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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
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')
-rw-r--r--audio/out/ao.c294
-rw-r--r--audio/out/ao.h140
-rw-r--r--audio/out/ao_alsa.c868
-rw-r--r--audio/out/ao_coreaudio.c1283
-rw-r--r--audio/out/ao_dsound.c648
-rw-r--r--audio/out/ao_jack.c361
-rw-r--r--audio/out/ao_lavc.c621
-rw-r--r--audio/out/ao_null.c129
-rw-r--r--audio/out/ao_openal.c280
-rw-r--r--audio/out/ao_oss.c560
-rw-r--r--audio/out/ao_pcm.c256
-rw-r--r--audio/out/ao_portaudio.c431
-rw-r--r--audio/out/ao_pulse.c554
-rw-r--r--audio/out/ao_rsound.c214
-rw-r--r--audio/out/audio_out_internal.h65
15 files changed, 6704 insertions, 0 deletions
diff --git a/audio/out/ao.c b/audio/out/ao.c
new file mode 100644
index 0000000000..ab8e60b753
--- /dev/null
+++ b/audio/out/ao.c
@@ -0,0 +1,294 @@
+/*
+ * 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 <stdio.h>
+#include <stdlib.h>
+#include <string.h>
+#include <assert.h>
+
+#include "talloc.h"
+
+#include "config.h"
+#include "audio_out.h"
+
+#include "mp_msg.h"
+
+// there are some globals:
+struct ao *global_ao;
+char *ao_subdevice = NULL;
+
+extern const struct ao_driver audio_out_oss;
+extern const struct ao_driver audio_out_coreaudio;
+extern const struct ao_driver audio_out_rsound;
+extern const struct ao_driver audio_out_pulse;
+extern const struct ao_driver audio_out_jack;
+extern const struct ao_driver audio_out_openal;
+extern const struct ao_driver audio_out_null;
+extern const struct ao_driver audio_out_alsa;
+extern const struct ao_driver audio_out_dsound;
+extern const struct ao_driver audio_out_pcm;
+extern const struct ao_driver audio_out_pss;
+extern const struct ao_driver audio_out_lavc;
+extern const struct ao_driver audio_out_portaudio;
+
+static const struct ao_driver * const audio_out_drivers[] = {
+// native:
+#ifdef CONFIG_COREAUDIO
+ &audio_out_coreaudio,
+#endif
+#ifdef CONFIG_PULSE
+ &audio_out_pulse,
+#endif
+#ifdef CONFIG_ALSA
+ &audio_out_alsa,
+#endif
+#ifdef CONFIG_OSS_AUDIO
+ &audio_out_oss,
+#endif
+#ifdef CONFIG_PORTAUDIO
+ &audio_out_portaudio,
+#endif
+#ifdef CONFIG_DSOUND
+ &audio_out_dsound,
+#endif
+ // wrappers:
+#ifdef CONFIG_JACK
+ &audio_out_jack,
+#endif
+#ifdef CONFIG_OPENAL
+ &audio_out_openal,
+#endif
+ &audio_out_null,
+ // should not be auto-selected:
+ &audio_out_pcm,
+#ifdef CONFIG_ENCODING
+ &audio_out_lavc,
+#endif
+#ifdef CONFIG_RSOUND
+ &audio_out_rsound,
+#endif
+ NULL
+};
+
+void list_audio_out(void)
+{
+ int i=0;
+ mp_tmsg(MSGT_AO, MSGL_INFO, "Available audio output drivers:\n");
+ mp_msg(MSGT_IDENTIFY, MSGL_INFO, "ID_AUDIO_OUTPUTS\n");
+ while (audio_out_drivers[i]) {
+ const ao_info_t *info = audio_out_drivers[i++]->info;
+ mp_msg(MSGT_GLOBAL, MSGL_INFO, "\t%s\t%s\n", info->short_name,
+ info->name);
+ }
+ mp_msg(MSGT_GLOBAL, MSGL_INFO,"\n");
+}
+
+struct ao *ao_create(struct MPOpts *opts, struct input_ctx *input)
+{
+ struct ao *r = talloc(NULL, struct ao);
+ *r = (struct ao){.outburst = 512, .buffersize = -1,
+ .opts = opts, .input_ctx = input };
+ return r;
+}
+
+void ao_init(struct ao *ao, char **ao_list)
+{
+ /* Caller adding child blocks is not supported as we may call
+ * talloc_free_children() to clean up after failed open attempts.
+ */
+ assert(talloc_total_blocks(ao) == 1);
+ struct ao backup = *ao;
+
+ if (!ao_list)
+ goto try_defaults;
+
+ // first try the preferred drivers, with their optional subdevice param:
+ while (*ao_list) {
+ char *ao_name = *ao_list;
+ if (!*ao_name)
+ goto try_defaults; // empty entry means try defaults
+ int ao_len;
+ char *params = strchr(ao_name, ':');
+ if (params) {
+ ao_len = params - ao_name;
+ params++;
+ } else
+ ao_len = strlen(ao_name);
+
+ mp_tmsg(MSGT_AO, MSGL_V,
+ "Trying preferred audio driver '%.*s', options '%s'\n",
+ ao_len, ao_name, params ? params : "[none]");
+
+ const struct ao_driver *audio_out = NULL;
+ for (int i = 0; audio_out_drivers[i]; i++) {
+ audio_out = audio_out_drivers[i];
+ if (!strncmp(audio_out->info->short_name, ao_name, ao_len))
+ break;
+ audio_out = NULL;
+ }
+ if (audio_out) {
+ // name matches, try it
+ ao->driver = audio_out;
+ if (audio_out->init(ao, params) >= 0) {
+ ao->driver = audio_out;
+ ao->initialized = true;
+ return;
+ }
+ mp_tmsg(MSGT_AO, MSGL_WARN,
+ "Failed to initialize audio driver '%s'\n", ao_name);
+ talloc_free_children(ao);
+ *ao = backup;
+ } else
+ mp_tmsg(MSGT_AO, MSGL_WARN, "No such audio driver '%.*s'\n",
+ ao_len, ao_name);
+ ++ao_list;
+ }
+ return;
+
+ try_defaults:
+ mp_tmsg(MSGT_AO, MSGL_V, "Trying every known audio driver...\n");
+
+ // now try the rest...
+ for (int i = 0; audio_out_drivers[i]; i++) {
+ const struct ao_driver *audio_out = audio_out_drivers[i];
+ ao->driver = audio_out;
+ ao->probing = true;
+ if (audio_out->init(ao, NULL) >= 0) {
+ ao->probing = false;
+ ao->initialized = true;
+ ao->driver = audio_out;
+ return;
+ }
+ talloc_free_children(ao);
+ *ao = backup;
+ }
+ return;
+}
+
+void ao_uninit(struct ao *ao, bool cut_audio)
+{
+ assert(ao->buffer.len >= ao->buffer_playable_size);
+ ao->buffer.len = ao->buffer_playable_size;
+ if (ao->initialized)
+ ao->driver->uninit(ao, cut_audio);
+ if (!cut_audio && ao->buffer.len)
+ mp_msg(MSGT_AO, MSGL_WARN, "Audio output truncated at end.\n");
+ talloc_free(ao);
+}
+
+int ao_play(struct ao *ao, void *data, int len, int flags)
+{
+ return ao->driver->play(ao, data, len, flags);
+}
+
+int ao_control(struct ao *ao, enum aocontrol cmd, void *arg)
+{
+ if (ao->driver->control)
+ return ao->driver->control(ao, cmd, arg);
+ return CONTROL_UNKNOWN;
+}
+
+double ao_get_delay(struct ao *ao)
+{
+ if (!ao->driver->get_delay) {
+ assert(ao->untimed);
+ return 0;
+ }
+ return ao->driver->get_delay(ao);
+}
+
+int ao_get_space(struct ao *ao)
+{
+ return ao->driver->get_space(ao);
+}
+
+void ao_reset(struct ao *ao)
+{
+ ao->buffer.len = 0;
+ ao->buffer_playable_size = 0;
+ if (ao->driver->reset)
+ ao->driver->reset(ao);
+}
+
+void ao_pause(struct ao *ao)
+{
+ if (ao->driver->pause)
+ ao->driver->pause(ao);
+}
+
+void ao_resume(struct ao *ao)
+{
+ if (ao->driver->resume)
+ ao->driver->resume(ao);
+}
+
+
+
+int old_ao_init(struct ao *ao, char *params)
+{
+ assert(!global_ao);
+ global_ao = ao;
+ ao_subdevice = params ? talloc_strdup(ao, params) : NULL;
+ if (ao->driver->old_functions->init(ao->samplerate, ao->channels,
+ ao->format, 0) == 0) {
+ global_ao = NULL;
+ return -1;
+ }
+ return 0;
+}
+
+void old_ao_uninit(struct ao *ao, bool cut_audio)
+{
+ ao->driver->old_functions->uninit(cut_audio);
+ global_ao = NULL;
+}
+
+int old_ao_play(struct ao *ao, void *data, int len, int flags)
+{
+ return ao->driver->old_functions->play(data, len, flags);
+}
+
+int old_ao_control(struct ao *ao, enum aocontrol cmd, void *arg)
+{
+ return ao->driver->old_functions->control(cmd, arg);
+}
+
+float old_ao_get_delay(struct ao *ao)
+{
+ return ao->driver->old_functions->get_delay();
+}
+
+int old_ao_get_space(struct ao *ao)
+{
+ return ao->driver->old_functions->get_space();
+}
+
+void old_ao_reset(struct ao *ao)
+{
+ ao->driver->old_functions->reset();
+}
+
+void old_ao_pause(struct ao *ao)
+{
+ ao->driver->old_functions->pause();
+}
+
+void old_ao_resume(struct ao *ao)
+{
+ ao->driver->old_functions->resume();
+}
diff --git a/audio/out/ao.h b/audio/out/ao.h
new file mode 100644
index 0000000000..9e172fd06c
--- /dev/null
+++ b/audio/out/ao.h
@@ -0,0 +1,140 @@
+/*
+ * 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.
+ */
+
+#ifndef MPLAYER_AUDIO_OUT_H
+#define MPLAYER_AUDIO_OUT_H
+
+#include <stdbool.h>
+
+#include "bstr.h"
+
+#define CONTROL_OK 1
+#define CONTROL_TRUE 1
+#define CONTROL_FALSE 0
+#define CONTROL_UNKNOWN -1
+#define CONTROL_ERROR -2
+#define CONTROL_NA -3
+
+enum aocontrol {
+ // _VOLUME commands take struct ao_control_vol pointer for input/output.
+ // If there's only one volume, SET should use average of left/right.
+ AOCONTROL_GET_VOLUME,
+ AOCONTROL_SET_VOLUME,
+ // _MUTE commands take a pointer to bool
+ AOCONTROL_GET_MUTE,
+ AOCONTROL_SET_MUTE,
+};
+
+#define AOPLAY_FINAL_CHUNK 1
+
+typedef struct ao_control_vol {
+ float left;
+ float right;
+} ao_control_vol_t;
+
+typedef struct ao_info {
+ /* driver name ("Matrox Millennium G200/G400" */
+ const char *name;
+ /* short name (for config strings) ("mga") */
+ const char *short_name;
+ /* author ("Aaron Holtzman <aholtzma@ess.engr.uvic.ca>") */
+ const char *author;
+ /* any additional comments */
+ const char *comment;
+} ao_info_t;
+
+/* interface towards mplayer and */
+typedef struct ao_old_functions {
+ int (*control)(int cmd, void *arg);
+ int (*init)(int rate, int channels, int format, int flags);
+ void (*uninit)(int immed);
+ void (*reset)(void);
+ int (*get_space)(void);
+ int (*play)(void *data, int len, int flags);
+ float (*get_delay)(void);
+ void (*pause)(void);
+ void (*resume)(void);
+} ao_functions_t;
+
+struct ao;
+
+struct ao_driver {
+ bool is_new;
+ const struct ao_info *info;
+ const struct ao_old_functions *old_functions;
+ int (*control)(struct ao *ao, enum aocontrol cmd, void *arg);
+ int (*init)(struct ao *ao, char *params);
+ void (*uninit)(struct ao *ao, bool cut_audio);
+ void (*reset)(struct ao*ao);
+ int (*get_space)(struct ao *ao);
+ int (*play)(struct ao *ao, void *data, int len, int flags);
+ float (*get_delay)(struct ao *ao);
+ void (*pause)(struct ao *ao);
+ void (*resume)(struct ao *ao);
+};
+
+/* global data used by mplayer and plugins */
+struct ao {
+ int samplerate;
+ int channels;
+ int format;
+ int bps;
+ int outburst;
+ int buffersize;
+ int brokenpts;
+ double pts;
+ struct bstr buffer;
+ int buffer_playable_size;
+ bool probing;
+ bool initialized;
+ bool untimed;
+ bool no_persistent_volume;
+ bool per_application_mixer;
+ const struct ao_driver *driver;
+ void *priv;
+ struct encode_lavc_context *encode_lavc_ctx;
+ struct MPOpts *opts;
+ struct input_ctx *input_ctx;
+};
+
+extern char *ao_subdevice;
+
+void list_audio_out(void);
+
+struct ao *ao_create(struct MPOpts *opts, struct input_ctx *input);
+void ao_init(struct ao *ao, char **ao_list);
+void ao_uninit(struct ao *ao, bool cut_audio);
+int ao_play(struct ao *ao, void *data, int len, int flags);
+int ao_control(struct ao *ao, enum aocontrol cmd, void *arg);
+double ao_get_delay(struct ao *ao);
+int ao_get_space(struct ao *ao);
+void ao_reset(struct ao *ao);
+void ao_pause(struct ao *ao);
+void ao_resume(struct ao *ao);
+
+int old_ao_control(struct ao *ao, enum aocontrol cmd, void *arg);
+int old_ao_init(struct ao *ao, char *params);
+void old_ao_uninit(struct ao *ao, bool cut_audio);
+void old_ao_reset(struct ao*ao);
+int old_ao_get_space(struct ao *ao);
+int old_ao_play(struct ao *ao, void *data, int len, int flags);
+float old_ao_get_delay(struct ao *ao);
+void old_ao_pause(struct ao *ao);
+void old_ao_resume(struct ao *ao);
+
+#endif /* MPLAYER_AUDIO_OUT_H */
diff --git a/audio/out/ao_alsa.c b/audio/out/ao_alsa.c
new file mode 100644
index 0000000000..27119112cb
--- /dev/null
+++ b/audio/out/ao_alsa.c
@@ -0,0 +1,868 @@
+/*
+ * ALSA 0.9.x-1.x audio output driver
+ *
+ * Copyright (C) 2004 Alex Beregszaszi
+ *
+ * modified for real ALSA 0.9.0 support by Zsolt Barat <joy@streamminister.de>
+ * additional AC-3 passthrough support by Andy Lo A Foe <andy@alsaplayer.org>
+ * 08/22/2002 iec958-init rewritten and merged with common init, zsolt
+ * 04/13/2004 merged with ao_alsa1.x, fixes provided by Jindrich Makovicka
+ * 04/25/2004 printfs converted to mp_msg, Zsolt.
+ *
+ * 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 <errno.h>
+#include <sys/time.h>
+#include <stdlib.h>
+#include <stdarg.h>
+#include <ctype.h>
+#include <math.h>
+#include <string.h>
+#include <alloca.h>
+
+#include "config.h"
+#include "subopt-helper.h"
+#include "mixer.h"
+#include "mp_msg.h"
+
+#define ALSA_PCM_NEW_HW_PARAMS_API
+#define ALSA_PCM_NEW_SW_PARAMS_API
+
+#include <alsa/asoundlib.h>
+
+#include "audio_out.h"
+#include "audio_out_internal.h"
+#include "libaf/format.h"
+
+static const ao_info_t info =
+{
+ "ALSA-0.9.x-1.x audio output",
+ "alsa",
+ "Alex Beregszaszi, Zsolt Barat <joy@streamminister.de>",
+ "under development"
+};
+
+LIBAO_EXTERN(alsa)
+
+static snd_pcm_t *alsa_handler;
+static snd_pcm_format_t alsa_format;
+
+#define BUFFER_TIME 500000 // 0.5 s
+#define FRAGCOUNT 16
+
+static size_t bytes_per_sample;
+
+static int alsa_can_pause;
+static snd_pcm_sframes_t prepause_frames;
+
+#define ALSA_DEVICE_SIZE 256
+
+static void alsa_error_handler(const char *file, int line, const char *function,
+ int err, const char *format, ...)
+{
+ char tmp[0xc00];
+ va_list va;
+
+ va_start(va, format);
+ vsnprintf(tmp, sizeof tmp, format, va);
+ va_end(va);
+
+ if (err)
+ mp_msg(MSGT_AO, MSGL_ERR, "[AO_ALSA] alsa-lib: %s:%i:(%s) %s: %s\n",
+ file, line, function, tmp, snd_strerror(err));
+ else
+ mp_msg(MSGT_AO, MSGL_ERR, "[AO_ALSA] alsa-lib: %s:%i:(%s) %s\n",
+ file, line, function, tmp);
+}
+
+/* to set/get/query special features/parameters */
+static int control(int cmd, void *arg)
+{
+ switch(cmd) {
+ case AOCONTROL_GET_MUTE:
+ case AOCONTROL_SET_MUTE:
+ case AOCONTROL_GET_VOLUME:
+ case AOCONTROL_SET_VOLUME:
+ {
+ int err;
+ snd_mixer_t *handle;
+ snd_mixer_elem_t *elem;
+ snd_mixer_selem_id_t *sid;
+
+ char *mix_name = "Master";
+ char *card = "default";
+ int mix_index = 0;
+
+ long pmin, pmax;
+ long get_vol, set_vol;
+ float f_multi;
+
+ if(AF_FORMAT_IS_AC3(ao_data.format) || AF_FORMAT_IS_IEC61937(ao_data.format))
+ return CONTROL_TRUE;
+
+ if(mixer_channel) {
+ char *test_mix_index;
+
+ mix_name = strdup(mixer_channel);
+ if ((test_mix_index = strchr(mix_name, ','))){
+ *test_mix_index = 0;
+ test_mix_index++;
+ mix_index = strtol(test_mix_index, &test_mix_index, 0);
+
+ if (*test_mix_index){
+ mp_tmsg(MSGT_AO,MSGL_ERR,
+ "[AO_ALSA] Invalid mixer index. Defaulting to 0.\n");
+ mix_index = 0 ;
+ }
+ }
+ }
+ if(mixer_device) card = mixer_device;
+
+ //allocate simple id
+ snd_mixer_selem_id_alloca(&sid);
+
+ //sets simple-mixer index and name
+ snd_mixer_selem_id_set_index(sid, mix_index);
+ snd_mixer_selem_id_set_name(sid, mix_name);
+
+ if (mixer_channel) {
+ free(mix_name);
+ mix_name = NULL;
+ }
+
+ if ((err = snd_mixer_open(&handle, 0)) < 0) {
+ mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Mixer open error: %s\n", snd_strerror(err));
+ return CONTROL_ERROR;
+ }
+
+ if ((err = snd_mixer_attach(handle, card)) < 0) {
+ mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Mixer attach %s error: %s\n",
+ card, snd_strerror(err));
+ snd_mixer_close(handle);
+ return CONTROL_ERROR;
+ }
+
+ if ((err = snd_mixer_selem_register(handle, NULL, NULL)) < 0) {
+ mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Mixer register error: %s\n", snd_strerror(err));
+ snd_mixer_close(handle);
+ return CONTROL_ERROR;
+ }
+ err = snd_mixer_load(handle);
+ if (err < 0) {
+ mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Mixer load error: %s\n", snd_strerror(err));
+ snd_mixer_close(handle);
+ return CONTROL_ERROR;
+ }
+
+ elem = snd_mixer_find_selem(handle, sid);
+ if (!elem) {
+ mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Unable to find simple control '%s',%i.\n",
+ snd_mixer_selem_id_get_name(sid), snd_mixer_selem_id_get_index(sid));
+ snd_mixer_close(handle);
+ return CONTROL_ERROR;
+ }
+
+ snd_mixer_selem_get_playback_volume_range(elem,&pmin,&pmax);
+ f_multi = (100 / (float)(pmax - pmin));
+
+ switch (cmd) {
+ case AOCONTROL_SET_VOLUME: {
+ ao_control_vol_t *vol = arg;
+ set_vol = vol->left / f_multi + pmin + 0.5;
+
+ //setting channels
+ if ((err = snd_mixer_selem_set_playback_volume(elem, SND_MIXER_SCHN_FRONT_LEFT, set_vol)) < 0) {
+ mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Error setting left channel, %s\n",
+ snd_strerror(err));
+ goto mixer_error;
+ }
+ mp_msg(MSGT_AO,MSGL_DBG2,"left=%li, ", set_vol);
+
+ set_vol = vol->right / f_multi + pmin + 0.5;
+
+ if ((err = snd_mixer_selem_set_playback_volume(elem, SND_MIXER_SCHN_FRONT_RIGHT, set_vol)) < 0) {
+ mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Error setting right channel, %s\n",
+ snd_strerror(err));
+ goto mixer_error;
+ }
+ mp_msg(MSGT_AO,MSGL_DBG2,"right=%li, pmin=%li, pmax=%li, mult=%f\n",
+ set_vol, pmin, pmax, f_multi);
+ break;
+ }
+ case AOCONTROL_GET_VOLUME: {
+ ao_control_vol_t *vol = arg;
+ snd_mixer_selem_get_playback_volume(elem, SND_MIXER_SCHN_FRONT_LEFT, &get_vol);
+ vol->left = (get_vol - pmin) * f_multi;
+ snd_mixer_selem_get_playback_volume(elem, SND_MIXER_SCHN_FRONT_RIGHT, &get_vol);
+ vol->right = (get_vol - pmin) * f_multi;
+ mp_msg(MSGT_AO,MSGL_DBG2,"left=%f, right=%f\n",vol->left,vol->right);
+ break;
+ }
+ case AOCONTROL_SET_MUTE: {
+ bool *mute = arg;
+ if (!snd_mixer_selem_has_playback_switch(elem))
+ goto mixer_error;
+ if (!snd_mixer_selem_has_playback_switch_joined(elem)) {
+ snd_mixer_selem_set_playback_switch(
+ elem, SND_MIXER_SCHN_FRONT_RIGHT, !*mute);
+ }
+ snd_mixer_selem_set_playback_switch(elem, SND_MIXER_SCHN_FRONT_LEFT,
+ !*mute);
+ break;
+ }
+ case AOCONTROL_GET_MUTE: {
+ bool *mute = arg;
+ if (!snd_mixer_selem_has_playback_switch(elem))
+ goto mixer_error;
+ int tmp = 1;
+ snd_mixer_selem_get_playback_switch(elem, SND_MIXER_SCHN_FRONT_LEFT,
+ &tmp);
+ *mute = !tmp;
+ if (!snd_mixer_selem_has_playback_switch_joined(elem)) {
+ snd_mixer_selem_get_playback_switch(
+ elem, SND_MIXER_SCHN_FRONT_RIGHT, &tmp);
+ *mute &= !tmp;
+ }
+ break;
+ }
+ }
+ snd_mixer_close(handle);
+ return CONTROL_OK;
+ mixer_error:
+ snd_mixer_close(handle);
+ return CONTROL_ERROR;
+ }
+
+ } //end switch
+ return CONTROL_UNKNOWN;
+}
+
+static void parse_device (char *dest, const char *src, int len)
+{
+ char *tmp;
+ memmove(dest, src, len);
+ dest[len] = 0;
+ while ((tmp = strrchr(dest, '.')))
+ tmp[0] = ',';
+ while ((tmp = strrchr(dest, '=')))
+ tmp[0] = ':';
+}
+
+static void print_help (void)
+{
+ mp_tmsg (MSGT_AO, MSGL_FATAL,
+ "\n[AO_ALSA] -ao alsa commandline help:\n"\
+ "[AO_ALSA] Example: mpv -ao alsa:device=hw=0.3\n"\
+ "[AO_ALSA] Sets first card fourth hardware device.\n\n"\
+ "[AO_ALSA] Options:\n"\
+ "[AO_ALSA] noblock\n"\
+ "[AO_ALSA] Opens device in non-blocking mode.\n"\
+ "[AO_ALSA] device=<device-name>\n"\
+ "[AO_ALSA] Sets device (change , to . and : to =)\n");
+}
+
+static int str_maxlen(void *strp) {
+ strarg_t *str = strp;
+ return str->len <= ALSA_DEVICE_SIZE;
+}
+
+static int try_open_device(const char *device, int open_mode, int try_ac3)
+{
+ int err, len;
+ char *ac3_device, *args;
+
+ if (try_ac3) {
+ /* to set the non-audio bit, use AES0=6 */
+ len = strlen(device);
+ ac3_device = malloc(len + 7 + 1);
+ if (!ac3_device)
+ return -ENOMEM;
+ strcpy(ac3_device, device);
+ args = strchr(ac3_device, ':');
+ if (!args) {
+ /* no existing parameters: add it behind device name */
+ strcat(ac3_device, ":AES0=6");
+ } else {
+ do
+ ++args;
+ while (isspace(*args));
+ if (*args == '\0') {
+ /* ":" but no parameters */
+ strcat(ac3_device, "AES0=6");
+ } else if (*args != '{') {
+ /* a simple list of parameters: add it at the end of the list */
+ strcat(ac3_device, ",AES0=6");
+ } else {
+ /* parameters in config syntax: add it inside the { } block */
+ do
+ --len;
+ while (len > 0 && isspace(ac3_device[len]));
+ if (ac3_device[len] == '}')
+ strcpy(ac3_device + len, " AES0=6}");
+ }
+ }
+ err = snd_pcm_open(&alsa_handler, ac3_device, SND_PCM_STREAM_PLAYBACK,
+ open_mode);
+ free(ac3_device);
+ if (!err)
+ return 0;
+ }
+ return snd_pcm_open(&alsa_handler, device, SND_PCM_STREAM_PLAYBACK,
+ open_mode);
+}
+
+/*
+ open & setup audio device
+ return: 1=success 0=fail
+*/
+static int init(int rate_hz, int channels, int format, int flags)
+{
+ int err;
+ int block;
+ strarg_t device;
+ snd_pcm_uframes_t chunk_size;
+ snd_pcm_uframes_t bufsize;
+ snd_pcm_uframes_t boundary;
+ const opt_t subopts[] = {
+ {"block", OPT_ARG_BOOL, &block, NULL},
+ {"device", OPT_ARG_STR, &device, str_maxlen},
+ {NULL}
+ };
+
+ char alsa_device[ALSA_DEVICE_SIZE + 1];
+ // make sure alsa_device is null-terminated even when using strncpy etc.
+ memset(alsa_device, 0, ALSA_DEVICE_SIZE + 1);
+
+ mp_msg(MSGT_AO,MSGL_V,"alsa-init: requested format: %d Hz, %d channels, %x\n", rate_hz,
+ channels, format);
+ alsa_handler = NULL;
+ mp_msg(MSGT_AO,MSGL_V,"alsa-init: using ALSA %s\n", snd_asoundlib_version());
+
+ prepause_frames = 0;
+
+ snd_lib_error_set_handler(alsa_error_handler);
+
+ ao_data.samplerate = rate_hz;
+ ao_data.format = format;
+ ao_data.channels = channels;
+
+ switch (format)
+ {
+ case AF_FORMAT_S8:
+ alsa_format = SND_PCM_FORMAT_S8;
+ break;
+ case AF_FORMAT_U8:
+ alsa_format = SND_PCM_FORMAT_U8;
+ break;
+ case AF_FORMAT_U16_LE:
+ alsa_format = SND_PCM_FORMAT_U16_LE;
+ break;
+ case AF_FORMAT_U16_BE:
+ alsa_format = SND_PCM_FORMAT_U16_BE;
+ break;
+ case AF_FORMAT_AC3_LE:
+ case AF_FORMAT_S16_LE:
+ case AF_FORMAT_IEC61937_LE:
+ alsa_format = SND_PCM_FORMAT_S16_LE;
+ break;
+ case AF_FORMAT_AC3_BE:
+ case AF_FORMAT_S16_BE:
+ case AF_FORMAT_IEC61937_BE:
+ alsa_format = SND_PCM_FORMAT_S16_BE;
+ break;
+ case AF_FORMAT_U32_LE:
+ alsa_format = SND_PCM_FORMAT_U32_LE;
+ break;
+ case AF_FORMAT_U32_BE:
+ alsa_format = SND_PCM_FORMAT_U32_BE;
+ break;
+ case AF_FORMAT_S32_LE:
+ alsa_format = SND_PCM_FORMAT_S32_LE;
+ break;
+ case AF_FORMAT_S32_BE:
+ alsa_format = SND_PCM_FORMAT_S32_BE;
+ break;
+ case AF_FORMAT_U24_LE:
+ alsa_format = SND_PCM_FORMAT_U24_3LE;
+ break;
+ case AF_FORMAT_U24_BE:
+ alsa_format = SND_PCM_FORMAT_U24_3BE;
+ break;
+ case AF_FORMAT_S24_LE:
+ alsa_format = SND_PCM_FORMAT_S24_3LE;
+ break;
+ case AF_FORMAT_S24_BE:
+ alsa_format = SND_PCM_FORMAT_S24_3BE;
+ break;
+ case AF_FORMAT_FLOAT_LE:
+ alsa_format = SND_PCM_FORMAT_FLOAT_LE;
+ break;
+ case AF_FORMAT_FLOAT_BE:
+ alsa_format = SND_PCM_FORMAT_FLOAT_BE;
+ break;
+ case AF_FORMAT_MU_LAW:
+ alsa_format = SND_PCM_FORMAT_MU_LAW;
+ break;
+ case AF_FORMAT_A_LAW:
+ alsa_format = SND_PCM_FORMAT_A_LAW;
+ break;
+
+ default:
+ alsa_format = SND_PCM_FORMAT_MPEG; //? default should be -1
+ break;
+ }
+
+ //subdevice parsing
+ // set defaults
+ block = 1;
+ /* switch for spdif
+ * sets opening sequence for SPDIF
+ * sets also the playback and other switches 'on the fly'
+ * while opening the abstract alias for the spdif subdevice
+ * 'iec958'
+ */
+ if (AF_FORMAT_IS_AC3(format) || AF_FORMAT_IS_IEC61937(format)) {
+ device.str = "iec958";
+ mp_msg(MSGT_AO,MSGL_V,"alsa-spdif-init: playing AC3/iec61937/iec958, %i channels\n", channels);
+ }
+ else
+ /* in any case for multichannel playback we should select
+ * appropriate device
+ */
+ switch (channels) {
+ case 1:
+ case 2:
+ device.str = "default";
+ mp_msg(MSGT_AO,MSGL_V,"alsa-init: setup for 1/2 channel(s)\n");
+ break;
+ case 4:
+ if (alsa_format == SND_PCM_FORMAT_FLOAT_LE)
+ // hack - use the converter plugin
+ device.str = "plug:surround40";
+ else
+ device.str = "surround40";
+ mp_msg(MSGT_AO,MSGL_V,"alsa-init: device set to surround40\n");
+ break;
+ case 6:
+ if (alsa_format == SND_PCM_FORMAT_FLOAT_LE)
+ device.str = "plug:surround51";
+ else
+ device.str = "surround51";
+ mp_msg(MSGT_AO,MSGL_V,"alsa-init: device set to surround51\n");
+ break;
+ case 8:
+ if (alsa_format == SND_PCM_FORMAT_FLOAT_LE)
+ device.str = "plug:surround71";
+ else
+ device.str = "surround71";
+ mp_msg(MSGT_AO,MSGL_V,"alsa-init: device set to surround71\n");
+ break;
+ default:
+ device.str = "default";
+ mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] %d channels are not supported.\n",channels);
+ }
+ device.len = strlen(device.str);
+ if (subopt_parse(ao_subdevice, subopts) != 0) {
+ print_help();
+ return 0;
+ }
+ parse_device(alsa_device, device.str, device.len);
+
+ mp_msg(MSGT_AO,MSGL_V,"alsa-init: using device %s\n", alsa_device);
+
+ alsa_can_pause = 1;
+
+ if (!alsa_handler) {
+ int open_mode = block ? 0 : SND_PCM_NONBLOCK;
+ int isac3 = AF_FORMAT_IS_AC3(format) || AF_FORMAT_IS_IEC61937(format);
+ //modes = 0, SND_PCM_NONBLOCK, SND_PCM_ASYNC
+ if ((err = try_open_device(alsa_device, open_mode, isac3)) < 0)
+ {
+ if (err != -EBUSY && !block) {
+ mp_tmsg(MSGT_AO,MSGL_INFO,"[AO_ALSA] Open in nonblock-mode failed, trying to open in block-mode.\n");
+ if ((err = try_open_device(alsa_device, 0, isac3)) < 0) {
+ mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Playback open error: %s\n", snd_strerror(err));
+ return 0;
+ }
+ } else {
+ mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Playback open error: %s\n", snd_strerror(err));
+ return 0;
+ }
+ }
+
+ if ((err = snd_pcm_nonblock(alsa_handler, 0)) < 0) {
+ mp_tmsg(MSGT_AO,MSGL_ERR,"[AL_ALSA] Error setting block-mode %s.\n", snd_strerror(err));
+ } else {
+ mp_msg(MSGT_AO,MSGL_V,"alsa-init: pcm opened in blocking mode\n");
+ }
+
+ snd_pcm_hw_params_t *alsa_hwparams;