio-drv-list=CHOICES Set audio driver list [default] (choices: alsa/co' - printf "%s\n" ' reaudio/default/dsound/jack/oss/pa/sdl/sndio)' + printf "%s\n" ' --audio-drv-list=CHOICES Set audio driver list [default] (choices: al' + printf "%s\n" ' sa/coreaudio/default/dsound/jack/oss/pa/' + printf "%s\n" ' pipewire/sdl/sndio)' printf "%s\n" ' --block-drv-ro-whitelist=VALUE' printf "%s\n" ' set block driver read-only whitelist (by default' printf "%s\n" ' affects only QEMU, not tools like qemu-img)' @@ -135,6 +136,7 @@ meson_options_help() { printf "%s\n" ' oss OSS sound support' printf "%s\n" ' pa PulseAudio sound support' printf "%s\n" ' parallels parallels image format support' + printf "%s\n" ' pipewire Pipewire sound support' printf "%s\n" ' png PNG support with libpng' printf "%s\n" ' pvrdma Enable PVRDMA support' printf "%s\n" ' qcow1 qcow1 image format support' @@ -370,6 +372,8 @@ _meson_option_parse() { --disable-pa) printf "%s" -Dpa=disabled ;; --enable-parallels) printf "%s" -Dparallels=enabled ;; --disable-parallels) printf "%s" -Dparallels=disabled ;; + --enable-pipewire) printf "%s" -Dpipewire=enabled ;; + --disable-pipewire) printf "%s" -Dpipewire=disabled ;; --with-pkgversion=*) quote_sh "-Dpkgversion=$2" ;; --enable-png) printf "%s" -Dpng=enabled ;; --disable-png) printf "%s" -Dpng=disabled ;;
This commit adds a new audiodev backend to allow QEMU to use Pipewire as both an audio sink and source. Signed-off-by: Dorinda Bassey <dbassey@redhat.com> --- v2: * Shorten commit message * fix copyright ownership and authour * use QEMU standard of 4 space indentation * verbose use of pipewire instead pf pw audio/audio.c | 3 + audio/audio_template.h | 4 + audio/meson.build | 1 + audio/pwaudio.c | 818 ++++++++++++++++++++++++++++++++++ meson.build | 7 + meson_options.txt | 4 +- qapi/audio.json | 45 ++ qemu-options.hx | 17 + scripts/meson-buildoptions.sh | 8 +- 9 files changed, 904 insertions(+), 3 deletions(-) create mode 100644 audio/pwaudio.c