summaryrefslogtreecommitdiffstats
path: root/audio/out/ao_pulse.c
diff options
context:
space:
mode:
authorwm4 <wm4@nowhere>2012-11-05 17:02:04 +0100
committerwm4 <wm4@nowhere>2012-11-12 20:06:14 +0100
commitd4bdd0473d6f43132257c9fb3848d829755167a3 (patch)
tree8021c2f7da1841393c8c832105e20cd527826d6c /audio/out/ao_pulse.c
parentbd48deba77bd5582c5829d6fe73a7d2571088aba (diff)
downloadmpv-d4bdd0473d6f43132257c9fb3848d829755167a3.tar.bz2
mpv-d4bdd0473d6f43132257c9fb3848d829755167a3.tar.xz
Rename directories, move files (step 1 of 2) (does not compile)
Tis drops the silly lib prefixes, and attempts to organize the tree in a more logical way. Make the top-level directory less cluttered as well. Renames the following directories: libaf -> audio/filter libao2 -> audio/out libvo -> video/out libmpdemux -> demux Split libmpcodecs: vf* -> video/filter vd*, dec_video.* -> video/decode mp_image*, img_format*, ... -> video/ ad*, dec_audio.* -> audio/decode libaf/format.* is moved to audio/ - this is similar to how mp_image.* is located in video/. Move most top-level .c/.h files to core. (talloc.c/.h is left on top- level, because it's external.) Park some of the more annoying files in compat/. Some of these are relicts from the time mplayer used ffmpeg internals. sub/ is not split, because it's too much of a mess (subtitle code is mixed with OSD display and rendering). Maybe the organization of core is not ideal: it mixes playback core (like mplayer.c) and utility helpers (like bstr.c/h). Should the need arise, the playback core will be moved somewhere else, while core contains all helper and common code.
Diffstat (limited to 'audio/out/ao_pulse.c')
-rw-r--r--audio/out/ao_pulse.c554
1 files changed, 554 insertions, 0 deletions
diff --git a/audio/out/ao_pulse.c b/audio/out/ao_pulse.c
new file mode 100644
index 0000000000..1d2ebc5281
--- /dev/null
+++ b/audio/out/ao_pulse.c
@@ -0,0 +1,554 @@
+/*
+ * 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 "libaf/format.h"
+#include "mp_msg.h"
+#include "audio_out.h"
+#include "input/input.h"
+
+#define PULSE_CLIENT_NAME "mpv"
+
+#define VOL_PA2MP(v) ((v) * 100 / PA_VOLUME_UI_MAX)
+#define VOL_MP2PA(v) ((v) * PA_VOLUME_UI_MAX / 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_MU_LAW, PA_SAMPLE_ULAW},
+ {AF_FORMAT_A_LAW, PA_SAMPLE_ALAW},
+ {AF_FORMAT_UNKNOWN, 0}
+};
+
+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;
+ }
+
+ ss.channels = ao->channels;
+ 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;
+ }
+
+ pa_channel_map_init_auto(&map, ss.channels, PA_CHANNEL_MAP_ALSA);
+ ao->bps = pa_bytes_per_second(&ss);
+
+ 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;
+
+ 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);
+ 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,
+};