Mercurial > mplayer.hg
view libao2/ao_openal.c @ 23588:a44fc12de744
Nico claims to never have had any problems with X11 compilation on Mandrake.
author | diego |
---|---|
date | Sat, 23 Jun 2007 08:22:10 +0000 |
parents | a29ae9b13a50 |
children | acfe034e5386 |
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/* * ao_openal.c - OpenAL audio output driver for MPlayer * * This driver is under the same license as MPlayer. * (http://www.mplayerhq.hu) * * Copyleft 2006 by Reimar Döffinger (Reimar.Doeffinger@stud.uni-karlsruhe.de) */ #include "config.h" #include <stdlib.h> #include <stdio.h> #include <inttypes.h> #ifdef OPENAL_AL_H #include <OpenAL/alc.h> #include <OpenAL/al.h> #else #include <AL/alc.h> #include <AL/al.h> #endif #include "mp_msg.h" #include "help_mp.h" #include "audio_out.h" #include "audio_out_internal.h" #include "libaf/af_format.h" #include "osdep/timer.h" #include "subopt-helper.h" static ao_info_t info = { "OpenAL audio output", "openal", "Reimar Döffinger <Reimar.Doeffinger@stud.uni-karlsruhe.de>", "" }; LIBAO_EXTERN(openal) #define MAX_CHANS 6 #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: mplayer -ao openal\n" "\nOptions:\n" ); } 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[6][3] = { {-1, 0, 0.5}, {1, 0, 0.5}, {-1, 0, -1}, {1, 0, -1}, {0, 0, 1}, {0, 0, 0.1}, }; ALCdevice *dev = NULL; ALCcontext *ctx = NULL; ALCint freq = 0; ALCint attribs[] = {ALC_FREQUENCY, rate, 0, 0}; int i; opt_t subopts[] = { {NULL} }; if (subopt_parse(ao_subdevice, subopts) != 0) { print_help(); return 0; } if (channels > MAX_CHANS) { mp_msg(MSGT_AO, MSGL_FATAL, "[OpenAL] Invalid number of channels: %i\n", channels); goto err_out; } dev = alcOpenDevice(NULL); 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); return 1; err_out: 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, i; for (s = 0; s < ao_data.channels; s++) { alGetSourcei(sources[s], AL_BUFFERS_PROCESSED, &p); for (i = 0; i < p; i++) { alSourceUnqueueBuffers(sources[s], 1, &buffers[s][unqueue_buf[s]]); unqueue_buf[s] = (unqueue_buf[s] + 1) % NUM_BUF; } } } /** * \brief stop playing and empty buffers (for seeking/pause) */ static void reset(void) { alSourceRewindv(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); return (NUM_BUF - 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.outburst; 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.outburst; } 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; }