mirror of https://github.com/mpv-player/mpv
608 lines
20 KiB
C
608 lines
20 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 MPlayer.
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*
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* MPlayer is free software; you can redistribute it and/or modify
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* it under the terms of the GNU General Public License as published by
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* the Free Software Foundation; either version 2 of the License, or
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* (at your option) any later version.
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*
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* MPlayer 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 General Public License for more details.
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*
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* You should have received a copy of the GNU General Public License along
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* with MPlayer; if not, write to the Free Software Foundation, Inc.,
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* 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
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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 <pulse/pulseaudio.h>
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#include "config.h"
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#include "audio/format.h"
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#include "mpvcore/mp_msg.h"
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#include "ao.h"
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#include "mpvcore/input/input.h"
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#define PULSE_CLIENT_NAME "mpv"
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#define VOL_PA2MP(v) ((v) * 100 / PA_VOLUME_NORM)
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#define VOL_MP2PA(v) ((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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bool broken_pause;
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int retval;
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char *cfg_host;
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char *cfg_sink;
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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 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_READY:
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case PA_STREAM_FAILED:
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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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mp_input_wakeup(ao->input_ctx);
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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 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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/**
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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 int waitop(struct priv *priv, pa_operation *op)
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{
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if (!op) {
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pa_threaded_mainloop_unlock(priv->mainloop);
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return 0;
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}
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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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pa_threaded_mainloop_unlock(priv->mainloop);
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return state == PA_OPERATION_DONE;
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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_S16_LE, PA_SAMPLE_S16LE},
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{AF_FORMAT_S16_BE, PA_SAMPLE_S16BE},
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{AF_FORMAT_S32_LE, PA_SAMPLE_S32LE},
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{AF_FORMAT_S32_BE, PA_SAMPLE_S32BE},
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{AF_FORMAT_FLOAT_LE, PA_SAMPLE_FLOAT32LE},
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{AF_FORMAT_FLOAT_BE, PA_SAMPLE_FLOAT32BE},
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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 const int speaker_map[][2] = {
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{PA_CHANNEL_POSITION_MONO, MP_SPEAKER_ID_FC},
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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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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, bool cut_audio)
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{
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struct priv *priv = ao->priv;
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if (priv->stream && !cut_audio) {
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pa_threaded_mainloop_lock(priv->mainloop);
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waitop(priv, pa_stream_drain(priv->stream, success_cb, ao));
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}
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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 init(struct ao *ao)
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{
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struct pa_sample_spec ss;
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struct pa_channel_map map;
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pa_proplist *proplist = NULL;
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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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char *sink = priv->cfg_sink && priv->cfg_sink[0] ? priv->cfg_sink : NULL;
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const char *version = pa_get_library_version();
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ao->per_application_mixer = true;
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priv->broken_pause = false;
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/* not sure which versions are affected, assume 0.9.11* to 0.9.14*
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* known bad: 0.9.14, 0.9.13
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* known good: 0.9.9, 0.9.10, 0.9.15
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* To test: pause, wait ca. 5 seconds, framestep and see if MPlayer
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* hangs somewhen. */
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if (strncmp(version, "0.9.1", 5) == 0 && version[5] >= '1'
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&& version[5] <= '4') {
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MP_WARN(ao, "working around probably broken pause functionality,\n"
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" see http://www.pulseaudio.org/ticket/440\n");
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priv->broken_pause = true;
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}
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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 (!(priv->context = pa_context_new(pa_threaded_mainloop_get_api(
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priv->mainloop), PULSE_CLIENT_NAME))) {
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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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pa_context_set_state_callback(priv->context, context_state_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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pa_threaded_mainloop_lock(priv->mainloop);
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if (pa_threaded_mainloop_start(priv->mainloop) < 0)
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goto unlock_and_fail;
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/* Wait until the context is ready */
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pa_threaded_mainloop_wait(priv->mainloop);
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if (pa_context_get_state(priv->context) != PA_CONTEXT_READY)
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goto unlock_and_fail;
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ss.channels = ao->channels.num;
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ss.rate = ao->samplerate;
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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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ss.format = fmt_map->pa_format;
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if (!pa_sample_spec_valid(&ss)) {
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MP_ERR(ao, "Invalid sample spec\n");
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goto unlock_and_fail;
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}
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if (!select_chmap(ao, &map))
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goto unlock_and_fail;
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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_ROLE, "video");
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if (!(priv->stream = pa_stream_new_with_proplist(priv->context,
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"audio stream", &ss,
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&map, proplist)))
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goto unlock_and_fail;
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pa_proplist_free(proplist);
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proplist = NULL;
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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_buffer_attr bufattr = {
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.maxlength = -1,
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.tlength = pa_usec_to_bytes(1000000, &ss),
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.prebuf = -1,
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.minreq = -1,
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.fragsize = -1,
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};
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if (pa_stream_connect_playback(priv->stream, sink, &bufattr,
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PA_STREAM_NOT_MONOTONIC, NULL, NULL) < 0)
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goto unlock_and_fail;
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/* Wait until the stream is ready */
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pa_threaded_mainloop_wait(priv->mainloop);
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if (pa_stream_get_state(priv->stream) != PA_STREAM_READY)
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goto unlock_and_fail;
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pa_threaded_mainloop_unlock(priv->mainloop);
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return 0;
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unlock_and_fail:
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if (priv->mainloop)
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pa_threaded_mainloop_unlock(priv->mainloop);
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fail:
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if (priv->context) {
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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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}
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if (proplist)
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pa_proplist_free(proplist);
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uninit(ao, true);
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return -1;
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}
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static void cork(struct ao *ao, bool pause)
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{
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struct priv *priv = ao->priv;
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pa_threaded_mainloop_lock(priv->mainloop);
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priv->retval = 0;
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if (!waitop(priv, pa_stream_cork(priv->stream, pause, success_cb, ao)) ||
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!priv->retval)
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GENERIC_ERR_MSG("pa_stream_cork() failed");
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}
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// Play the specified data to the pulseaudio server
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static int play(struct ao *ao, void *data, int len, int flags)
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{
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struct priv *priv = ao->priv;
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pa_threaded_mainloop_lock(priv->mainloop);
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if (pa_stream_write(priv->stream, data, len, NULL, 0,
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PA_SEEK_RELATIVE) < 0) {
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GENERIC_ERR_MSG("pa_stream_write() failed");
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len = -1;
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}
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if (flags & AOPLAY_FINAL_CHUNK) {
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// Force start in case the stream was too short for prebuf
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pa_operation *op = pa_stream_trigger(priv->stream, NULL, NULL);
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pa_operation_unref(op);
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}
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pa_threaded_mainloop_unlock(priv->mainloop);
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return len;
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}
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// Reset the audio stream, i.e. flush the playback buffer on the server side
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static void reset(struct ao *ao)
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{
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// pa_stream_flush() works badly if not corked
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cork(ao, true);
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struct priv *priv = ao->priv;
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pa_threaded_mainloop_lock(priv->mainloop);
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priv->retval = 0;
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if (!waitop(priv, pa_stream_flush(priv->stream, success_cb, ao)) ||
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!priv->retval)
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GENERIC_ERR_MSG("pa_stream_flush() failed");
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cork(ao, false);
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}
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// Pause the audio stream by corking it on the server
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static void pause(struct ao *ao)
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{
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cork(ao, true);
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}
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// Resume the audio stream by uncorking it on the server
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static void resume(struct ao *ao)
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{
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struct priv *priv = ao->priv;
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/* Without this, certain versions will cause an infinite hang because
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* pa_stream_writable_size returns 0 always.
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* Note that this workaround causes A-V desync after pause. */
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if (priv->broken_pause)
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reset(ao);
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cork(ao, false);
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}
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// Return number of bytes that may be written to the server without blocking
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static int get_space(struct ao *ao)
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{
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struct priv *priv = ao->priv;
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pa_threaded_mainloop_lock(priv->mainloop);
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size_t space = pa_stream_writable_size(priv->stream);
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pa_threaded_mainloop_unlock(priv->mainloop);
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return space;
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}
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// Return the current latency in seconds
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static float get_delay(struct ao *ao)
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{
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/* This code basically does what pa_stream_get_latency() _should_
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* do, but doesn't due to multiple known bugs in PulseAudio (at
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* PulseAudio version 2.1). In particular, the timing interpolation
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* mode (PA_STREAM_INTERPOLATE_TIMING) can return completely bogus
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* values, and the non-interpolating code has a bug causing too
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* large results at end of stream (so a stream never seems to finish).
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* This code can still return wrong values in some cases due to known
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* PulseAudio bugs that can not be worked around on the client side.
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*
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* We always query the server for latest timing info. This may take
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* too long to work well with remote audio servers, but at least
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* this should be enough to fix the normal local playback case.
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*/
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struct priv *priv = ao->priv;
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pa_threaded_mainloop_lock(priv->mainloop);
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if (!waitop(priv, pa_stream_update_timing_info(priv->stream, NULL, NULL))) {
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GENERIC_ERR_MSG("pa_stream_update_timing_info() failed");
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return 0;
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}
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pa_threaded_mainloop_lock(priv->mainloop);
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const pa_timing_info *ti = pa_stream_get_timing_info(priv->stream);
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if (!ti) {
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pa_threaded_mainloop_unlock(priv->mainloop);
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GENERIC_ERR_MSG("pa_stream_get_timing_info() failed");
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return 0;
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}
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const struct pa_sample_spec *ss = pa_stream_get_sample_spec(priv->stream);
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if (!ss) {
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pa_threaded_mainloop_unlock(priv->mainloop);
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GENERIC_ERR_MSG("pa_stream_get_sample_spec() failed");
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return 0;
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}
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// data left in PulseAudio's main buffers (not written to sink yet)
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int64_t latency = pa_bytes_to_usec(ti->write_index - ti->read_index, ss);
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// since this info may be from a while ago, playback has progressed since
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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;
|
|
}
|
|
|
|
/* 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_stream_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;
|
|
}
|
|
default:
|
|
return CONTROL_UNKNOWN;
|
|
}
|
|
}
|
|
|
|
#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,
|
|
.priv_size = sizeof(struct priv),
|
|
.options = (const struct m_option[]) {
|
|
OPT_STRING("host", cfg_host, 0),
|
|
OPT_STRING("sink", cfg_sink, 0),
|
|
{0}
|
|
},
|
|
};
|