From 82390d212123514157b315206a2b86489717a669 Mon Sep 17 00:00:00 2001
From: =?UTF-8?q?Thomas=20Wei=C3=9Fschuh?= <thomas@t-8ch.de>
Date: Thu, 6 Jan 2022 21:54:21 +0100
Subject: [PATCH] ao_pipewire: Add PipeWire audio backend

The AO provides a way for mpv to directly submit audio to the PipeWire
audio server.
Doing this directly instead of going through the various compatibility
layers provided by PipeWire has the following advantages:

* It reduces complexity of going through the compatibility layers
* It allows a richer integration between mpv and PipeWire
  (for example for metadata)
* Some users report issues with the compatibility layers that to not
  occur with the native AO

For now the AO is ordered after all the other relevant AOs, so it will
most probably not be picked up by default.
This is for the following reasons:

* Currently it is not possible to detect if the PipeWire daemon that mpv
  connects to is actually driving the system audio.
  (https://gitlab.freedesktop.org/pipewire/pipewire/-/issues/1835)
* It gives the AO time to stabilize before it is used by everyone.

Based-on-patch-by: Oschowa <oschowa@web.de>
Based-on-patch-by: Andreas Kempf <aakempf@gmail.com>
Helped-by: Ivan <etircopyhdot@gmail.com>
(cherry picked from commit 87aba146eddd2bc3fe8819180e2814e7947ff1f2)
---
 DOCS/man/ao.rst         |  11 ++
 audio/out/ao.c          |   4 +
 audio/out/ao_pipewire.c | 396 ++++++++++++++++++++++++++++++++++++++++
 wscript                 |   4 +
 wscript_build.py        |   1 +
 5 files changed, 416 insertions(+)
 create mode 100644 audio/out/ao_pipewire.c

diff --git a/DOCS/man/ao.rst b/DOCS/man/ao.rst
index 97c213a5dd8..ffa30251dd8 100644
--- a/DOCS/man/ao.rst
+++ b/DOCS/man/ao.rst
@@ -146,6 +146,17 @@ Available audio output drivers are:
         Allow mpv to use PulseAudio even if the sink is suspended (default: no).
         Can be useful if PulseAudio is running as a bridge to jack and mpv has its sink-input set to the one jack is using.
 
+``pipewire``
+    PipeWire audio output driver
+
+    The following global options are supported by this audio output:
+
+    ``--pipewire-buffer=<1-2000|native>``
+        Set the audio buffer size in milliseconds. A higher value buffers
+        more data, and has a lower probability of buffer underruns. A smaller
+        value makes the audio stream react faster, e.g. to playback speed
+        changes.
+
 ``sdl``
     SDL 1.2+ audio output driver. Should work on any platform supported by SDL
     1.2, but may require the ``SDL_AUDIODRIVER`` environment variable to be set
diff --git a/audio/out/ao.c b/audio/out/ao.c
index 950f935c812..48bd695e118 100644
--- a/audio/out/ao.c
+++ b/audio/out/ao.c
@@ -41,6 +41,7 @@ extern const struct ao_driver audio_out_audiounit;
 extern const struct ao_driver audio_out_coreaudio;
 extern const struct ao_driver audio_out_coreaudio_exclusive;
 extern const struct ao_driver audio_out_rsound;
+extern const struct ao_driver audio_out_pipewire;
 extern const struct ao_driver audio_out_sndio;
 extern const struct ao_driver audio_out_pulse;
 extern const struct ao_driver audio_out_jack;
@@ -89,6 +90,9 @@ static const struct ao_driver * const audio_out_drivers[] = {
 #if HAVE_SDL2_AUDIO
     &audio_out_sdl,
 #endif
+#if HAVE_PIPEWIRE
+    &audio_out_pipewire,
+#endif
 #if HAVE_SNDIO
     &audio_out_sndio,
 #endif
diff --git a/audio/out/ao_pipewire.c b/audio/out/ao_pipewire.c
new file mode 100644
index 00000000000..ab2fcf576b7
--- /dev/null
+++ b/audio/out/ao_pipewire.c
@@ -0,0 +1,396 @@
+
+#include <pipewire/pipewire.h>
+#include <spa/param/audio/format-utils.h>
+#include <spa/param/props.h>
+#include <math.h>
+
+#include "common/msg.h"
+#include "options/m_config.h"
+#include "options/m_option.h"
+#include "ao.h"
+#include "audio/format.h"
+#include "config.h"
+#include "generated/version.h"
+#include "internal.h"
+#include "osdep/timer.h"
+
+// Added in Pipewire 0.3.33
+// remove the fallback when we require a newer version
+#ifndef PW_KEY_NODE_RATE
+#define PW_KEY_NODE_RATE "node.rate"
+#endif
+
+struct priv {
+    struct pw_thread_loop *loop;
+    struct pw_stream *stream;
+
+    int buffer_msec;
+    bool muted;
+    float volume[2];
+};
+
+static enum spa_audio_format af_fmt_to_pw(struct ao *ao, enum af_format format)
+{
+    switch (format) {
+    case AF_FORMAT_U8:          return SPA_AUDIO_FORMAT_U8;
+    case AF_FORMAT_S16:         return SPA_AUDIO_FORMAT_S16;
+    case AF_FORMAT_S32:         return SPA_AUDIO_FORMAT_S32;
+    case AF_FORMAT_FLOAT:       return SPA_AUDIO_FORMAT_F32;
+    case AF_FORMAT_DOUBLE:      return SPA_AUDIO_FORMAT_F64;
+    case AF_FORMAT_U8P:         return SPA_AUDIO_FORMAT_U8P;
+    case AF_FORMAT_S16P:        return SPA_AUDIO_FORMAT_S16P;
+    case AF_FORMAT_S32P:        return SPA_AUDIO_FORMAT_S32P;
+    case AF_FORMAT_FLOATP:      return SPA_AUDIO_FORMAT_F32P;
+    case AF_FORMAT_DOUBLEP:     return SPA_AUDIO_FORMAT_F64P;
+    default:
+                                MP_WARN(ao, "Unhandled format %d\n", format);
+                                return SPA_AUDIO_FORMAT_UNKNOWN;
+    }
+}
+
+static enum spa_audio_channel mp_speaker_id_to_spa(struct ao *ao, enum mp_speaker_id mp_speaker_id)
+{
+    switch (mp_speaker_id) {
+    case MP_SPEAKER_ID_FL:   return SPA_AUDIO_CHANNEL_FL;
+    case MP_SPEAKER_ID_FR:   return SPA_AUDIO_CHANNEL_FR;
+    case MP_SPEAKER_ID_FC:   return SPA_AUDIO_CHANNEL_FC;
+    case MP_SPEAKER_ID_LFE:  return SPA_AUDIO_CHANNEL_LFE;
+    case MP_SPEAKER_ID_BL:   return SPA_AUDIO_CHANNEL_RL;
+    case MP_SPEAKER_ID_BR:   return SPA_AUDIO_CHANNEL_RR;
+    case MP_SPEAKER_ID_FLC:  return SPA_AUDIO_CHANNEL_FLC;
+    case MP_SPEAKER_ID_FRC:  return SPA_AUDIO_CHANNEL_FRC;
+    case MP_SPEAKER_ID_BC:   return SPA_AUDIO_CHANNEL_RC;
+    case MP_SPEAKER_ID_SL:   return SPA_AUDIO_CHANNEL_SL;
+    case MP_SPEAKER_ID_SR:   return SPA_AUDIO_CHANNEL_SR;
+    case MP_SPEAKER_ID_TC:   return SPA_AUDIO_CHANNEL_TC;
+    case MP_SPEAKER_ID_TFL:  return SPA_AUDIO_CHANNEL_TFL;
+    case MP_SPEAKER_ID_TFC:  return SPA_AUDIO_CHANNEL_TFC;
+    case MP_SPEAKER_ID_TFR:  return SPA_AUDIO_CHANNEL_TFR;
+    case MP_SPEAKER_ID_TBL:  return SPA_AUDIO_CHANNEL_TRL;
+    case MP_SPEAKER_ID_TBC:  return SPA_AUDIO_CHANNEL_TRC;
+    case MP_SPEAKER_ID_TBR:  return SPA_AUDIO_CHANNEL_TRR;
+    case MP_SPEAKER_ID_DL:   return SPA_AUDIO_CHANNEL_FL;
+    case MP_SPEAKER_ID_DR:   return SPA_AUDIO_CHANNEL_FR;
+    case MP_SPEAKER_ID_WL:   return SPA_AUDIO_CHANNEL_FL;
+    case MP_SPEAKER_ID_WR:   return SPA_AUDIO_CHANNEL_FR;
+    case MP_SPEAKER_ID_SDL:  return SPA_AUDIO_CHANNEL_SL;
+    case MP_SPEAKER_ID_SDR:  return SPA_AUDIO_CHANNEL_SL;
+    case MP_SPEAKER_ID_LFE2: return SPA_AUDIO_CHANNEL_LFE2;
+    case MP_SPEAKER_ID_NA:   return SPA_AUDIO_CHANNEL_NA;
+    default:
+                             MP_WARN(ao, "Unhandled channel %d\n", mp_speaker_id);
+                             return SPA_AUDIO_CHANNEL_UNKNOWN;
+    };
+}
+
+static void on_process(void *userdata)
+{
+    struct ao *ao = userdata;
+    struct priv *p = ao->priv;
+    struct pw_time time;
+    struct pw_buffer *b;
+    void *data[MP_NUM_CHANNELS];
+
+    if ((b = pw_stream_dequeue_buffer(p->stream)) == NULL) {
+        pw_log_warn("out of buffers: %m");
+        return;
+    }
+
+    struct spa_buffer *buf = b->buffer;
+
+    int bytes_per_channel = buf->datas[0].maxsize / ao->channels.num;
+    int nframes = bytes_per_channel / ao->sstride;
+
+    for (int i = 0; i < buf->n_datas; i++) {
+        data[i] = buf->datas[i].data;
+        buf->datas[i].chunk->size = bytes_per_channel;
+        buf->datas[i].chunk->offset = 0;
+    }
+
+    pw_stream_get_time(p->stream, &time);
+    if (time.rate.denom == 0)
+        time.rate.denom = ao->samplerate;
+    if (time.rate.num == 0)
+        time.rate.num = 1;
+
+    int64_t end_time = mp_time_us();
+    /* time.queued is always going to be 0, so we don't need to care */
+    end_time += (nframes * 1e6 / ao->samplerate) +
+                ((float) time.delay * SPA_USEC_PER_SEC * time.rate.num / time.rate.denom);
+
+    ao_read_data(ao, data, nframes, end_time);
+
+    pw_stream_queue_buffer(p->stream, b);
+}
+
+static void on_param_changed(void *userdata, uint32_t id, const struct spa_pod *param)
+{
+    struct ao *ao = userdata;
+    struct priv *p = ao->priv;
+    const struct spa_pod *params[1];
+    uint8_t buffer[1024];
+    struct spa_pod_builder b = SPA_POD_BUILDER_INIT(buffer, sizeof(buffer));
+
+    if (param == NULL || id != SPA_PARAM_Format)
+        return;
+
+    int buffer_size = ao->device_buffer * af_fmt_to_bytes(ao->format) * ao->channels.num;
+
+    params[0] = spa_pod_builder_add_object(&b,
+                    SPA_TYPE_OBJECT_ParamBuffers, SPA_PARAM_Buffers,
+                    SPA_PARAM_BUFFERS_blocks,     SPA_POD_Int(ao->num_planes),
+                    SPA_PARAM_BUFFERS_size,       SPA_POD_Int(buffer_size),
+                    SPA_PARAM_BUFFERS_stride,     SPA_POD_Int(ao->sstride));
+
+    pw_stream_update_params(p->stream, params, 1);
+}
+
+static void on_state_changed(void *userdata, enum pw_stream_state old, enum pw_stream_state state, const char *error)
+{
+    struct ao *ao = userdata;
+    MP_DBG(ao, "Stream state changed: old_state=%d state=%d error=%s\n", old, state, error);
+
+    if (state == PW_STREAM_STATE_ERROR) {
+        MP_WARN(ao, "Stream in error state, trying to reload...\n");
+        ao_request_reload(ao);
+    }
+}
+
+static float spa_volume_to_mp_volume(float vol)
+{
+        return cbrt(vol) * 100;
+}
+
+static float mp_volume_to_spa_volume(float vol)
+{
+        vol /= 100;
+        return vol * vol * vol;
+}
+
+static void on_control_info(void *userdata, uint32_t id,
+        const struct pw_stream_control *control)
+{
+    struct ao *ao = userdata;
+    struct priv *p = ao->priv;
+
+    switch (id) {
+        case SPA_PROP_mute:
+            if (control->n_values == 1)
+                p->muted = control->values[0] >= 0.5;
+            break;
+        case SPA_PROP_channelVolumes:
+            if (control->n_values == 1) {
+                p->volume[0] = control->values[0];
+                p->volume[1] = control->values[0];
+            } else if (control->n_values == 2) {
+                p->volume[0] = control->values[0];
+                p->volume[1] = control->values[1];
+            }
+            break;
+    }
+}
+
+static const struct pw_stream_events stream_events = {
+    .version = PW_VERSION_STREAM_EVENTS,
+    .param_changed = on_param_changed,
+    .process = on_process,
+    .state_changed = on_state_changed,
+    .control_info = on_control_info,
+};
+
+static void uninit(struct ao *ao)
+{
+    struct priv *p = ao->priv;
+    if (p->loop)
+        pw_thread_loop_stop(p->loop);
+    if (p->stream)
+        pw_stream_destroy(p->stream);
+    p->stream = NULL;
+    if (p->loop)
+        pw_thread_loop_destroy(p->loop);
+    p->loop = NULL;
+    pw_deinit();
+}
+
+static int init(struct ao *ao)
+{
+    struct priv *p = ao->priv;
+    uint8_t buffer[1024];
+    struct spa_pod_builder b = SPA_POD_BUILDER_INIT(buffer, sizeof(buffer));
+    const struct spa_pod *params[1];
+    struct pw_properties *props = pw_properties_new(
+        PW_KEY_MEDIA_TYPE, "Audio",
+        PW_KEY_MEDIA_CATEGORY, "Playback",
+        PW_KEY_MEDIA_ROLE, "Movie",
+        PW_KEY_NODE_NAME, ao->client_name,
+        PW_KEY_NODE_DESCRIPTION, ao->client_name,
+        PW_KEY_APP_NAME, ao->client_name,
+        PW_KEY_APP_ID, ao->client_name,
+        PW_KEY_APP_ICON_NAME, ao->client_name,
+        PW_KEY_NODE_ALWAYS_PROCESS, "true",
+        NULL
+    );
+
+    ao->device_buffer = p->buffer_msec * ao->samplerate / 1000;
+
+    pw_properties_setf(props, PW_KEY_NODE_LATENCY, "%d/%d", ao->device_buffer, ao->samplerate);
+    pw_properties_setf(props, PW_KEY_NODE_RATE, "1/%d", ao->samplerate);
+
+    enum spa_audio_format spa_format = af_fmt_to_pw(ao, ao->format);
+    if (spa_format == SPA_AUDIO_FORMAT_UNKNOWN) {
+        ao->format = AF_FORMAT_FLOATP;
+        spa_format = SPA_AUDIO_FORMAT_F32P;
+    }
+
+    struct spa_audio_info_raw audio_info = {
+        .format = spa_format,
+        .rate = ao->samplerate,
+        .channels = ao->channels.num,
+    };
+
+    for (int i = 0; i < ao->channels.num; i++)
+        audio_info.position[i] = mp_speaker_id_to_spa(ao, ao->channels.speaker[i]);
+
+    params[0] = spa_format_audio_raw_build(&b, SPA_PARAM_EnumFormat, &audio_info);
+
+    if (af_fmt_is_planar(ao->format)) {
+        ao->num_planes = ao->channels.num;
+        ao->sstride = af_fmt_to_bytes(ao->format);
+    } else {
+        ao->num_planes = 1;
+        ao->sstride = ao->channels.num * af_fmt_to_bytes(ao->format);
+    }
+
+    pw_init(NULL, NULL);
+
+    p->loop = pw_thread_loop_new("ao-pipewire", NULL);
+    if (p->loop == NULL)
+        goto error;
+
+    p->stream = pw_stream_new_simple(
+                    pw_thread_loop_get_loop(p->loop),
+                    "audio-src",
+                    props,
+                    &stream_events,
+                    ao);
+    if (p->stream == NULL)
+        goto error;
+
+    if (pw_stream_connect(p->stream,
+                    PW_DIRECTION_OUTPUT,
+                    PW_ID_ANY,
+                    PW_STREAM_FLAG_AUTOCONNECT |
+                    PW_STREAM_FLAG_INACTIVE |
+                    PW_STREAM_FLAG_MAP_BUFFERS |
+                    PW_STREAM_FLAG_RT_PROCESS,
+                    params, 1) < 0)
+        goto error;
+
+    if (pw_thread_loop_start(p->loop) < 0)
+        goto error;
+
+    return 0;
+
+error:
+    uninit(ao);
+    return -1;
+}
+
+static void reset(struct ao *ao)
+{
+    struct priv *p = ao->priv;
+    pw_thread_loop_lock(p->loop);
+    pw_stream_set_active(p->stream, false);
+    pw_stream_flush(p->stream, false);
+    pw_thread_loop_unlock(p->loop);
+}
+
+static void start(struct ao *ao)
+{
+    struct priv *p = ao->priv;
+    pw_thread_loop_lock(p->loop);
+    pw_stream_set_active(p->stream, true);
+    pw_thread_loop_unlock(p->loop);
+}
+
+#define CONTROL_RET(r) (!r ? CONTROL_OK : CONTROL_ERROR)
+
+static int control(struct ao *ao, enum aocontrol cmd, void *arg)
+{
+    struct priv *p = ao->priv;
+
+    switch (cmd) {
+        case AOCONTROL_GET_VOLUME: {
+                struct ao_control_vol *vol = arg;
+                vol->left = spa_volume_to_mp_volume(p->volume[0]);
+                vol->right = spa_volume_to_mp_volume(p->volume[1]);
+                return CONTROL_OK;
+        }
+        case AOCONTROL_GET_MUTE: {
+                bool *muted = arg;
+                *muted = p->muted;
+                return CONTROL_OK;
+        }
+        case AOCONTROL_SET_VOLUME:
+        case AOCONTROL_SET_MUTE:
+        case AOCONTROL_UPDATE_STREAM_TITLE: {
+            int ret;
+
+            pw_thread_loop_lock(p->loop);
+            switch (cmd) {
+                case AOCONTROL_SET_VOLUME: {
+                    struct ao_control_vol *vol = arg;
+                    float left = mp_volume_to_spa_volume(vol->left), right = mp_volume_to_spa_volume(vol->right);
+                    ret = CONTROL_RET(pw_stream_set_control(p->stream, SPA_PROP_channelVolumes, 2, &left, &right));
+                    break;
+               }
+                case AOCONTROL_SET_MUTE: {
+                    bool *muted = arg;
+                    float value = *muted ? 1.f : 0.f;
+                    ret = CONTROL_RET(pw_stream_set_control(p->stream, SPA_PROP_mute, 1, &value));
+                    break;
+                }
+                case AOCONTROL_UPDATE_STREAM_TITLE: {
+                    char *title = arg;
+                    struct spa_dict_item items[1];
+                    items[0] = SPA_DICT_ITEM_INIT(PW_KEY_MEDIA_NAME, title);
+                    ret = CONTROL_RET(pw_stream_update_properties(p->stream, &SPA_DICT_INIT(items, MP_ARRAY_SIZE(items))));
+                    break;
+                }
+                default:
+                    ret = CONTROL_NA;
+            }
+            pw_thread_loop_unlock(p->loop);
+            return ret;
+        }
+        default:
+            return CONTROL_UNKNOWN;
+    }
+}
+
+#define OPT_BASE_STRUCT struct priv
+
+const struct ao_driver audio_out_pipewire = {
+    .description = "PipeWire audio output",
+    .name        = "pipewire",
+
+    .init        = init,
+    .uninit      = uninit,
+    .reset       = reset,
+    .start       = start,
+
+    .control     = control,
+
+    .priv_size = sizeof(struct priv),
+    .priv_defaults = &(const struct priv)
+    {
+        .loop = NULL,
+        .stream = NULL,
+        .buffer_msec = 20,
+    },
+    .options_prefix = "pipewire",
+    .options = (const struct m_option[]) {
+        {"buffer", OPT_INT(buffer_msec), M_RANGE(1, 2000)},
+        {0}
+    },
+};
diff --git a/wscript b/wscript
index 1651187b347..4188d4686f7 100644
--- a/wscript
+++ b/wscript
@@ -434,6 +434,10 @@ audio_output_features = [
         'desc': 'OSSv4 audio output',
         'func': check_statement(['sys/soundcard.h'], 'int x = SNDCTL_DSP_SETPLAYVOL'),
         'deps': 'posix && gpl',
+    }, {
+        'name': '--pipewire',
+        'desc': 'PipeWire audio output',
+        'func': check_pkg_config('libpipewire-0.3', '>= 0.3.0')
     }, {
         'name': '--sndio',
         'desc': 'sndio audio input/output',
diff --git a/wscript_build.py b/wscript_build.py
index 8244dbe3d3a..3d303825a7d 100644
--- a/wscript_build.py
+++ b/wscript_build.py
@@ -246,6 +246,7 @@ def swift(task):
         ( "audio/out/ao_opensles.c",             "opensles" ),
         ( "audio/out/ao_oss.c",                  "oss-audio" ),
         ( "audio/out/ao_pcm.c" ),
+        ( "audio/out/ao_pipewire.c",             "pipewire" ),
         ( "audio/out/ao_pulse.c",                "pulse" ),
         ( "audio/out/ao_sdl.c",                  "sdl2-audio" ),
         ( "audio/out/ao_sndio.c",                "sndio" ),
