From d4bdd0473d6f43132257c9fb3848d829755167a3 Mon Sep 17 00:00:00 2001 From: wm4 Date: Mon, 5 Nov 2012 17:02:04 +0100 Subject: 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. --- audio/out/ao_pulse.c | 554 +++++++++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 554 insertions(+) create mode 100644 audio/out/ao_pulse.c (limited to 'audio/out/ao_pulse.c') 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 +#include +#include + +#include + +#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, +}; -- cgit v1.2.3