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mpv/audio/out/ao_pulse.c
wm4 380fc765e4 audio/out: prepare for non-interleaved audio 
This comes with two internal AO API changes:

1. ao_driver.play now can take non-interleaved audio. For this purpose,
the data pointer is changed to void **data, where data[0] corresponds to
the pointer in the old API. Also, the len argument as well as the return
value are now in samples, not bytes. "Sample" in this context means the
unit of the smallest possible audio frame, i.e. sample_size * channels.

2. ao_driver.get_space now returns samples instead of bytes. (Similar to
the play function.)

Change all AOs to use the new API.

The AO API as exposed to the rest of the player still uses the old API.
It's emulated in ao.c. This is purely to split the commits changing all
AOs and the commits adding actual support for outputting N-I audio.
2013-11-12 23:27:51 +01:00

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/*
* PulseAudio audio output driver.
* Copyright (C) 2006 Lennart Poettering
* Copyright (C) 2007 Reimar Doeffinger
*
* This file is part of 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 "mpvcore/mp_msg.h"
#include "ao.h"
#include "mpvcore/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;
char *cfg_host;
char *cfg_sink;
};
#define GENERIC_ERR_MSG(str) \
MP_ERR(ao, str": %s\n", \
pa_strerror(pa_context_errno(((struct priv *)ao->priv)->context)))
static void context_state_cb(pa_context *c, void *userdata)
{
struct ao *ao = userdata;
struct priv *priv = ao->priv;
switch (pa_context_get_state(c)) {
case PA_CONTEXT_READY:
case PA_CONTEXT_TERMINATED:
case PA_CONTEXT_FAILED:
pa_threaded_mainloop_signal(priv->mainloop, 0);
break;
}
}
static void 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_FRONT_LEFT, MP_SPEAKER_ID_FL},
{PA_CHANNEL_POSITION_FRONT_RIGHT, MP_SPEAKER_ID_FR},
{PA_CHANNEL_POSITION_FRONT_CENTER, MP_SPEAKER_ID_FC},
{PA_CHANNEL_POSITION_REAR_CENTER, MP_SPEAKER_ID_BC},
{PA_CHANNEL_POSITION_REAR_LEFT, MP_SPEAKER_ID_BL},
{PA_CHANNEL_POSITION_REAR_RIGHT, MP_SPEAKER_ID_BR},
{PA_CHANNEL_POSITION_LFE, MP_SPEAKER_ID_LFE},
{PA_CHANNEL_POSITION_FRONT_LEFT_OF_CENTER, MP_SPEAKER_ID_FLC},
{PA_CHANNEL_POSITION_FRONT_RIGHT_OF_CENTER, MP_SPEAKER_ID_FRC},
{PA_CHANNEL_POSITION_SIDE_LEFT, MP_SPEAKER_ID_SL},
{PA_CHANNEL_POSITION_SIDE_RIGHT, MP_SPEAKER_ID_SR},
{PA_CHANNEL_POSITION_TOP_CENTER, MP_SPEAKER_ID_TC},
{PA_CHANNEL_POSITION_TOP_FRONT_LEFT, MP_SPEAKER_ID_TFL},
{PA_CHANNEL_POSITION_TOP_FRONT_RIGHT, MP_SPEAKER_ID_TFR},
{PA_CHANNEL_POSITION_TOP_FRONT_CENTER, MP_SPEAKER_ID_TFC},
{PA_CHANNEL_POSITION_TOP_REAR_LEFT, MP_SPEAKER_ID_TBL},
{PA_CHANNEL_POSITION_TOP_REAR_RIGHT, MP_SPEAKER_ID_TBR},
{PA_CHANNEL_POSITION_TOP_REAR_CENTER, MP_SPEAKER_ID_TBC},
{PA_CHANNEL_POSITION_INVALID, -1}
};
static bool chmap_pa_from_mp(pa_channel_map *dst, struct mp_chmap *src)
{
if (src->num > PA_CHANNELS_MAX)
return false;
dst->channels = src->num;
if (mp_chmap_equals(src, &(const struct mp_chmap)MP_CHMAP_INIT_MONO)) {
dst->map[0] = PA_CHANNEL_POSITION_MONO;
return true;
}
for (int n = 0; n < src->num; n++) {
int mp_speaker = src->speaker[n];
int pa_speaker = PA_CHANNEL_POSITION_INVALID;
for (int i = 0; speaker_map[i][1] != -1; i++) {
if (speaker_map[i][1] == mp_speaker) {
pa_speaker = speaker_map[i][0];
break;
}
}
if (pa_speaker == PA_CHANNEL_POSITION_INVALID)
return false;
dst->map[n] = pa_speaker;
}
return true;
}
static bool select_chmap(struct ao *ao, pa_channel_map *dst)
{
struct mp_chmap_sel sel = {0};
for (int n = 0; speaker_map[n][1] != -1; n++)
mp_chmap_sel_add_speaker(&sel, speaker_map[n][1]);
return ao_chmap_sel_adjust(ao, &sel, &ao->channels) &&
chmap_pa_from_mp(dst, &ao->channels);
}
static void 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)
{
struct pa_sample_spec ss;
struct pa_channel_map map;
pa_proplist *proplist = NULL;
struct priv *priv = ao->priv;
char *host = priv->cfg_host && priv->cfg_host[0] ? priv->cfg_host : NULL;
char *sink = priv->cfg_sink && priv->cfg_sink[0] ? priv->cfg_sink : NULL;
const char *version = pa_get_library_version();
ao->per_application_mixer = true;
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_WARN(ao, "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_ERR(ao, "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_ERR(ao, "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;
ao->format = af_fmt_from_planar(ao->format);
const struct format_map *fmt_map = format_maps;
while (fmt_map->mp_format != ao->format) {
if (fmt_map->mp_format == AF_FORMAT_UNKNOWN) {
MP_VERBOSE(ao, "Unsupported format, using default\n");
fmt_map = format_maps;
break;
}
fmt_map++;
}
ao->format = fmt_map->mp_format;
ss.format = fmt_map->pa_format;
if (!pa_sample_spec_valid(&ss)) {
MP_ERR(ao, "Invalid sample spec\n");
goto unlock_and_fail;
}
if (!select_chmap(ao, &map))
goto unlock_and_fail;
if (!(proplist = pa_proplist_new())) {
MP_ERR(ao, "Failed to allocate proplist\n");
goto unlock_and_fail;
}
(void)pa_proplist_sets(proplist, PA_PROP_MEDIA_ROLE, "video");
if (!(priv->stream = pa_stream_new_with_proplist(priv->context,
"audio stream", &ss,
&map, proplist)))
goto unlock_and_fail;
pa_proplist_free(proplist);
proplist = NULL;
pa_stream_set_state_callback(priv->stream, stream_state_cb, ao);
pa_stream_set_write_callback(priv->stream, stream_request_cb, ao);
pa_stream_set_latency_update_callback(priv->stream,
stream_latency_update_cb, ao);
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);
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("Init failed");
}
if (proplist)
pa_proplist_free(proplist);
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("pa_stream_cork() failed");
}
// Play the specified data to the pulseaudio server
static int play(struct ao *ao, void **data, int samples, int flags)
{
struct priv *priv = ao->priv;
pa_threaded_mainloop_lock(priv->mainloop);
if (pa_stream_write(priv->stream, data[0], samples * ao->sstride, NULL, 0,
PA_SEEK_RELATIVE) < 0) {
GENERIC_ERR_MSG("pa_stream_write() failed");
samples = -1;
}
if (flags & AOPLAY_FINAL_CHUNK) {
// Force start in case the stream was too short for prebuf
pa_operation *op = pa_stream_trigger(priv->stream, NULL, NULL);
pa_operation_unref(op);
}
pa_threaded_mainloop_unlock(priv->mainloop);
return samples;
}
// Reset the audio stream, i.e. flush the playback buffer on the server side
static void reset(struct ao *ao)
{
// pa_stream_flush() works badly if not corked
cork(ao, true);
struct priv *priv = ao->priv;
pa_threaded_mainloop_lock(priv->mainloop);
priv->retval = 0;
if (!waitop(priv, pa_stream_flush(priv->stream, success_cb, ao)) ||
!priv->retval)
GENERIC_ERR_MSG("pa_stream_flush() failed");
cork(ao, false);
}
// Pause the audio stream by corking it on the server
static void pause(struct ao *ao)
{
cork(ao, true);
}
// Resume the audio stream by uncorking it on the server
static void resume(struct ao *ao)
{
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 samples that may be written to the server without blocking
static int get_space(struct ao *ao)
{
struct priv *priv = ao->priv;
pa_threaded_mainloop_lock(priv->mainloop);
size_t space = pa_stream_writable_size(priv->stream);
pa_threaded_mainloop_unlock(priv->mainloop);
return space / ao->sstride;
}
// 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("pa_stream_update_timing_info() failed");
return 0;
}
pa_threaded_mainloop_lock(priv->mainloop);
const pa_timing_info *ti = pa_stream_get_timing_info(priv->stream);
if (!ti) {
pa_threaded_mainloop_unlock(priv->mainloop);
GENERIC_ERR_MSG("pa_stream_get_timing_info() failed");
return 0;
}
const struct pa_sample_spec *ss = pa_stream_get_sample_spec(priv->stream);
if (!ss) {
pa_threaded_mainloop_unlock(priv->mainloop);
GENERIC_ERR_MSG("pa_stream_get_sample_spec() failed");
return 0;
}
// data left in PulseAudio's main buffers (not written to sink yet)
int64_t latency = pa_bytes_to_usec(ti->write_index - ti->read_index, ss);
// since this info may be from a while ago, playback has progressed since
latency -= ti->transport_usec;
// data already moved from buffers to sink, but not played yet
int64_t sink_latency = ti->sink_usec;
if (!ti->playing)
/* At the end of a stream, part of the data "left" in the sink may
* be padding silence after the end; that should be subtracted to
* get the amount of real audio from our stream. This adjustment
* is missing from Pulseaudio's own get_latency calculations
* (as of PulseAudio 2.1). */
sink_latency -= pa_bytes_to_usec(ti->since_underrun, ss);
if (sink_latency > 0)
latency += sink_latency;
if (latency < 0)
latency = 0;
pa_threaded_mainloop_unlock(priv->mainloop);
return latency / 1e6;
}
/* 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;
}
case AOCONTROL_UPDATE_STREAM_TITLE: {
char *title = (char *)arg;
pa_threaded_mainloop_lock(priv->mainloop);
if (!waitop(priv, pa_stream_set_name(priv->stream, title,
success_cb, ao)))
{
GENERIC_ERR_MSG("pa_stream_set_name() failed");
return CONTROL_ERROR;
}
return CONTROL_OK;
}
default:
return CONTROL_UNKNOWN;
}
}
#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}
},
};
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