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mpv/audio/out/ao_pulse.c
wm4 c1002f6a28 ao_pulse: attempt to fall back to an arbitrary sample format 
Normally, PulseAudio accepts any combination of sample format, sample
rate, channel count/map. Sometimes it does not. For example, the channel
rate or channel count have fixed maximum values. We should not fail
fatally in such cases, but attempt to fall back to a working format.

We could just send pass an "unset" format to Pulse, but this is not too
attractive. Pulse could use a format which we do not support, and also
doing so much for an obscure corner case is not reasonable. So just pick
a format that is very likely supported.

This still could fail at runtime (the stream could fail instead of going
to the ready state), but this sounds also too complicated. In
particular, it doesn't look like pulse will tell us the cause of the
stream failure. (Or maybe it does - but I didn't find anything.)

Last but not least, our fallback could be less dumb, and e.g. try to fix
only one of samplerate or channel count first to reduce the loss, but
this is also not particularly worthy the effort.

Fixes #2654.
2016-01-05 19:52:05 +01:00

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/*
* PulseAudio audio output driver.
* Copyright (C) 2006 Lennart Poettering
* Copyright (C) 2007 Reimar Doeffinger
*
* This file is part of mpv.
*
* mpv is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License as published by
* the Free Software Foundation; either version 2 of the License, or
* (at your option) any later version.
*
* mpv is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
* GNU General Public License for more details.
*
* You should have received a copy of the GNU General Public License along
* with mpv. If not, see <http://www.gnu.org/licenses/>.
*/
#include <stdlib.h>
#include <stdbool.h>
#include <string.h>
#include <stdint.h>
#include <pthread.h>
#include <pulse/pulseaudio.h>
#include "config.h"
#include "audio/format.h"
#include "common/msg.h"
#include "options/m_option.h"
#include "ao.h"
#include "internal.h"
#define VOL_PA2MP(v) ((v) * 100 / PA_VOLUME_NORM)
#define VOL_MP2PA(v) ((v) * PA_VOLUME_NORM / 100)
struct priv {
// PulseAudio playback stream object
struct pa_stream *stream;
// PulseAudio connection context
struct pa_context *context;
// Main event loop object
struct pa_threaded_mainloop *mainloop;
// temporary during control()
struct pa_sink_input_info pi;
int retval;
// for wakeup handling
pthread_mutex_t wakeup_lock;
pthread_cond_t wakeup;
int wakeup_status;
char *cfg_host;
char *cfg_sink;
int cfg_buffer;
int cfg_latency_hacks;
};
#define GENERIC_ERR_MSG(str) \
MP_ERR(ao, str": %s\n", \
pa_strerror(pa_context_errno(((struct priv *)ao->priv)->context)))
static void context_state_cb(pa_context *c, void *userdata)
{
struct ao *ao = userdata;
struct priv *priv = ao->priv;
switch (pa_context_get_state(c)) {
case PA_CONTEXT_READY:
case PA_CONTEXT_TERMINATED:
case PA_CONTEXT_FAILED:
pa_threaded_mainloop_signal(priv->mainloop, 0);
break;
}
}
static void subscribe_cb(pa_context *c, pa_subscription_event_type_t t,
uint32_t idx, void *userdata)
{
struct ao *ao = userdata;
int type = t & PA_SUBSCRIPTION_MASK_SINK;
int fac = t & PA_SUBSCRIPTION_EVENT_FACILITY_MASK;
if ((type == PA_SUBSCRIPTION_EVENT_NEW || type == PA_SUBSCRIPTION_EVENT_REMOVE)
&& fac == PA_SUBSCRIPTION_EVENT_SINK)
{
ao_hotplug_event(ao);
}
}
static void context_success_cb(pa_context *c, int success, void *userdata)
{
struct ao *ao = userdata;
struct priv *priv = ao->priv;
priv->retval = success;
pa_threaded_mainloop_signal(priv->mainloop, 0);
}
static void stream_state_cb(pa_stream *s, void *userdata)
{
struct ao *ao = userdata;
struct priv *priv = ao->priv;
switch (pa_stream_get_state(s)) {
case PA_STREAM_FAILED:
MP_VERBOSE(ao, "Stream failed.\n");
ao_request_reload(ao);
pa_threaded_mainloop_signal(priv->mainloop, 0);
break;
case PA_STREAM_READY:
case PA_STREAM_TERMINATED:
pa_threaded_mainloop_signal(priv->mainloop, 0);
break;
}
}
static void wakeup(struct ao *ao)
{
struct priv *priv = ao->priv;
pthread_mutex_lock(&priv->wakeup_lock);
priv->wakeup_status = 1;
pthread_cond_signal(&priv->wakeup);
pthread_mutex_unlock(&priv->wakeup_lock);
}
static void stream_request_cb(pa_stream *s, size_t length, void *userdata)
{
struct ao *ao = userdata;
struct priv *priv = ao->priv;
wakeup(ao);
pa_threaded_mainloop_signal(priv->mainloop, 0);
}
static int wait_audio(struct ao *ao, pthread_mutex_t *lock)
{
struct priv *priv = ao->priv;
// We don't use this mutex, because pulse like to call stream_request_cb
// while we have the central mutex held.
pthread_mutex_unlock(lock);
pthread_mutex_lock(&priv->wakeup_lock);
while (!priv->wakeup_status)
pthread_cond_wait(&priv->wakeup, &priv->wakeup_lock);
priv->wakeup_status = 0;
pthread_mutex_unlock(&priv->wakeup_lock);
pthread_mutex_lock(lock);
return 0;
}
static void stream_latency_update_cb(pa_stream *s, void *userdata)
{
struct ao *ao = userdata;
struct priv *priv = ao->priv;
pa_threaded_mainloop_signal(priv->mainloop, 0);
}
static void success_cb(pa_stream *s, int success, void *userdata)
{
struct ao *ao = userdata;
struct priv *priv = ao->priv;
priv->retval = success;
pa_threaded_mainloop_signal(priv->mainloop, 0);
}
/**
* \brief waits for a pulseaudio operation to finish, frees it and
* unlocks the mainloop
* \param op operation to wait for
* \return 1 if operation has finished normally (DONE state), 0 otherwise
*/
static int waitop(struct priv *priv, pa_operation *op)
{
if (!op) {
pa_threaded_mainloop_unlock(priv->mainloop);
return 0;
}
pa_operation_state_t state = pa_operation_get_state(op);
while (state == PA_OPERATION_RUNNING) {
pa_threaded_mainloop_wait(priv->mainloop);
state = pa_operation_get_state(op);
}
pa_operation_unref(op);
pa_threaded_mainloop_unlock(priv->mainloop);
return state == PA_OPERATION_DONE;
}
static const struct format_map {
int mp_format;
pa_sample_format_t pa_format;
} format_maps[] = {
{AF_FORMAT_S16, PA_SAMPLE_S16NE},
{AF_FORMAT_S32, PA_SAMPLE_S32NE},
{AF_FORMAT_FLOAT, PA_SAMPLE_FLOAT32NE},
{AF_FORMAT_U8, PA_SAMPLE_U8},
{AF_FORMAT_UNKNOWN, 0}
};
static pa_encoding_t map_digital_format(int format)
{
switch (format) {
case AF_FORMAT_S_AC3: return PA_ENCODING_AC3_IEC61937;
case AF_FORMAT_S_EAC3: return PA_ENCODING_EAC3_IEC61937;
case AF_FORMAT_S_MP3: return PA_ENCODING_MPEG_IEC61937;
case AF_FORMAT_S_DTS:
case AF_FORMAT_S_DTSHD: return PA_ENCODING_DTS_IEC61937;
#ifdef PA_ENCODING_MPEG2_AAC_IEC61937
case AF_FORMAT_S_AAC: return PA_ENCODING_MPEG2_AAC_IEC61937;
#endif
default:
if (af_fmt_is_spdif(format))
return PA_ENCODING_ANY;
return PA_ENCODING_PCM;
}
}
static const int speaker_map[][2] = {
{PA_CHANNEL_POSITION_FRONT_LEFT, MP_SPEAKER_ID_FL},
{PA_CHANNEL_POSITION_FRONT_RIGHT, MP_SPEAKER_ID_FR},
{PA_CHANNEL_POSITION_FRONT_CENTER, MP_SPEAKER_ID_FC},
{PA_CHANNEL_POSITION_REAR_CENTER, MP_SPEAKER_ID_BC},
{PA_CHANNEL_POSITION_REAR_LEFT, MP_SPEAKER_ID_BL},
{PA_CHANNEL_POSITION_REAR_RIGHT, MP_SPEAKER_ID_BR},
{PA_CHANNEL_POSITION_LFE, MP_SPEAKER_ID_LFE},
{PA_CHANNEL_POSITION_FRONT_LEFT_OF_CENTER, MP_SPEAKER_ID_FLC},
{PA_CHANNEL_POSITION_FRONT_RIGHT_OF_CENTER, MP_SPEAKER_ID_FRC},
{PA_CHANNEL_POSITION_SIDE_LEFT, MP_SPEAKER_ID_SL},
{PA_CHANNEL_POSITION_SIDE_RIGHT, MP_SPEAKER_ID_SR},
{PA_CHANNEL_POSITION_TOP_CENTER, MP_SPEAKER_ID_TC},
{PA_CHANNEL_POSITION_TOP_FRONT_LEFT, MP_SPEAKER_ID_TFL},
{PA_CHANNEL_POSITION_TOP_FRONT_RIGHT, MP_SPEAKER_ID_TFR},
{PA_CHANNEL_POSITION_TOP_FRONT_CENTER, MP_SPEAKER_ID_TFC},
{PA_CHANNEL_POSITION_TOP_REAR_LEFT, MP_SPEAKER_ID_TBL},
{PA_CHANNEL_POSITION_TOP_REAR_RIGHT, MP_SPEAKER_ID_TBR},
{PA_CHANNEL_POSITION_TOP_REAR_CENTER, MP_SPEAKER_ID_TBC},
{PA_CHANNEL_POSITION_INVALID, -1}
};
static bool chmap_pa_from_mp(pa_channel_map *dst, struct mp_chmap *src)
{
if (src->num > PA_CHANNELS_MAX)
return false;
dst->channels = src->num;
if (mp_chmap_equals(src, &(const struct mp_chmap)MP_CHMAP_INIT_MONO)) {
dst->map[0] = PA_CHANNEL_POSITION_MONO;
return true;
}
for (int n = 0; n < src->num; n++) {
int mp_speaker = src->speaker[n];
int pa_speaker = PA_CHANNEL_POSITION_INVALID;
for (int i = 0; speaker_map[i][1] != -1; i++) {
if (speaker_map[i][1] == mp_speaker) {
pa_speaker = speaker_map[i][0];
break;
}
}
if (pa_speaker == PA_CHANNEL_POSITION_INVALID)
return false;
dst->map[n] = pa_speaker;
}
return true;
}
static bool select_chmap(struct ao *ao, pa_channel_map *dst)
{
struct mp_chmap_sel sel = {0};
for (int n = 0; speaker_map[n][1] != -1; n++)
mp_chmap_sel_add_speaker(&sel, speaker_map[n][1]);
return ao_chmap_sel_adjust(ao, &sel, &ao->channels) &&
chmap_pa_from_mp(dst, &ao->channels);
}
static void drain(struct ao *ao)
{
struct priv *priv = ao->priv;
if (priv->stream) {
pa_threaded_mainloop_lock(priv->mainloop);
waitop(priv, pa_stream_drain(priv->stream, success_cb, ao));
}
}
static void uninit(struct ao *ao)
{
struct priv *priv = ao->priv;
if (priv->mainloop)
pa_threaded_mainloop_stop(priv->mainloop);
if (priv->stream) {
pa_stream_disconnect(priv->stream);
pa_stream_unref(priv->stream);
priv->stream = NULL;
}
if (priv->context) {
pa_context_disconnect(priv->context);
pa_context_unref(priv->context);
priv->context = NULL;
}
if (priv->mainloop) {
pa_threaded_mainloop_free(priv->mainloop);
priv->mainloop = NULL;
}
pthread_cond_destroy(&priv->wakeup);
pthread_mutex_destroy(&priv->wakeup_lock);
}
static int pa_init_boilerplate(struct ao *ao)
{
struct priv *priv = ao->priv;
char *host = priv->cfg_host && priv->cfg_host[0] ? priv->cfg_host : NULL;
bool locked = false;
pthread_mutex_init(&priv->wakeup_lock, NULL);
pthread_cond_init(&priv->wakeup, NULL);
if (!(priv->mainloop = pa_threaded_mainloop_new())) {
MP_ERR(ao, "Failed to allocate main loop\n");
goto fail;
}
if (pa_threaded_mainloop_start(priv->mainloop) < 0)
goto fail;
pa_threaded_mainloop_lock(priv->mainloop);
locked = true;
if (!(priv->context = pa_context_new(pa_threaded_mainloop_get_api(
priv->mainloop), ao->client_name)))
{
MP_ERR(ao, "Failed to allocate context\n");
goto fail;
}
MP_VERBOSE(ao, "Library version: %s\n", pa_get_library_version());
MP_VERBOSE(ao, "Proto: %lu\n",
(long)pa_context_get_protocol_version(priv->context));
MP_VERBOSE(ao, "Server proto: %lu\n",
(long)pa_context_get_server_protocol_version(priv->context));
pa_context_set_state_callback(priv->context, context_state_cb, ao);
pa_context_set_subscribe_callback(priv->context, subscribe_cb, ao);
if (pa_context_connect(priv->context, host, 0, NULL) < 0)
goto fail;
/* Wait until the context is ready */
while (1) {
int state = pa_context_get_state(priv->context);
if (state == PA_CONTEXT_READY)
break;
if (!PA_CONTEXT_IS_GOOD(state))
goto fail;
pa_threaded_mainloop_wait(priv->mainloop);
}
pa_threaded_mainloop_unlock(priv->mainloop);
return 0;
fail:
if (locked)
pa_threaded_mainloop_unlock(priv->mainloop);
if (priv->context) {
pa_threaded_mainloop_lock(priv->mainloop);
if (!(pa_context_errno(priv->context) == PA_ERR_CONNECTIONREFUSED
&& ao->probing))
GENERIC_ERR_MSG("Init failed");
pa_threaded_mainloop_unlock(priv->mainloop);
}
uninit(ao);
return -1;
}
static bool set_format(struct ao *ao, pa_format_info *format)
{
ao->format = af_fmt_from_planar(ao->format);
format->encoding = map_digital_format(ao->format);
if (format->encoding == PA_ENCODING_PCM) {
const struct format_map *fmt_map = format_maps;
while (fmt_map->mp_format != ao->format) {
if (fmt_map->mp_format == AF_FORMAT_UNKNOWN) {
MP_VERBOSE(ao, "Unsupported format, using default\n");
fmt_map = format_maps;
break;
}
fmt_map++;
}
ao->format = fmt_map->mp_format;
pa_format_info_set_sample_format(format, fmt_map->pa_format);
}
struct pa_channel_map map;
if (!select_chmap(ao, &map))
return false;
pa_format_info_set_rate(format, ao->samplerate);
pa_format_info_set_channels(format, ao->channels.num);
pa_format_info_set_channel_map(format, &map);
return ao->samplerate < PA_RATE_MAX && pa_format_info_valid(format);
}
static int init(struct ao *ao)
{
pa_proplist *proplist = NULL;
pa_format_info *format = NULL;
struct priv *priv = ao->priv;
char *sink = priv->cfg_sink && priv->cfg_sink[0] ? priv->cfg_sink : ao->device;
if (pa_init_boilerplate(ao) < 0)
return -1;
pa_threaded_mainloop_lock(priv->mainloop);
if (!(proplist = pa_proplist_new())) {
MP_ERR(ao, "Failed to allocate proplist\n");
goto unlock_and_fail;
}
(void)pa_proplist_sets(proplist, PA_PROP_MEDIA_ICON_NAME, ao->client_name);
if (!(format = pa_format_info_new()))
goto unlock_and_fail;
if (!set_format(ao, format)) {
ao->channels = (struct mp_chmap) MP_CHMAP_INIT_STEREO;
ao->samplerate = 48000;
ao->format = AF_FORMAT_FLOAT;
if (!set_format(ao, format)) {
MP_ERR(ao, "Invalid audio format\n");
goto unlock_and_fail;
}
}
if (!(priv->stream = pa_stream_new_extended(priv->context, "audio stream",
&format, 1, proplist)))
goto unlock_and_fail;
pa_format_info_free(format);
format = NULL;
pa_proplist_free(proplist);
proplist = NULL;
pa_stream_set_state_callback(priv->stream, stream_state_cb, ao);
pa_stream_set_write_callback(priv->stream, stream_request_cb, ao);
pa_stream_set_latency_update_callback(priv->stream,
stream_latency_update_cb, ao);
int buf_size = af_fmt_seconds_to_bytes(ao->format, priv->cfg_buffer / 1000.0,
ao->channels.num, ao->samplerate);
pa_buffer_attr bufattr = {
.maxlength = -1,
.tlength = buf_size > 0 ? buf_size : (uint32_t)-1,
.prebuf = -1,
.minreq = -1,
.fragsize = -1,
};
int flags = PA_STREAM_NOT_MONOTONIC;
if (!priv->cfg_latency_hacks)
flags |= PA_STREAM_INTERPOLATE_TIMING|PA_STREAM_AUTO_TIMING_UPDATE;
if (pa_stream_connect_playback(priv->stream, sink, &bufattr,
flags, NULL, NULL) < 0)
goto unlock_and_fail;
/* Wait until the stream is ready */
while (1) {
int state = pa_stream_get_state(priv->stream);
if (state == PA_STREAM_READY)
break;
if (!PA_STREAM_IS_GOOD(state))
goto unlock_and_fail;
pa_threaded_mainloop_wait(priv->mainloop);
}
if (pa_stream_is_suspended(priv->stream)) {
MP_ERR(ao, "The stream is suspended. Bailing out.\n");
goto unlock_and_fail;
}
pa_threaded_mainloop_unlock(priv->mainloop);
return 0;
unlock_and_fail:
pa_threaded_mainloop_unlock(priv->mainloop);
if (format)
pa_format_info_free(format);
if (proplist)
pa_proplist_free(proplist);
uninit(ao);
return -1;
}
static void cork(struct ao *ao, bool pause)
{
struct priv *priv = ao->priv;
pa_threaded_mainloop_lock(priv->mainloop);
priv->retval = 0;
if (!waitop(priv, pa_stream_cork(priv->stream, pause, success_cb, ao)) ||
!priv->retval)
GENERIC_ERR_MSG("pa_stream_cork() failed");
}
// Play the specified data to the pulseaudio server
static int play(struct ao *ao, void **data, int samples, int flags)
{
struct priv *priv = ao->priv;
pa_threaded_mainloop_lock(priv->mainloop);
if (pa_stream_write(priv->stream, data[0], samples * ao->sstride, NULL, 0,
PA_SEEK_RELATIVE) < 0) {
GENERIC_ERR_MSG("pa_stream_write() failed");
samples = -1;
}
if (flags & AOPLAY_FINAL_CHUNK) {
// Force start in case the stream was too short for prebuf
pa_operation *op = pa_stream_trigger(priv->stream, NULL, NULL);
pa_operation_unref(op);
}
pa_threaded_mainloop_unlock(priv->mainloop);
return samples;
}
// Reset the audio stream, i.e. flush the playback buffer on the server side
static void reset(struct ao *ao)
{
// pa_stream_flush() works badly if not corked
cork(ao, true);
struct priv *priv = ao->priv;
pa_threaded_mainloop_lock(priv->mainloop);
priv->retval = 0;
if (!waitop(priv, pa_stream_flush(priv->stream, success_cb, ao)) ||
!priv->retval)
GENERIC_ERR_MSG("pa_stream_flush() failed");
cork(ao, false);
}
// Pause the audio stream by corking it on the server
static void pause(struct ao *ao)
{
cork(ao, true);
}
// Resume the audio stream by uncorking it on the server
static void resume(struct ao *ao)
{
cork(ao, false);
}
// Return number of samples that may be written to the server without blocking
static int get_space(struct ao *ao)
{
struct priv *priv = ao->priv;
pa_threaded_mainloop_lock(priv->mainloop);
size_t space = pa_stream_writable_size(priv->stream);
pa_threaded_mainloop_unlock(priv->mainloop);
return space / ao->sstride;
}
static double get_delay_hackfixed(struct ao *ao)
{
/* This code basically does what pa_stream_get_latency() _should_
* do, but doesn't due to multiple known bugs in PulseAudio (at
* PulseAudio version 2.1). In particular, the timing interpolation
* mode (PA_STREAM_INTERPOLATE_TIMING) can return completely bogus
* values, and the non-interpolating code has a bug causing too
* large results at end of stream (so a stream never seems to finish).
* This code can still return wrong values in some cases due to known
* PulseAudio bugs that can not be worked around on the client side.
*
* We always query the server for latest timing info. This may take
* too long to work well with remote audio servers, but at least
* this should be enough to fix the normal local playback case.
*/
struct priv *priv = ao->priv;
pa_threaded_mainloop_lock(priv->mainloop);
if (!waitop(priv, pa_stream_update_timing_info(priv->stream, NULL, NULL))) {
GENERIC_ERR_MSG("pa_stream_update_timing_info() failed");
return 0;
}
pa_threaded_mainloop_lock(priv->mainloop);
const pa_timing_info *ti = pa_stream_get_timing_info(priv->stream);
if (!ti) {
pa_threaded_mainloop_unlock(priv->mainloop);
GENERIC_ERR_MSG("pa_stream_get_timing_info() failed");
return 0;
}
const struct pa_sample_spec *ss = pa_stream_get_sample_spec(priv->stream);
if (!ss) {
pa_threaded_mainloop_unlock(priv->mainloop);
GENERIC_ERR_MSG("pa_stream_get_sample_spec() failed");
return 0;
}
// data left in PulseAudio's main buffers (not written to sink yet)
int64_t latency = pa_bytes_to_usec(ti->write_index - ti->read_index, ss);
// since this info may be from a while ago, playback has progressed since
latency -= ti->transport_usec;
// data already moved from buffers to sink, but not played yet
int64_t sink_latency = ti->sink_usec;
if (!ti->playing)
/* At the end of a stream, part of the data "left" in the sink may
* be padding silence after the end; that should be subtracted to
* get the amount of real audio from our stream. This adjustment
* is missing from Pulseaudio's own get_latency calculations
* (as of PulseAudio 2.1). */
sink_latency -= pa_bytes_to_usec(ti->since_underrun, ss);
if (sink_latency > 0)
latency += sink_latency;
if (latency < 0)
latency = 0;
pa_threaded_mainloop_unlock(priv->mainloop);
return latency / 1e6;
}
static double get_delay_pulse(struct ao *ao)
{
struct priv *priv = ao->priv;
pa_usec_t latency = (pa_usec_t) -1;
pa_threaded_mainloop_lock(priv->mainloop);
while (pa_stream_get_latency(priv->stream, &latency, NULL) < 0) {
if (pa_context_errno(priv->context) != PA_ERR_NODATA) {
GENERIC_ERR_MSG("pa_stream_get_latency() failed");
break;
}
/* Wait until latency data is available again */
pa_threaded_mainloop_wait(priv->mainloop);
}
pa_threaded_mainloop_unlock(priv->mainloop);
return latency == (pa_usec_t) -1 ? 0 : latency / 1000000.0;
}
// Return the current latency in seconds
static double get_delay(struct ao *ao)
{
struct priv *priv = ao->priv;
if (priv->cfg_latency_hacks) {
return get_delay_hackfixed(ao);
} else {
return get_delay_pulse(ao);
}
}
/* A callback function that is called when the
* pa_context_get_sink_input_info() operation completes. Saves the
* volume field of the specified structure to the global variable volume.
*/
static void info_func(struct pa_context *c, const struct pa_sink_input_info *i,
int is_last, void *userdata)
{
struct ao *ao = userdata;
struct priv *priv = ao->priv;
if (is_last < 0) {
GENERIC_ERR_MSG("Failed to get sink input info");
return;
}
if (!i)
return;
priv->pi = *i;
pa_threaded_mainloop_signal(priv->mainloop, 0);
}
static int control(struct ao *ao, enum aocontrol cmd, void *arg)
{
struct priv *priv = ao->priv;
switch (cmd) {
case AOCONTROL_GET_MUTE:
case AOCONTROL_GET_VOLUME: {
uint32_t devidx = pa_stream_get_index(priv->stream);
pa_threaded_mainloop_lock(priv->mainloop);
if (!waitop(priv, pa_context_get_sink_input_info(priv->context, devidx,
info_func, ao))) {
GENERIC_ERR_MSG("pa_context_get_sink_input_info() failed");
return CONTROL_ERROR;
}
// Warning: some information in pi might be unaccessible, because
// we naively copied the struct, without updating pointers etc.
// Pointers might point to invalid data, accessors might fail.
if (cmd == AOCONTROL_GET_VOLUME) {
ao_control_vol_t *vol = arg;
if (priv->pi.volume.channels != 2)
vol->left = vol->right =
VOL_PA2MP(pa_cvolume_avg(&priv->pi.volume));
else {
vol->left = VOL_PA2MP(priv->pi.volume.values[0]);
vol->right = VOL_PA2MP(priv->pi.volume.values[1]);
}
} else if (cmd == AOCONTROL_GET_MUTE) {
bool *mute = arg;
*mute = priv->pi.mute;
}
return CONTROL_OK;
}
case AOCONTROL_SET_MUTE:
case AOCONTROL_SET_VOLUME: {
pa_operation *o;
pa_threaded_mainloop_lock(priv->mainloop);
uint32_t stream_index = pa_stream_get_index(priv->stream);
if (cmd == AOCONTROL_SET_VOLUME) {
const ao_control_vol_t *vol = arg;
struct pa_cvolume volume;
pa_cvolume_reset(&volume, ao->channels.num);
if (volume.channels != 2)
pa_cvolume_set(&volume, volume.channels, VOL_MP2PA(vol->left));
else {
volume.values[0] = VOL_MP2PA(vol->left);
volume.values[1] = VOL_MP2PA(vol->right);
}
o = pa_context_set_sink_input_volume(priv->context, stream_index,
&volume, NULL, NULL);
if (!o) {
pa_threaded_mainloop_unlock(priv->mainloop);
GENERIC_ERR_MSG("pa_context_set_sink_input_volume() failed");
return CONTROL_ERROR;
}
} else if (cmd == AOCONTROL_SET_MUTE) {
const bool *mute = arg;
o = pa_context_set_sink_input_mute(priv->context, stream_index,
*mute, NULL, NULL);
if (!o) {
pa_threaded_mainloop_unlock(priv->mainloop);
GENERIC_ERR_MSG("pa_context_set_sink_input_mute() failed");
return CONTROL_ERROR;
}
} else
abort();
/* We don't wait for completion here */
pa_operation_unref(o);
pa_threaded_mainloop_unlock(priv->mainloop);
return CONTROL_OK;
}
case AOCONTROL_HAS_PER_APP_VOLUME:
return CONTROL_TRUE;
case AOCONTROL_UPDATE_STREAM_TITLE: {
char *title = (char *)arg;
pa_threaded_mainloop_lock(priv->mainloop);
if (!waitop(priv, pa_stream_set_name(priv->stream, title,
success_cb, ao)))
{
GENERIC_ERR_MSG("pa_stream_set_name() failed");
return CONTROL_ERROR;
}
return CONTROL_OK;
}
default:
return CONTROL_UNKNOWN;
}
}
struct sink_cb_ctx {
struct ao *ao;
struct ao_device_list *list;
};
static void sink_info_cb(pa_context *c, const pa_sink_info *i, int eol, void *ud)
{
struct sink_cb_ctx *ctx = ud;
struct priv *priv = ctx->ao->priv;
if (eol) {
pa_threaded_mainloop_signal(priv->mainloop, 0); // wakeup waitop()
return;
}
struct ao_device_desc entry = {.name = i->name, .desc = i->description};
ao_device_list_add(ctx->list, ctx->ao, &entry);
}
static int hotplug_init(struct ao *ao)
{
struct priv *priv = ao->priv;
if (pa_init_boilerplate(ao) < 0)
return -1;
pa_threaded_mainloop_lock(priv->mainloop);
waitop(priv, pa_context_subscribe(priv->context, PA_SUBSCRIPTION_MASK_SINK,
context_success_cb, ao));
return 0;
}
static void list_devs(struct ao *ao, struct ao_device_list *list)
{
struct priv *priv = ao->priv;
struct sink_cb_ctx ctx = {ao, list};
pa_threaded_mainloop_lock(priv->mainloop);
waitop(priv, pa_context_get_sink_info_list(priv->context, sink_info_cb, &ctx));
}
static void hotplug_uninit(struct ao *ao)
{
uninit(ao);
}
#define OPT_BASE_STRUCT struct priv
const struct ao_driver audio_out_pulse = {
.description = "PulseAudio audio output",
.name = "pulse",
.control = control,
.init = init,
.uninit = uninit,
.reset = reset,
.get_space = get_space,
.play = play,
.get_delay = get_delay,
.pause = pause,
.resume = resume,
.drain = drain,
.wait = wait_audio,
.wakeup = wakeup,
.hotplug_init = hotplug_init,
.hotplug_uninit = hotplug_uninit,
.list_devs = list_devs,
.priv_size = sizeof(struct priv),
.priv_defaults = &(const struct priv) {
.cfg_buffer = 250,
},
.options = (const struct m_option[]) {
OPT_STRING("host", cfg_host, 0),
OPT_STRING("sink", cfg_sink, 0),
OPT_CHOICE_OR_INT("buffer", cfg_buffer, 0, 1, 2000, ({"native", 0})),
OPT_FLAG("latency-hacks", cfg_latency_hacks, 0),
{0}
},
};
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