mirror of
https://github.com/mpv-player/mpv
synced 2024-11-18 21:16:10 +01:00
41aefce730
Switch the internal channel order to libavcodec's. If the channel number mismatches at some point, use libavresample for up- or downmixing. Remove the old af_pan automatic downmixing. The libavcodec channel order should be equivalent to WAVEFORMATEX order, at least nowadays. reorder_ch.h assumes that WAVEFORMATEX and libavcodec might be different, but all defined channels have the same mappings. Remove the downmixing with af_pan as well as the channel conversion with af_channels from af.c, and prefer af_lavrresample for this. The automatic downmixing behavior should be the same as before (if the --channels option is set to 2, which is the default, the audio output is forced to 2 channels, and libavresample does all downmixing). Note that mpv still can't do channel layouts. It will pick the default channel layout according to the channel count. This will be fixed later by passing down the channel layout as well. af_hrtf depends on the order of the input channels, so reorder to ALSA (for which this code was written). This is better than changing the filter code, which is more risky. ao_pulse can accept waveext order directly, so set that as channel mapping.
553 lines
18 KiB
C
553 lines
18 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 "core/mp_msg.h"
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#include "ao.h"
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#include "core/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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};
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#define GENERIC_ERR_MSG(ctx, str) \
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mp_msg(MSGT_AO, MSGL_ERR, "AO: [pulse] "str": %s\n", \
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pa_strerror(pa_context_errno(ctx)))
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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 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, char *params)
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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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char *devarg = NULL;
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char *host = NULL;
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char *sink = NULL;
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const char *version = pa_get_library_version();
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struct priv *priv = talloc_zero(ao, struct priv);
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ao->priv = priv;
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ao->per_application_mixer = true;
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if (params) {
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devarg = strdup(params);
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sink = strchr(devarg, ':');
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if (sink)
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*sink++ = 0;
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if (devarg[0])
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host = devarg;
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}
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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_msg(MSGT_AO, MSGL_WARN,
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"[pulse] 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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ss.channels = ao->channels;
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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_msg(MSGT_AO, MSGL_V,
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"AO: [pulse] 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_msg(MSGT_AO, MSGL_ERR, "AO: [pulse] Invalid sample spec\n");
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goto fail;
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}
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pa_channel_map_init_auto(&map, ss.channels, PA_CHANNEL_MAP_WAVEEX);
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ao->bps = pa_bytes_per_second(&ss);
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if (!(priv->mainloop = pa_threaded_mainloop_new())) {
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mp_msg(MSGT_AO, MSGL_ERR, "AO: [pulse] 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_msg(MSGT_AO, MSGL_ERR, "AO: [pulse] 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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if (!(priv->stream = pa_stream_new(priv->context, "audio stream", &ss,
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&map)))
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goto unlock_and_fail;
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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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free(devarg);
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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(priv->context, "Init failed");
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}
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free(devarg);
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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(priv->context, "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(priv->context, "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(priv->context, "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(priv->context, "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(priv->context, "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(priv->context, "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;
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// data already moved from buffers to sink, but not played yet
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int64_t sink_latency = ti->sink_usec;
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if (!ti->playing)
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/* At the end of a stream, part of the data "left" in the sink may
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* be padding silence after the end; that should be subtracted to
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* get the amount of real audio from our stream. This adjustment
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* is missing from Pulseaudio's own get_latency calculations
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* (as of PulseAudio 2.1). */
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sink_latency -= pa_bytes_to_usec(ti->since_underrun, ss);
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if (sink_latency > 0)
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latency += sink_latency;
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if (latency < 0)
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latency = 0;
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pa_threaded_mainloop_unlock(priv->mainloop);
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return latency / 1e6;
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}
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/* A callback function that is called when the
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* pa_context_get_sink_input_info() operation completes. Saves the
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* volume field of the specified structure to the global variable volume.
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*/
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static void info_func(struct pa_context *c, const struct pa_sink_input_info *i,
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int is_last, 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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if (is_last < 0) {
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GENERIC_ERR_MSG(priv->context, "Failed to get sink input info");
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return;
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}
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if (!i)
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return;
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priv->pi = *i;
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pa_threaded_mainloop_signal(priv->mainloop, 0);
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}
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static int control(struct ao *ao, enum aocontrol cmd, void *arg)
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{
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struct priv *priv = ao->priv;
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switch (cmd) {
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case AOCONTROL_GET_MUTE:
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case AOCONTROL_GET_VOLUME: {
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uint32_t devidx = pa_stream_get_index(priv->stream);
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pa_threaded_mainloop_lock(priv->mainloop);
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if (!waitop(priv, pa_context_get_sink_input_info(priv->context, devidx,
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info_func, ao))) {
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GENERIC_ERR_MSG(priv->context,
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"pa_stream_get_sink_input_info() failed");
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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);
|
|
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(priv->context,
|
|
"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(priv->context,
|
|
"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;
|
|
}
|
|
}
|
|
|
|
const struct ao_driver audio_out_pulse = {
|
|
.is_new = true,
|
|
.info = &(const struct ao_info) {
|
|
"PulseAudio audio output",
|
|
"pulse",
|
|
"Lennart Poettering",
|
|
"",
|
|
},
|
|
.control = control,
|
|
.init = init,
|
|
.uninit = uninit,
|
|
.reset = reset,
|
|
.get_space = get_space,
|
|
.play = play,
|
|
.get_delay = get_delay,
|
|
.pause = pause,
|
|
.resume = resume,
|
|
};
|