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d0b129971a
mpv/audio/out/ao_pulse.c
wm4 f88193091b audio/out: don't require AOs to set ao->bps 
Some still do, because they use the value in other places of the init
function. ao_portaudio is tricky and reads ao->bps in the stream
thread, which might be started on initialization (not sure about that,
but better safe than sorry).
2013-06-16 19:32:18 +02:00

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/*
* PulseAudio audio output driver.
* Copyright (C) 2006 Lennart Poettering
* Copyright (C) 2007 Reimar Doeffinger
*
* This file is part of MPlayer.
*
* MPlayer is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License as published by
* the Free Software Foundation; either version 2 of the License, or
* (at your option) any later version.
*
* MPlayer is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
* GNU General Public License for more details.
*
* You should have received a copy of the GNU General Public License along
* with MPlayer; if not, write to the Free Software Foundation, Inc.,
* 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
*/
#include <stdlib.h>
#include <stdbool.h>
#include <string.h>
#include <pulse/pulseaudio.h>
#include "config.h"
#include "audio/format.h"
#include "core/mp_msg.h"
#include "ao.h"
#include "core/input/input.h"
#define PULSE_CLIENT_NAME "mpv"
#define VOL_PA2MP(v) ((v) * 100 / PA_VOLUME_NORM)
#define VOL_MP2PA(v) ((v) * PA_VOLUME_NORM / 100)
struct priv {
// PulseAudio playback stream object
struct pa_stream *stream;
// PulseAudio connection context
struct pa_context *context;
// Main event loop object
struct pa_threaded_mainloop *mainloop;
// temporary during control()
struct pa_sink_input_info pi;
bool broken_pause;
int retval;
};
#define GENERIC_ERR_MSG(ctx, str) \
mp_msg(MSGT_AO, MSGL_ERR, "AO: [pulse] "str": %s\n", \
pa_strerror(pa_context_errno(ctx)))
static void context_state_cb(pa_context *c, void *userdata)
{
struct ao *ao = userdata;
struct priv *priv = ao->priv;
switch (pa_context_get_state(c)) {
case PA_CONTEXT_READY:
case PA_CONTEXT_TERMINATED:
case PA_CONTEXT_FAILED:
pa_threaded_mainloop_signal(priv->mainloop, 0);
break;
}
}
static void stream_state_cb(pa_stream *s, void *userdata)
{
struct ao *ao = userdata;
struct priv *priv = ao->priv;
switch (pa_stream_get_state(s)) {
case PA_STREAM_READY:
case PA_STREAM_FAILED:
case PA_STREAM_TERMINATED:
pa_threaded_mainloop_signal(priv->mainloop, 0);
break;
}
}
static void stream_request_cb(pa_stream *s, size_t length, void *userdata)
{
struct ao *ao = userdata;
struct priv *priv = ao->priv;
mp_input_wakeup(ao->input_ctx);
pa_threaded_mainloop_signal(priv->mainloop, 0);
}
static void stream_latency_update_cb(pa_stream *s, void *userdata)
{
struct ao *ao = userdata;
struct priv *priv = ao->priv;
pa_threaded_mainloop_signal(priv->mainloop, 0);
}
static void success_cb(pa_stream *s, int success, void *userdata)
{
struct ao *ao = userdata;
struct priv *priv = ao->priv;
priv->retval = success;
pa_threaded_mainloop_signal(priv->mainloop, 0);
}
/**
* \brief waits for a pulseaudio operation to finish, frees it and
* unlocks the mainloop
* \param op operation to wait for
* \return 1 if operation has finished normally (DONE state), 0 otherwise
*/
static int waitop(struct priv *priv, pa_operation *op)
{
if (!op) {
pa_threaded_mainloop_unlock(priv->mainloop);
return 0;
}
pa_operation_state_t state = pa_operation_get_state(op);
while (state == PA_OPERATION_RUNNING) {
pa_threaded_mainloop_wait(priv->mainloop);
state = pa_operation_get_state(op);
}
pa_operation_unref(op);
pa_threaded_mainloop_unlock(priv->mainloop);
return state == PA_OPERATION_DONE;
}
static const struct format_map {
int mp_format;
pa_sample_format_t pa_format;
} format_maps[] = {
{AF_FORMAT_S16_LE, PA_SAMPLE_S16LE},
{AF_FORMAT_S16_BE, PA_SAMPLE_S16BE},
{AF_FORMAT_S32_LE, PA_SAMPLE_S32LE},
{AF_FORMAT_S32_BE, PA_SAMPLE_S32BE},
{AF_FORMAT_FLOAT_LE, PA_SAMPLE_FLOAT32LE},
{AF_FORMAT_FLOAT_BE, PA_SAMPLE_FLOAT32BE},
{AF_FORMAT_U8, PA_SAMPLE_U8},
{AF_FORMAT_UNKNOWN, 0}
};
static const int speaker_map[][2] = {
{PA_CHANNEL_POSITION_MONO, MP_SPEAKER_ID_FC},
{PA_CHANNEL_POSITION_FRONT_LEFT, MP_SPEAKER_ID_FL},
{PA_CHANNEL_POSITION_FRONT_RIGHT, MP_SPEAKER_ID_FR},
{PA_CHANNEL_POSITION_FRONT_CENTER, MP_SPEAKER_ID_FC},
{PA_CHANNEL_POSITION_REAR_CENTER, MP_SPEAKER_ID_BC},
{PA_CHANNEL_POSITION_REAR_LEFT, MP_SPEAKER_ID_BL},
{PA_CHANNEL_POSITION_REAR_RIGHT, MP_SPEAKER_ID_BR},
{PA_CHANNEL_POSITION_LFE, MP_SPEAKER_ID_LFE},
{PA_CHANNEL_POSITION_FRONT_LEFT_OF_CENTER, MP_SPEAKER_ID_FLC},
{PA_CHANNEL_POSITION_FRONT_RIGHT_OF_CENTER, MP_SPEAKER_ID_FRC},
{PA_CHANNEL_POSITION_SIDE_LEFT, MP_SPEAKER_ID_SL},
{PA_CHANNEL_POSITION_SIDE_RIGHT, MP_SPEAKER_ID_SR},
{PA_CHANNEL_POSITION_TOP_CENTER, MP_SPEAKER_ID_TC},
{PA_CHANNEL_POSITION_TOP_FRONT_LEFT, MP_SPEAKER_ID_TFL},
{PA_CHANNEL_POSITION_TOP_FRONT_RIGHT, MP_SPEAKER_ID_TFR},
{PA_CHANNEL_POSITION_TOP_FRONT_CENTER, MP_SPEAKER_ID_TFC},
{PA_CHANNEL_POSITION_TOP_REAR_LEFT, MP_SPEAKER_ID_TBL},
{PA_CHANNEL_POSITION_TOP_REAR_RIGHT, MP_SPEAKER_ID_TBR},
{PA_CHANNEL_POSITION_TOP_REAR_CENTER, MP_SPEAKER_ID_TBC},
{PA_CHANNEL_POSITION_INVALID, -1}
};
static bool chmap_pa_from_mp(pa_channel_map *dst, struct mp_chmap *src)
{
if (src->num > PA_CHANNELS_MAX)
return false;
dst->channels = src->num;
for (int n = 0; n < src->num; n++) {
int mp_speaker = src->speaker[n];
int pa_speaker = PA_CHANNEL_POSITION_INVALID;
for (int i = 0; speaker_map[i][1] != -1; i++) {
if (speaker_map[i][1] == mp_speaker) {
pa_speaker = speaker_map[i][0];
break;
}
}
if (pa_speaker == PA_CHANNEL_POSITION_INVALID)
return false;
dst->map[n] = pa_speaker;
}
return true;
}
static bool select_chmap(struct ao *ao, pa_channel_map *dst)
{
struct mp_chmap_sel sel = {0};
for (int n = 0; speaker_map[n][1] != -1; n++)
mp_chmap_sel_add_speaker(&sel, speaker_map[n][1]);
return ao_chmap_sel_adjust(ao, &sel, &ao->channels) &&
chmap_pa_from_mp(dst, &ao->channels);
}
static void uninit(struct ao *ao, bool cut_audio)
{
struct priv *priv = ao->priv;
if (priv->stream && !cut_audio) {
pa_threaded_mainloop_lock(priv->mainloop);
waitop(priv, pa_stream_drain(priv->stream, success_cb, ao));
}
if (priv->mainloop)
pa_threaded_mainloop_stop(priv->mainloop);
if (priv->stream) {
pa_stream_disconnect(priv->stream);
pa_stream_unref(priv->stream);
priv->stream = NULL;
}
if (priv->context) {
pa_context_disconnect(priv->context);
pa_context_unref(priv->context);
priv->context = NULL;
}
if (priv->mainloop) {
pa_threaded_mainloop_free(priv->mainloop);
priv->mainloop = NULL;
}
}
static int init(struct ao *ao, char *params)
{
struct pa_sample_spec ss;
struct pa_channel_map map;
char *devarg = NULL;
char *host = NULL;
char *sink = NULL;
const char *version = pa_get_library_version();
struct priv *priv = talloc_zero(ao, struct priv);
ao->priv = priv;
ao->per_application_mixer = true;
if (params) {
devarg = strdup(params);
sink = strchr(devarg, ':');
if (sink)
*sink++ = 0;
if (devarg[0])
host = devarg;
}
priv->broken_pause = false;
/* not sure which versions are affected, assume 0.9.11* to 0.9.14*
* known bad: 0.9.14, 0.9.13
* known good: 0.9.9, 0.9.10, 0.9.15
* To test: pause, wait ca. 5 seconds, framestep and see if MPlayer
* hangs somewhen. */
if (strncmp(version, "0.9.1", 5) == 0 && version[5] >= '1'
&& version[5] <= '4') {
mp_msg(MSGT_AO, MSGL_WARN,
"[pulse] working around probably broken pause functionality,\n"
" see http://www.pulseaudio.org/ticket/440\n");
priv->broken_pause = true;
}
if (!(priv->mainloop = pa_threaded_mainloop_new())) {
mp_msg(MSGT_AO, MSGL_ERR, "AO: [pulse] Failed to allocate main loop\n");
goto fail;
}
if (!(priv->context = pa_context_new(pa_threaded_mainloop_get_api(
priv->mainloop), PULSE_CLIENT_NAME))) {
mp_msg(MSGT_AO, MSGL_ERR, "AO: [pulse] Failed to allocate context\n");
goto fail;
}
pa_context_set_state_callback(priv->context, context_state_cb, ao);
if (pa_context_connect(priv->context, host, 0, NULL) < 0)
goto fail;
pa_threaded_mainloop_lock(priv->mainloop);
if (pa_threaded_mainloop_start(priv->mainloop) < 0)
goto unlock_and_fail;
/* Wait until the context is ready */
pa_threaded_mainloop_wait(priv->mainloop);
if (pa_context_get_state(priv->context) != PA_CONTEXT_READY)
goto unlock_and_fail;
ss.channels = ao->channels.num;
ss.rate = ao->samplerate;
const struct format_map *fmt_map = format_maps;
while (fmt_map->mp_format != ao->format) {
if (fmt_map->mp_format == AF_FORMAT_UNKNOWN) {
mp_msg(MSGT_AO, MSGL_V,
"AO: [pulse] Unsupported format, using default\n");
fmt_map = format_maps;
break;
}
fmt_map++;
}
ao->format = fmt_map->mp_format;
ss.format = fmt_map->pa_format;
if (!pa_sample_spec_valid(&ss)) {
mp_msg(MSGT_AO, MSGL_ERR, "AO: [pulse] Invalid sample spec\n");
goto fail;
}
if (!select_chmap(ao, &map))
goto fail;
if (!(priv->stream = pa_stream_new(priv->context, "audio stream", &ss,
&map)))
goto unlock_and_fail;
pa_stream_set_state_callback(priv->stream, stream_state_cb, ao);
pa_stream_set_write_callback(priv->stream, stream_request_cb, ao);
pa_stream_set_latency_update_callback(priv->stream,
stream_latency_update_cb, ao);
pa_buffer_attr bufattr = {
.maxlength = -1,
.tlength = pa_usec_to_bytes(1000000, &ss),
.prebuf = -1,
.minreq = -1,
.fragsize = -1,
};
if (pa_stream_connect_playback(priv->stream, sink, &bufattr,
PA_STREAM_NOT_MONOTONIC, NULL, NULL) < 0)
goto unlock_and_fail;
/* Wait until the stream is ready */
pa_threaded_mainloop_wait(priv->mainloop);
if (pa_stream_get_state(priv->stream) != PA_STREAM_READY)
goto unlock_and_fail;
pa_threaded_mainloop_unlock(priv->mainloop);
free(devarg);
return 0;
unlock_and_fail:
if (priv->mainloop)
pa_threaded_mainloop_unlock(priv->mainloop);
fail:
if (priv->context) {
if (!(pa_context_errno(priv->context) == PA_ERR_CONNECTIONREFUSED
&& ao->probing))
GENERIC_ERR_MSG(priv->context, "Init failed");
}
free(devarg);
uninit(ao, true);
return -1;
}
static void cork(struct ao *ao, bool pause)
{
struct priv *priv = ao->priv;
pa_threaded_mainloop_lock(priv->mainloop);
priv->retval = 0;
if (!waitop(priv, pa_stream_cork(priv->stream, pause, success_cb, ao)) ||
!priv->retval)
GENERIC_ERR_MSG(priv->context, "pa_stream_cork() failed");
}
// Play the specified data to the pulseaudio server
static int play(struct ao *ao, void *data, int len, int flags)
{
struct priv *priv = ao->priv;
pa_threaded_mainloop_lock(priv->mainloop);
if (pa_stream_write(priv->stream, data, len, NULL, 0,
PA_SEEK_RELATIVE) < 0) {
GENERIC_ERR_MSG(priv->context, "pa_stream_write() failed");
len = -1;
}
if (flags & AOPLAY_FINAL_CHUNK) {
// Force start in case the stream was too short for prebuf
pa_operation *op = pa_stream_trigger(priv->stream, NULL, NULL);
pa_operation_unref(op);
}
pa_threaded_mainloop_unlock(priv->mainloop);
return len;
}
// 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(priv->context, "pa_stream_flush() failed");
cork(ao, false);
}
// Pause the audio stream by corking it on the server
static void pause(struct ao *ao)
{
cork(ao, true);
}
// Resume the audio stream by uncorking it on the server
static void resume(struct ao *ao)
{
struct priv *priv = ao->priv;
/* Without this, certain versions will cause an infinite hang because
* pa_stream_writable_size returns 0 always.
* Note that this workaround causes A-V desync after pause. */
if (priv->broken_pause)
reset(ao);
cork(ao, false);
}
// Return number of bytes that may be written to the server without blocking
static int get_space(struct ao *ao)
{
struct priv *priv = ao->priv;
pa_threaded_mainloop_lock(priv->mainloop);
size_t space = pa_stream_writable_size(priv->stream);
pa_threaded_mainloop_unlock(priv->mainloop);
return space;
}
// Return the current latency in seconds
static float get_delay(struct ao *ao)
{
/* This code basically does what pa_stream_get_latency() _should_
* do, but doesn't due to multiple known bugs in PulseAudio (at
* PulseAudio version 2.1). In particular, the timing interpolation
* mode (PA_STREAM_INTERPOLATE_TIMING) can return completely bogus
* values, and the non-interpolating code has a bug causing too
* large results at end of stream (so a stream never seems to finish).
* This code can still return wrong values in some cases due to known
* PulseAudio bugs that can not be worked around on the client side.
*
* We always query the server for latest timing info. This may take
* too long to work well with remote audio servers, but at least
* this should be enough to fix the normal local playback case.
*/
struct priv *priv = ao->priv;
pa_threaded_mainloop_lock(priv->mainloop);
if (!waitop(priv, pa_stream_update_timing_info(priv->stream, NULL, NULL))) {
GENERIC_ERR_MSG(priv->context, "pa_stream_update_timing_info() failed");
return 0;
}
pa_threaded_mainloop_lock(priv->mainloop);
const pa_timing_info *ti = pa_stream_get_timing_info(priv->stream);
if (!ti) {
pa_threaded_mainloop_unlock(priv->mainloop);
GENERIC_ERR_MSG(priv->context, "pa_stream_get_timing_info() failed");
return 0;
}
const struct pa_sample_spec *ss = pa_stream_get_sample_spec(priv->stream);
if (!ss) {
pa_threaded_mainloop_unlock(priv->mainloop);
GENERIC_ERR_MSG(priv->context, "pa_stream_get_sample_spec() failed");
return 0;
}
// data left in PulseAudio's main buffers (not written to sink yet)
int64_t latency = pa_bytes_to_usec(ti->write_index - ti->read_index, ss);
// since this info may be from a while ago, playback has progressed since
latency -= ti->transport_usec;
// data already moved from buffers to sink, but not played yet
int64_t sink_latency = ti->sink_usec;
if (!ti->playing)
/* At the end of a stream, part of the data "left" in the sink may
* be padding silence after the end; that should be subtracted to
* get the amount of real audio from our stream. This adjustment
* is missing from Pulseaudio's own get_latency calculations
* (as of PulseAudio 2.1). */
sink_latency -= pa_bytes_to_usec(ti->since_underrun, ss);
if (sink_latency > 0)
latency += sink_latency;
if (latency < 0)
latency = 0;
pa_threaded_mainloop_unlock(priv->mainloop);
return latency / 1e6;
}
/* 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(priv->context, "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(priv->context,
"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(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 = {
.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,
};
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