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-rw-r--r--audio/out/ao_openal.c280
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diff --git a/audio/out/ao_openal.c b/audio/out/ao_openal.c
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+/*
+ * OpenAL audio output driver for MPlayer
+ *
+ * Copyleft 2006 by Reimar Döffinger (Reimar.Doeffinger@stud.uni-karlsruhe.de)
+ *
+ * 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
+ * along with MPlayer; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+#include "config.h"
+
+#include <stdlib.h>
+#include <stdio.h>
+#include <inttypes.h>
+#ifdef OPENAL_AL_H
+#include <OpenAL/alc.h>
+#include <OpenAL/al.h>
+#include <OpenAL/alext.h>
+#else
+#include <AL/alc.h>
+#include <AL/al.h>
+#include <AL/alext.h>
+#endif
+
+#include "mp_msg.h"
+
+#include "audio_out.h"
+#include "audio_out_internal.h"
+#include "libaf/format.h"
+#include "osdep/timer.h"
+#include "subopt-helper.h"
+
+static const ao_info_t info =
+{
+ "OpenAL audio output",
+ "openal",
+ "Reimar Döffinger <Reimar.Doeffinger@stud.uni-karlsruhe.de>",
+ ""
+};
+
+LIBAO_EXTERN(openal)
+
+#define MAX_CHANS 8
+#define NUM_BUF 128
+#define CHUNK_SIZE 512
+static ALuint buffers[MAX_CHANS][NUM_BUF];
+static ALuint sources[MAX_CHANS];
+
+static int cur_buf[MAX_CHANS];
+static int unqueue_buf[MAX_CHANS];
+static int16_t *tmpbuf;
+
+
+static int control(int cmd, void *arg) {
+ switch (cmd) {
+ case AOCONTROL_GET_VOLUME:
+ case AOCONTROL_SET_VOLUME: {
+ ALfloat volume;
+ ao_control_vol_t *vol = (ao_control_vol_t *)arg;
+ if (cmd == AOCONTROL_SET_VOLUME) {
+ volume = (vol->left + vol->right) / 200.0;
+ alListenerf(AL_GAIN, volume);
+ }
+ alGetListenerf(AL_GAIN, &volume);
+ vol->left = vol->right = volume * 100;
+ return CONTROL_TRUE;
+ }
+ }
+ return CONTROL_UNKNOWN;
+}
+
+/**
+ * \brief print suboption usage help
+ */
+static void print_help(void) {
+ mp_msg(MSGT_AO, MSGL_FATAL,
+ "\n-ao openal commandline help:\n"
+ "Example: mpv -ao openal:device=subdevice\n"
+ "\nOptions:\n"
+ " device=subdevice\n"
+ " Audio device OpenAL should use. Devices can be listed\n"
+ " with -ao openal:device=help\n"
+ );
+}
+
+static void list_devices(void) {
+ if (alcIsExtensionPresent(NULL, "ALC_ENUMERATE_ALL_EXT") != AL_TRUE) {
+ mp_msg(MSGT_AO, MSGL_FATAL, "Device listing not supported.\n");
+ return;
+ }
+ const char *list = alcGetString(NULL, ALC_ALL_DEVICES_SPECIFIER);
+ mp_msg(MSGT_AO, MSGL_FATAL, "OpenAL devices:\n");
+ while (list && *list) {
+ mp_msg(MSGT_AO, MSGL_FATAL, " '%s'\n", list);
+ list = list + strlen(list) + 1;
+ }
+}
+
+static int init(int rate, int channels, int format, int flags) {
+ float position[3] = {0, 0, 0};
+ float direction[6] = {0, 0, 1, 0, -1, 0};
+ float sppos[MAX_CHANS][3] = {
+ {-1, 0, 0.5}, {1, 0, 0.5},
+ {-1, 0, -1}, {1, 0, -1},
+ {0, 0, 1}, {0, 0, 0.1},
+ {-1, 0, 0}, {1, 0, 0},
+ };
+ ALCdevice *dev = NULL;
+ ALCcontext *ctx = NULL;
+ ALCint freq = 0;
+ ALCint attribs[] = {ALC_FREQUENCY, rate, 0, 0};
+ int i;
+ char *device = NULL;
+ const opt_t subopts[] = {
+ {"device", OPT_ARG_MSTRZ, &device, NULL},
+ {NULL}
+ };
+ global_ao->no_persistent_volume = true;
+ if (subopt_parse(ao_subdevice, subopts) != 0) {
+ print_help();
+ return 0;
+ }
+ if (device && strcmp(device, "help") == 0) {
+ list_devices();
+ goto err_out;
+ }
+ if (channels > MAX_CHANS) {
+ mp_msg(MSGT_AO, MSGL_FATAL, "[OpenAL] Invalid number of channels: %i\n", channels);
+ goto err_out;
+ }
+ dev = alcOpenDevice(device);
+ if (!dev) {
+ mp_msg(MSGT_AO, MSGL_FATAL, "[OpenAL] could not open device\n");
+ goto err_out;
+ }
+ ctx = alcCreateContext(dev, attribs);
+ alcMakeContextCurrent(ctx);
+ alListenerfv(AL_POSITION, position);
+ alListenerfv(AL_ORIENTATION, direction);
+ alGenSources(channels, sources);
+ for (i = 0; i < channels; i++) {
+ cur_buf[i] = 0;
+ unqueue_buf[i] = 0;
+ alGenBuffers(NUM_BUF, buffers[i]);
+ alSourcefv(sources[i], AL_POSITION, sppos[i]);
+ alSource3f(sources[i], AL_VELOCITY, 0, 0, 0);
+ }
+ if (channels == 1)
+ alSource3f(sources[0], AL_POSITION, 0, 0, 1);
+ ao_data.channels = channels;
+ alcGetIntegerv(dev, ALC_FREQUENCY, 1, &freq);
+ if (alcGetError(dev) == ALC_NO_ERROR && freq)
+ rate = freq;
+ ao_data.samplerate = rate;
+ ao_data.format = AF_FORMAT_S16_NE;
+ ao_data.bps = channels * rate * 2;
+ ao_data.buffersize = CHUNK_SIZE * NUM_BUF;
+ ao_data.outburst = channels * CHUNK_SIZE;
+ tmpbuf = malloc(CHUNK_SIZE);
+ free(device);
+ return 1;
+
+err_out:
+ free(device);
+ return 0;
+}
+
+// close audio device
+static void uninit(int immed) {
+ ALCcontext *ctx = alcGetCurrentContext();
+ ALCdevice *dev = alcGetContextsDevice(ctx);
+ free(tmpbuf);
+ if (!immed) {
+ ALint state;
+ alGetSourcei(sources[0], AL_SOURCE_STATE, &state);
+ while (state == AL_PLAYING) {
+ usec_sleep(10000);
+ alGetSourcei(sources[0], AL_SOURCE_STATE, &state);
+ }
+ }
+ reset();
+ alcMakeContextCurrent(NULL);
+ alcDestroyContext(ctx);
+ alcCloseDevice(dev);
+}
+
+static void unqueue_buffers(void) {
+ ALint p;
+ int s;
+ for (s = 0; s < ao_data.channels; s++) {
+ int till_wrap = NUM_BUF - unqueue_buf[s];
+ alGetSourcei(sources[s], AL_BUFFERS_PROCESSED, &p);
+ if (p >= till_wrap) {
+ alSourceUnqueueBuffers(sources[s], till_wrap, &buffers[s][unqueue_buf[s]]);
+ unqueue_buf[s] = 0;
+ p -= till_wrap;
+ }
+ if (p) {
+ alSourceUnqueueBuffers(sources[s], p, &buffers[s][unqueue_buf[s]]);
+ unqueue_buf[s] += p;
+ }
+ }
+}
+
+/**
+ * \brief stop playing and empty buffers (for seeking/pause)
+ */
+static void reset(void) {
+ alSourceStopv(ao_data.channels, sources);
+ unqueue_buffers();
+}
+
+/**
+ * \brief stop playing, keep buffers (for pause)
+ */
+static void audio_pause(void) {
+ alSourcePausev(ao_data.channels, sources);
+}
+
+/**
+ * \brief resume playing, after audio_pause()
+ */
+static void audio_resume(void) {
+ alSourcePlayv(ao_data.channels, sources);
+}
+
+static int get_space(void) {
+ ALint queued;
+ unqueue_buffers();
+ alGetSourcei(sources[0], AL_BUFFERS_QUEUED, &queued);
+ queued = NUM_BUF - queued - 3;
+ if (queued < 0) return 0;
+ return queued * CHUNK_SIZE * ao_data.channels;
+}
+
+/**
+ * \brief write data into buffer and reset underrun flag
+ */
+static int play(void *data, int len, int flags) {
+ ALint state;
+ int i, j, k;
+ int ch;
+ int16_t *d = data;
+ len /= ao_data.channels * CHUNK_SIZE;
+ for (i = 0; i < len; i++) {
+ for (ch = 0; ch < ao_data.channels; ch++) {
+ for (j = 0, k = ch; j < CHUNK_SIZE / 2; j++, k += ao_data.channels)
+ tmpbuf[j] = d[k];
+ alBufferData(buffers[ch][cur_buf[ch]], AL_FORMAT_MONO16, tmpbuf,
+ CHUNK_SIZE, ao_data.samplerate);
+ alSourceQueueBuffers(sources[ch], 1, &buffers[ch][cur_buf[ch]]);
+ cur_buf[ch] = (cur_buf[ch] + 1) % NUM_BUF;
+ }
+ d += ao_data.channels * CHUNK_SIZE / 2;
+ }
+ alGetSourcei(sources[0], AL_SOURCE_STATE, &state);
+ if (state != AL_PLAYING) // checked here in case of an underrun
+ alSourcePlayv(ao_data.channels, sources);
+ return len * ao_data.channels * CHUNK_SIZE;
+}
+
+static float get_delay(void) {
+ ALint queued;
+ unqueue_buffers();
+ alGetSourcei(sources[0], AL_BUFFERS_QUEUED, &queued);
+ return queued * CHUNK_SIZE / 2 / (float)ao_data.samplerate;
+}
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