mirror of https://github.com/mpv-player/mpv
826 lines
26 KiB
C
826 lines
26 KiB
C
/*
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* PulseAudio audio output driver.
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* Copyright (C) 2006 Lennart Poettering
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* Copyright (C) 2007 Reimar Doeffinger
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*
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* This file is part of mpv.
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*
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* mpv is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Lesser General Public
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* License as published by the Free Software Foundation; either
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* version 2.1 of the License, or (at your option) any later version.
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*
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* mpv is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
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* GNU Lesser General Public License for more details.
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*
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* You should have received a copy of the GNU Lesser General Public
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* License along with mpv. If not, see <http://www.gnu.org/licenses/>.
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*/
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#include <stdlib.h>
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#include <stdbool.h>
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#include <string.h>
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#include <stdint.h>
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#include <math.h>
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#include <pthread.h>
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#include <pulse/pulseaudio.h>
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#include "config.h"
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#include "audio/format.h"
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#include "common/msg.h"
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#include "options/m_option.h"
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#include "ao.h"
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#include "internal.h"
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#define VOL_PA2MP(v) ((v) * 100.0 / PA_VOLUME_NORM)
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#define VOL_MP2PA(v) lrint((v) * PA_VOLUME_NORM / 100)
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struct priv {
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// PulseAudio playback stream object
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struct pa_stream *stream;
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// PulseAudio connection context
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struct pa_context *context;
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// Main event loop object
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struct pa_threaded_mainloop *mainloop;
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// temporary during control()
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struct pa_sink_input_info pi;
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int retval;
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bool playing;
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char *cfg_host;
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int cfg_buffer;
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int cfg_latency_hacks;
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int cfg_allow_suspended;
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};
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#define GENERIC_ERR_MSG(str) \
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MP_ERR(ao, str": %s\n", \
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pa_strerror(pa_context_errno(((struct priv *)ao->priv)->context)))
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static void context_state_cb(pa_context *c, void *userdata)
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{
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struct ao *ao = userdata;
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struct priv *priv = ao->priv;
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switch (pa_context_get_state(c)) {
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case PA_CONTEXT_READY:
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case PA_CONTEXT_TERMINATED:
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case PA_CONTEXT_FAILED:
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pa_threaded_mainloop_signal(priv->mainloop, 0);
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break;
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}
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}
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static void subscribe_cb(pa_context *c, pa_subscription_event_type_t t,
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uint32_t idx, void *userdata)
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{
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struct ao *ao = userdata;
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int type = t & PA_SUBSCRIPTION_MASK_SINK;
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int fac = t & PA_SUBSCRIPTION_EVENT_FACILITY_MASK;
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if ((type == PA_SUBSCRIPTION_EVENT_NEW || type == PA_SUBSCRIPTION_EVENT_REMOVE)
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&& fac == PA_SUBSCRIPTION_EVENT_SINK)
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{
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ao_hotplug_event(ao);
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}
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}
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static void context_success_cb(pa_context *c, int success, void *userdata)
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{
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struct ao *ao = userdata;
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struct priv *priv = ao->priv;
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priv->retval = success;
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pa_threaded_mainloop_signal(priv->mainloop, 0);
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}
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static void stream_state_cb(pa_stream *s, void *userdata)
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{
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struct ao *ao = userdata;
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struct priv *priv = ao->priv;
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switch (pa_stream_get_state(s)) {
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case PA_STREAM_FAILED:
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MP_VERBOSE(ao, "Stream failed.\n");
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ao_request_reload(ao);
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pa_threaded_mainloop_signal(priv->mainloop, 0);
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break;
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case PA_STREAM_READY:
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case PA_STREAM_TERMINATED:
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pa_threaded_mainloop_signal(priv->mainloop, 0);
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break;
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}
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}
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static void stream_request_cb(pa_stream *s, size_t length, void *userdata)
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{
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struct ao *ao = userdata;
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struct priv *priv = ao->priv;
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ao_wakeup_playthread(ao);
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pa_threaded_mainloop_signal(priv->mainloop, 0);
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}
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static void stream_latency_update_cb(pa_stream *s, void *userdata)
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{
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struct ao *ao = userdata;
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struct priv *priv = ao->priv;
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pa_threaded_mainloop_signal(priv->mainloop, 0);
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}
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static void underflow_cb(pa_stream *s, void *userdata)
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{
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struct ao *ao = userdata;
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struct priv *priv = ao->priv;
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priv->playing = false;
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ao_wakeup_playthread(ao);
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pa_threaded_mainloop_signal(priv->mainloop, 0);
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}
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static void success_cb(pa_stream *s, int success, void *userdata)
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{
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struct ao *ao = userdata;
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struct priv *priv = ao->priv;
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priv->retval = success;
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pa_threaded_mainloop_signal(priv->mainloop, 0);
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}
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// Like waitop(), but keep the lock (even if it may unlock temporarily).
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static bool waitop_no_unlock(struct priv *priv, pa_operation *op)
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{
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if (!op)
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return false;
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pa_operation_state_t state = pa_operation_get_state(op);
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while (state == PA_OPERATION_RUNNING) {
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pa_threaded_mainloop_wait(priv->mainloop);
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state = pa_operation_get_state(op);
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}
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pa_operation_unref(op);
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return state == PA_OPERATION_DONE;
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}
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/**
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* \brief waits for a pulseaudio operation to finish, frees it and
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* unlocks the mainloop
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* \param op operation to wait for
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* \return 1 if operation has finished normally (DONE state), 0 otherwise
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*/
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static bool waitop(struct priv *priv, pa_operation *op)
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{
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bool r = waitop_no_unlock(priv, op);
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pa_threaded_mainloop_unlock(priv->mainloop);
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return r;
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}
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static const struct format_map {
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int mp_format;
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pa_sample_format_t pa_format;
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} format_maps[] = {
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{AF_FORMAT_FLOAT, PA_SAMPLE_FLOAT32NE},
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{AF_FORMAT_S32, PA_SAMPLE_S32NE},
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{AF_FORMAT_S16, PA_SAMPLE_S16NE},
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{AF_FORMAT_U8, PA_SAMPLE_U8},
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{AF_FORMAT_UNKNOWN, 0}
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};
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static pa_encoding_t map_digital_format(int format)
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{
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switch (format) {
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case AF_FORMAT_S_AC3: return PA_ENCODING_AC3_IEC61937;
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case AF_FORMAT_S_EAC3: return PA_ENCODING_EAC3_IEC61937;
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case AF_FORMAT_S_MP3: return PA_ENCODING_MPEG_IEC61937;
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case AF_FORMAT_S_DTS: return PA_ENCODING_DTS_IEC61937;
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#ifdef PA_ENCODING_DTSHD_IEC61937
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case AF_FORMAT_S_DTSHD: return PA_ENCODING_DTSHD_IEC61937;
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#endif
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#ifdef PA_ENCODING_MPEG2_AAC_IEC61937
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case AF_FORMAT_S_AAC: return PA_ENCODING_MPEG2_AAC_IEC61937;
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#endif
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#ifdef PA_ENCODING_TRUEHD_IEC61937
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case AF_FORMAT_S_TRUEHD: return PA_ENCODING_TRUEHD_IEC61937;
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#endif
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default:
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if (af_fmt_is_spdif(format))
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return PA_ENCODING_ANY;
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return PA_ENCODING_PCM;
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}
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}
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static const int speaker_map[][2] = {
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{PA_CHANNEL_POSITION_FRONT_LEFT, MP_SPEAKER_ID_FL},
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{PA_CHANNEL_POSITION_FRONT_RIGHT, MP_SPEAKER_ID_FR},
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{PA_CHANNEL_POSITION_FRONT_CENTER, MP_SPEAKER_ID_FC},
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{PA_CHANNEL_POSITION_REAR_CENTER, MP_SPEAKER_ID_BC},
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{PA_CHANNEL_POSITION_REAR_LEFT, MP_SPEAKER_ID_BL},
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{PA_CHANNEL_POSITION_REAR_RIGHT, MP_SPEAKER_ID_BR},
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{PA_CHANNEL_POSITION_LFE, MP_SPEAKER_ID_LFE},
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{PA_CHANNEL_POSITION_FRONT_LEFT_OF_CENTER, MP_SPEAKER_ID_FLC},
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{PA_CHANNEL_POSITION_FRONT_RIGHT_OF_CENTER, MP_SPEAKER_ID_FRC},
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{PA_CHANNEL_POSITION_SIDE_LEFT, MP_SPEAKER_ID_SL},
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{PA_CHANNEL_POSITION_SIDE_RIGHT, MP_SPEAKER_ID_SR},
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{PA_CHANNEL_POSITION_TOP_CENTER, MP_SPEAKER_ID_TC},
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{PA_CHANNEL_POSITION_TOP_FRONT_LEFT, MP_SPEAKER_ID_TFL},
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{PA_CHANNEL_POSITION_TOP_FRONT_RIGHT, MP_SPEAKER_ID_TFR},
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{PA_CHANNEL_POSITION_TOP_FRONT_CENTER, MP_SPEAKER_ID_TFC},
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{PA_CHANNEL_POSITION_TOP_REAR_LEFT, MP_SPEAKER_ID_TBL},
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{PA_CHANNEL_POSITION_TOP_REAR_RIGHT, MP_SPEAKER_ID_TBR},
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{PA_CHANNEL_POSITION_TOP_REAR_CENTER, MP_SPEAKER_ID_TBC},
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{PA_CHANNEL_POSITION_INVALID, -1}
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};
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static bool chmap_pa_from_mp(pa_channel_map *dst, struct mp_chmap *src)
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{
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if (src->num > PA_CHANNELS_MAX)
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return false;
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dst->channels = src->num;
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if (mp_chmap_equals(src, &(const struct mp_chmap)MP_CHMAP_INIT_MONO)) {
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dst->map[0] = PA_CHANNEL_POSITION_MONO;
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return true;
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}
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for (int n = 0; n < src->num; n++) {
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int mp_speaker = src->speaker[n];
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int pa_speaker = PA_CHANNEL_POSITION_INVALID;
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for (int i = 0; speaker_map[i][1] != -1; i++) {
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if (speaker_map[i][1] == mp_speaker) {
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pa_speaker = speaker_map[i][0];
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break;
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}
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}
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if (pa_speaker == PA_CHANNEL_POSITION_INVALID)
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return false;
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dst->map[n] = pa_speaker;
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}
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return true;
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}
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static bool select_chmap(struct ao *ao, pa_channel_map *dst)
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{
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struct mp_chmap_sel sel = {0};
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for (int n = 0; speaker_map[n][1] != -1; n++)
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mp_chmap_sel_add_speaker(&sel, speaker_map[n][1]);
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return ao_chmap_sel_adjust(ao, &sel, &ao->channels) &&
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chmap_pa_from_mp(dst, &ao->channels);
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}
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static void uninit(struct ao *ao)
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{
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struct priv *priv = ao->priv;
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if (priv->mainloop)
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pa_threaded_mainloop_stop(priv->mainloop);
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if (priv->stream) {
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pa_stream_disconnect(priv->stream);
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pa_stream_unref(priv->stream);
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priv->stream = NULL;
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}
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if (priv->context) {
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pa_context_disconnect(priv->context);
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pa_context_unref(priv->context);
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priv->context = NULL;
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}
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if (priv->mainloop) {
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pa_threaded_mainloop_free(priv->mainloop);
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priv->mainloop = NULL;
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}
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}
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static int pa_init_boilerplate(struct ao *ao)
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{
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struct priv *priv = ao->priv;
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char *host = priv->cfg_host && priv->cfg_host[0] ? priv->cfg_host : NULL;
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bool locked = false;
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if (!(priv->mainloop = pa_threaded_mainloop_new())) {
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MP_ERR(ao, "Failed to allocate main loop\n");
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goto fail;
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}
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if (pa_threaded_mainloop_start(priv->mainloop) < 0)
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goto fail;
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pa_threaded_mainloop_lock(priv->mainloop);
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locked = true;
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if (!(priv->context = pa_context_new(pa_threaded_mainloop_get_api(
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priv->mainloop), ao->client_name)))
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{
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MP_ERR(ao, "Failed to allocate context\n");
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goto fail;
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}
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MP_VERBOSE(ao, "Library version: %s\n", pa_get_library_version());
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MP_VERBOSE(ao, "Proto: %lu\n",
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(long)pa_context_get_protocol_version(priv->context));
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MP_VERBOSE(ao, "Server proto: %lu\n",
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(long)pa_context_get_server_protocol_version(priv->context));
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pa_context_set_state_callback(priv->context, context_state_cb, ao);
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pa_context_set_subscribe_callback(priv->context, subscribe_cb, ao);
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if (pa_context_connect(priv->context, host, 0, NULL) < 0)
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goto fail;
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/* Wait until the context is ready */
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while (1) {
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int state = pa_context_get_state(priv->context);
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if (state == PA_CONTEXT_READY)
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break;
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if (!PA_CONTEXT_IS_GOOD(state))
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goto fail;
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pa_threaded_mainloop_wait(priv->mainloop);
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}
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pa_threaded_mainloop_unlock(priv->mainloop);
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return 0;
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fail:
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if (locked)
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pa_threaded_mainloop_unlock(priv->mainloop);
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if (priv->context) {
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pa_threaded_mainloop_lock(priv->mainloop);
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if (!(pa_context_errno(priv->context) == PA_ERR_CONNECTIONREFUSED
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&& ao->probing))
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GENERIC_ERR_MSG("Init failed");
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pa_threaded_mainloop_unlock(priv->mainloop);
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}
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uninit(ao);
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return -1;
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}
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static bool set_format(struct ao *ao, pa_format_info *format)
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{
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ao->format = af_fmt_from_planar(ao->format);
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format->encoding = map_digital_format(ao->format);
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if (format->encoding == PA_ENCODING_PCM) {
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const struct format_map *fmt_map = format_maps;
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while (fmt_map->mp_format != ao->format) {
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if (fmt_map->mp_format == AF_FORMAT_UNKNOWN) {
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MP_VERBOSE(ao, "Unsupported format, using default\n");
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fmt_map = format_maps;
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break;
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}
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fmt_map++;
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}
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ao->format = fmt_map->mp_format;
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pa_format_info_set_sample_format(format, fmt_map->pa_format);
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}
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struct pa_channel_map map;
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if (!select_chmap(ao, &map))
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return false;
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pa_format_info_set_rate(format, ao->samplerate);
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pa_format_info_set_channels(format, ao->channels.num);
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pa_format_info_set_channel_map(format, &map);
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return ao->samplerate < PA_RATE_MAX && pa_format_info_valid(format);
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}
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static int init(struct ao *ao)
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{
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pa_proplist *proplist = NULL;
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pa_format_info *format = NULL;
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struct priv *priv = ao->priv;
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char *sink = ao->device;
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|
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if (pa_init_boilerplate(ao) < 0)
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return -1;
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pa_threaded_mainloop_lock(priv->mainloop);
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if (!(proplist = pa_proplist_new())) {
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MP_ERR(ao, "Failed to allocate proplist\n");
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goto unlock_and_fail;
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}
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(void)pa_proplist_sets(proplist, PA_PROP_MEDIA_ICON_NAME, ao->client_name);
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if (!(format = pa_format_info_new()))
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goto unlock_and_fail;
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if (!set_format(ao, format)) {
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ao->channels = (struct mp_chmap) MP_CHMAP_INIT_STEREO;
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ao->samplerate = 48000;
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ao->format = AF_FORMAT_FLOAT;
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if (!set_format(ao, format)) {
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MP_ERR(ao, "Invalid audio format\n");
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goto unlock_and_fail;
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}
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}
|
|
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if (!(priv->stream = pa_stream_new_extended(priv->context, "audio stream",
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&format, 1, proplist)))
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goto unlock_and_fail;
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pa_format_info_free(format);
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format = NULL;
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|
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pa_proplist_free(proplist);
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proplist = NULL;
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|
|
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pa_stream_set_state_callback(priv->stream, stream_state_cb, ao);
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pa_stream_set_write_callback(priv->stream, stream_request_cb, ao);
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pa_stream_set_latency_update_callback(priv->stream,
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stream_latency_update_cb, ao);
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pa_stream_set_underflow_callback(priv->stream, underflow_cb, ao);
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uint32_t buf_size = ao->samplerate * (priv->cfg_buffer / 1000.0) *
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af_fmt_to_bytes(ao->format) * ao->channels.num;
|
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pa_buffer_attr bufattr = {
|
|
.maxlength = -1,
|
|
.tlength = buf_size > 0 ? buf_size : -1,
|
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.prebuf = 0,
|
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.minreq = -1,
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.fragsize = -1,
|
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};
|
|
|
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int flags = PA_STREAM_NOT_MONOTONIC;
|
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if (!priv->cfg_latency_hacks)
|
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flags |= PA_STREAM_INTERPOLATE_TIMING|PA_STREAM_AUTO_TIMING_UPDATE;
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|
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if (pa_stream_connect_playback(priv->stream, sink, &bufattr,
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flags, NULL, NULL) < 0)
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goto unlock_and_fail;
|
|
|
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/* Wait until the stream is ready */
|
|
while (1) {
|
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int state = pa_stream_get_state(priv->stream);
|
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if (state == PA_STREAM_READY)
|
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break;
|
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if (!PA_STREAM_IS_GOOD(state))
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goto unlock_and_fail;
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pa_threaded_mainloop_wait(priv->mainloop);
|
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}
|
|
|
|
if (pa_stream_is_suspended(priv->stream) && !priv->cfg_allow_suspended) {
|
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MP_ERR(ao, "The stream is suspended. Bailing out.\n");
|
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goto unlock_and_fail;
|
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}
|
|
|
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const pa_buffer_attr* final_bufattr = pa_stream_get_buffer_attr(priv->stream);
|
|
if(!final_bufattr) {
|
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MP_ERR(ao, "PulseAudio didn't tell us what buffer sizes it set. Bailing out.\n");
|
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goto unlock_and_fail;
|
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}
|
|
ao->device_buffer = final_bufattr->tlength /
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af_fmt_to_bytes(ao->format) / ao->channels.num;
|
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|
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pa_threaded_mainloop_unlock(priv->mainloop);
|
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return 0;
|
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|
|
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_no_unlock(priv, pa_stream_cork(priv->stream, pause, success_cb, ao))
|
|
&& priv->retval)
|
|
{
|
|
priv->playing = true;
|
|
} else {
|
|
GENERIC_ERR_MSG("pa_stream_cork() failed");
|
|
priv->playing = false;
|
|
}
|
|
pa_threaded_mainloop_unlock(priv->mainloop);
|
|
}
|
|
|
|
// Play the specified data to the pulseaudio server
|
|
static bool audio_write(struct ao *ao, void **data, int samples)
|
|
{
|
|
struct priv *priv = ao->priv;
|
|
bool res = true;
|
|
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");
|
|
res = false;
|
|
}
|
|
pa_threaded_mainloop_unlock(priv->mainloop);
|
|
return res;
|
|
}
|
|
|
|
static void start(struct ao *ao)
|
|
{
|
|
cork(ao, false);
|
|
}
|
|
|
|
// 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");
|
|
}
|
|
|
|
static bool set_pause(struct ao *ao, bool paused)
|
|
{
|
|
cork(ao, paused);
|
|
return true;
|
|
}
|
|
|
|
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;
|
|
if (!waitop_no_unlock(priv, pa_stream_update_timing_info(priv->stream,
|
|
NULL, NULL)))
|
|
{
|
|
GENERIC_ERR_MSG("pa_stream_update_timing_info() failed");
|
|
return 0;
|
|
}
|
|
const pa_timing_info *ti = pa_stream_get_timing_info(priv->stream);
|
|
if (!ti) {
|
|
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) {
|
|
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;
|
|
return latency / 1e6;
|
|
}
|
|
|
|
static double get_delay_pulse(struct ao *ao)
|
|
{
|
|
struct priv *priv = ao->priv;
|
|
pa_usec_t latency = (pa_usec_t) -1;
|
|
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);
|
|
}
|
|
return latency == (pa_usec_t) -1 ? 0 : latency / 1000000.0;
|
|
}
|
|
|
|
static void audio_get_state(struct ao *ao, struct mp_pcm_state *state)
|
|
{
|
|
struct priv *priv = ao->priv;
|
|
|
|
pa_threaded_mainloop_lock(priv->mainloop);
|
|
|
|
size_t space = pa_stream_writable_size(priv->stream);
|
|
state->free_samples = space == (size_t)-1 ? 0 : space / ao->sstride;
|
|
|
|
state->queued_samples = ao->device_buffer - state->free_samples; // dunno
|
|
|
|
if (priv->cfg_latency_hacks) {
|
|
state->delay = get_delay_hackfixed(ao);
|
|
} else {
|
|
state->delay = get_delay_pulse(ao);
|
|
}
|
|
|
|
state->playing = priv->playing;
|
|
|
|
pa_threaded_mainloop_unlock(priv->mainloop);
|
|
|
|
// Otherwise, PA will keep hammering us for underruns (which it does instead
|
|
// of stopping the stream automatically).
|
|
if (!state->playing)
|
|
cork(ao, true);
|
|
}
|
|
|
|
/* 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_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_state = audio_get_state,
|
|
.write = audio_write,
|
|
.start = start,
|
|
.set_pause = set_pause,
|
|
.hotplug_init = hotplug_init,
|
|
.hotplug_uninit = hotplug_uninit,
|
|
.list_devs = list_devs,
|
|
.priv_size = sizeof(struct priv),
|
|
.priv_defaults = &(const struct priv) {
|
|
.cfg_buffer = 100,
|
|
},
|
|
.options = (const struct m_option[]) {
|
|
{"host", OPT_STRING(cfg_host)},
|
|
{"buffer", OPT_CHOICE(cfg_buffer, {"native", 0}),
|
|
M_RANGE(1, 2000)},
|
|
{"latency-hacks", OPT_FLAG(cfg_latency_hacks)},
|
|
{"allow-suspended", OPT_FLAG(cfg_allow_suspended)},
|
|
{0}
|
|
},
|
|
.options_prefix = "pulse",
|
|
};
|