diff options
Diffstat (limited to 'src/output')
| -rw-r--r-- | src/output/alsa_plugin.c | 69 | ||||
| -rw-r--r-- | src/output/fifo_output_plugin.c (renamed from src/output/fifo_plugin.c) | 6 | ||||
| -rw-r--r-- | src/output/httpd_client.c | 5 | ||||
| -rw-r--r-- | src/output/httpd_internal.h | 6 | ||||
| -rw-r--r-- | src/output/httpd_output_plugin.c | 22 | ||||
| -rw-r--r-- | src/output/jack_output_plugin.c (renamed from src/output/jack_plugin.c) | 257 | ||||
| -rw-r--r-- | src/output/openal_plugin.c | 273 | ||||
| -rw-r--r-- | src/output/oss_plugin.c | 3 | ||||
| -rw-r--r-- | src/output/pulse_output_plugin.c | 823 | ||||
| -rw-r--r-- | src/output/pulse_output_plugin.h | 72 | ||||
| -rw-r--r-- | src/output/pulse_plugin.c | 179 | ||||
| -rw-r--r-- | src/output/recorder_output_plugin.c | 214 | ||||
| -rw-r--r-- | src/output/shout_plugin.c | 13 |
13 files changed, 1655 insertions, 287 deletions
diff --git a/src/output/alsa_plugin.c b/src/output/alsa_plugin.c index 818c83ca2..870115998 100644 --- a/src/output/alsa_plugin.c +++ b/src/output/alsa_plugin.c @@ -183,6 +183,19 @@ get_bitformat(const struct audio_format *af) return SND_PCM_FORMAT_UNKNOWN; } +static snd_pcm_format_t +byteswap_bitformat(snd_pcm_format_t fmt) +{ + switch(fmt) { + case SND_PCM_FORMAT_S16_LE: return SND_PCM_FORMAT_S16_BE; + case SND_PCM_FORMAT_S24_LE: return SND_PCM_FORMAT_S24_BE; + case SND_PCM_FORMAT_S32_LE: return SND_PCM_FORMAT_S32_BE; + case SND_PCM_FORMAT_S16_BE: return SND_PCM_FORMAT_S16_LE; + case SND_PCM_FORMAT_S24_BE: return SND_PCM_FORMAT_S24_LE; + case SND_PCM_FORMAT_S32_BE: return SND_PCM_FORMAT_S32_LE; + default: return SND_PCM_FORMAT_UNKNOWN; + } +} /** * Set up the snd_pcm_t object which was opened by the caller. Set up * the configured settings and the audio format. @@ -208,7 +221,6 @@ alsa_setup(struct alsa_data *ad, struct audio_format *audio_format, configure_hw: /* configure HW params */ snd_pcm_hw_params_alloca(&hwparams); - cmd = "snd_pcm_hw_params_any"; err = snd_pcm_hw_params_any(ad->pcm, hwparams); if (err < 0) @@ -236,13 +248,38 @@ configure_hw: } err = snd_pcm_hw_params_set_format(ad->pcm, hwparams, bitformat); + if (err == -EINVAL && + byteswap_bitformat(bitformat) != SND_PCM_FORMAT_UNKNOWN) { + err = snd_pcm_hw_params_set_format(ad->pcm, hwparams, + byteswap_bitformat(bitformat)); + if (err == 0) { + g_debug("ALSA device \"%s\": converting %u bit to reverse-endian\n", + alsa_device(ad), audio_format->bits); + audio_format->reverse_endian = 1; + } + } if (err == -EINVAL && (audio_format->bits == 24 || audio_format->bits == 16)) { /* fall back to 32 bit, let pcm_convert.c do the conversion */ err = snd_pcm_hw_params_set_format(ad->pcm, hwparams, SND_PCM_FORMAT_S32); - if (err == 0) + if (err == 0) { + g_debug("ALSA device \"%s\": converting %u bit to 32 bit\n", + alsa_device(ad), audio_format->bits); + audio_format->bits = 32; + } + } + if (err == -EINVAL && (audio_format->bits == 24 || + audio_format->bits == 16)) { + /* fall back to 32 bit, let pcm_convert.c do the conversion */ + err = snd_pcm_hw_params_set_format(ad->pcm, hwparams, + byteswap_bitformat(SND_PCM_FORMAT_S32)); + if (err == 0) { + g_debug("ALSA device \"%s\": converting %u bit to 32 bit backward-endian\n", + alsa_device(ad), audio_format->bits); audio_format->bits = 32; + audio_format->reverse_endian = 1; + } } if (err == -EINVAL && audio_format->bits != 16) { @@ -255,6 +292,17 @@ configure_hw: audio_format->bits = 16; } } + if (err == -EINVAL && audio_format->bits != 16) { + /* fall back to 16 bit, let pcm_convert.c do the conversion */ + err = snd_pcm_hw_params_set_format(ad->pcm, hwparams, + byteswap_bitformat(SND_PCM_FORMAT_S16)); + if (err == 0) { + g_debug("ALSA device \"%s\": converting %u bit to 16 bit backward-endian\n", + alsa_device(ad), audio_format->bits); + audio_format->bits = 16; + audio_format->reverse_endian = 1; + } + } if (err < 0) { g_set_error(error, alsa_output_quark(), err, @@ -448,11 +496,19 @@ alsa_recover(struct alsa_data *ad, int err) } static void +alsa_drain(void *data) +{ + struct alsa_data *ad = data; + + snd_pcm_drain(ad->pcm); +} + +static void alsa_cancel(void *data) { struct alsa_data *ad = data; - alsa_recover(ad, snd_pcm_drop(ad->pcm)); + snd_pcm_drop(ad->pcm); } static void @@ -460,9 +516,6 @@ alsa_close(void *data) { struct alsa_data *ad = data; - if (snd_pcm_state(ad->pcm) == SND_PCM_STATE_RUNNING) - snd_pcm_drain(ad->pcm); - snd_pcm_close(ad->pcm); } @@ -494,7 +547,9 @@ const struct audio_output_plugin alsaPlugin = { .finish = alsa_finish, .open = alsa_open, .play = alsa_play, + .drain = alsa_drain, .cancel = alsa_cancel, .close = alsa_close, - .mixer_plugin = &alsa_mixer, + + .mixer_plugin = &alsa_mixer_plugin, }; diff --git a/src/output/fifo_plugin.c b/src/output/fifo_output_plugin.c index 76bbe8cfa..d3145748f 100644 --- a/src/output/fifo_plugin.c +++ b/src/output/fifo_output_plugin.c @@ -17,9 +17,9 @@ * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. */ -#include "../output_api.h" -#include "../utils.h" -#include "../timer.h" +#include "output_api.h" +#include "utils.h" +#include "timer.h" #include <glib.h> diff --git a/src/output/httpd_client.c b/src/output/httpd_client.c index 52a398e3b..8157ebb44 100644 --- a/src/output/httpd_client.c +++ b/src/output/httpd_client.c @@ -482,11 +482,6 @@ httpd_client_queue_size(const struct httpd_client *client) return size; } -/* g_queue_clear() was introduced in GLib 2.14 */ -#if !GLIB_CHECK_VERSION(2,14,0) -#define g_queue_clear(q) do { g_queue_free(q); q = g_queue_new(); } while (0) -#endif - void httpd_client_cancel(struct httpd_client *client) { diff --git a/src/output/httpd_internal.h b/src/output/httpd_internal.h index 2257e27a2..83e9498c6 100644 --- a/src/output/httpd_internal.h +++ b/src/output/httpd_internal.h @@ -97,6 +97,12 @@ struct httpd_output { * function. */ char buffer[32768]; + + /** + * The maximum and current number of clients connected + * at the same time. + */ + guint clients_max, clients_cnt; }; /** diff --git a/src/output/httpd_output_plugin.c b/src/output/httpd_output_plugin.c index 9fdf46456..675297cd3 100644 --- a/src/output/httpd_output_plugin.c +++ b/src/output/httpd_output_plugin.c @@ -76,6 +76,8 @@ httpd_output_init(G_GNUC_UNUSED const struct audio_format *audio_format, else httpd->content_type = "application/octet-stream"; + httpd->clients_max = config_get_block_unsigned(param,"max_clients", 0); + /* initialize listen address */ sin = (struct sockaddr_in *)&httpd->address; @@ -124,6 +126,7 @@ httpd_client_add(struct httpd_output *httpd, int fd) httpd->encoder->plugin->tag == NULL); httpd->clients = g_list_prepend(httpd->clients, client); + httpd->clients_cnt++; /* pass metadata to client */ if (httpd->metadata) @@ -146,10 +149,16 @@ httpd_listen_in_event(G_GNUC_UNUSED GIOChannel *source, connected */ fd = accept(httpd->fd, (struct sockaddr*)&sa, &sa_length); - if (fd >= 0) - httpd_client_add(httpd, fd); - else if (fd < 0 && errno != EINTR) + if (fd >= 0) { + /* can we allow additional client */ + if (!httpd->clients_max || + httpd->clients_cnt < httpd->clients_max) + httpd_client_add(httpd, fd); + else + close(fd); + } else if (fd < 0 && errno != EINTR) { g_warning("accept() failed: %s", g_strerror(errno)); + } g_mutex_unlock(httpd->mutex); @@ -237,6 +246,7 @@ httpd_output_open(void *data, struct audio_format *audio_format, /* initialize other attributes */ httpd->clients = NULL; + httpd->clients_cnt = 0; httpd->timer = timer_new(audio_format); g_mutex_unlock(httpd->mutex); @@ -281,6 +291,7 @@ httpd_output_remove_client(struct httpd_output *httpd, assert(client != NULL); httpd->clients = g_list_remove(httpd->clients, client); + httpd->clients_cnt--; } void @@ -433,9 +444,8 @@ httpd_output_tag(void *data, const struct tag *tag) page_unref (httpd->metadata); httpd->metadata = - icy_server_metadata_page(tag, TAG_ITEM_ALBUM, - TAG_ITEM_ARTIST, - TAG_ITEM_TITLE, + icy_server_metadata_page(tag, TAG_ALBUM, + TAG_ARTIST, TAG_TITLE, TAG_NUM_OF_ITEM_TYPES); if (httpd->metadata != NULL) { g_mutex_lock(httpd->mutex); diff --git a/src/output/jack_plugin.c b/src/output/jack_output_plugin.c index 5dc1eca24..fba921e04 100644 --- a/src/output/jack_plugin.c +++ b/src/output/jack_output_plugin.c @@ -17,7 +17,7 @@ * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. */ -#include "../output_api.h" +#include "output_api.h" #include "config.h" #include <assert.h> @@ -47,7 +47,7 @@ struct jack_data { /* configuration */ char *output_ports[2]; - int ringbuffer_size; + size_t ringbuffer_size; /* the current audio format */ struct audio_format audio_format; @@ -58,6 +58,12 @@ struct jack_data { jack_ringbuffer_t *ringbuffer[2]; bool shutdown; + + /** + * While this flag is set, the "process" callback generates + * silence. + */ + bool pause; }; /** @@ -69,43 +75,6 @@ jack_output_quark(void) return g_quark_from_static_string("jack_output"); } -static void -mpd_jack_client_free(struct jack_data *jd) -{ - assert(jd != NULL); - - if (jd->client != NULL) { - jack_deactivate(jd->client); - jack_client_close(jd->client); - jd->client = NULL; - } - - for (unsigned i = 0; i < G_N_ELEMENTS(jd->ringbuffer); ++i) { - if (jd->ringbuffer[i] != NULL) { - jack_ringbuffer_free(jd->ringbuffer[i]); - jd->ringbuffer[i] = NULL; - } - } -} - -static void -mpd_jack_free(struct jack_data *jd) -{ - assert(jd != NULL); - - for (unsigned i = 0; i < G_N_ELEMENTS(jd->output_ports); ++i) - g_free(jd->output_ports[i]); - - g_free(jd); -} - -static void -mpd_jack_finish(void *data) -{ - struct jack_data *jd = data; - mpd_jack_free(jd); -} - static int mpd_jack_process(jack_nframes_t nframes, void *arg) { @@ -116,6 +85,19 @@ mpd_jack_process(jack_nframes_t nframes, void *arg) if (nframes <= 0) return 0; + if (jd->pause) { + /* generate silence while MPD is paused */ + + for (unsigned i = 0; i < G_N_ELEMENTS(jd->ringbuffer); ++i) { + out = jack_port_get_buffer(jd->ports[i], nframes); + + for (jack_nframes_t f = 0; f < nframes; ++f) + out[f] = 0.0; + } + + return 0; + } + for (unsigned i = 0; i < G_N_ELEMENTS(jd->ringbuffer); ++i) { available = jack_ringbuffer_read_space(jd->ringbuffer[i]); assert(available % sample_size == 0); @@ -166,9 +148,65 @@ mpd_jack_info(const char *msg) } #endif +/** + * Disconnect the JACK client. + */ +static void +mpd_jack_disconnect(struct jack_data *jd) +{ + assert(jd != NULL); + assert(jd->client != NULL); + + jack_deactivate(jd->client); + jack_client_close(jd->client); + jd->client = NULL; +} + +/** + * Connect the JACK client and performs some basic setup + * (e.g. register callbacks). + */ +static bool +mpd_jack_connect(struct jack_data *jd, GError **error_r) +{ + assert(jd != NULL); + + jd->shutdown = false; + + if ((jd->client = jack_client_new(jd->name)) == NULL) { + g_set_error(error_r, jack_output_quark(), 0, + "Failed to connect to JACK server"); + return false; + } + + jack_set_process_callback(jd->client, mpd_jack_process, jd); + jack_on_shutdown(jd->client, mpd_jack_shutdown, jd); + + for (unsigned i = 0; i < G_N_ELEMENTS(jd->ports); ++i) { + jd->ports[i] = jack_port_register(jd->client, port_names[i], + JACK_DEFAULT_AUDIO_TYPE, + JackPortIsOutput, 0); + if (jd->ports[i] == NULL) { + g_set_error(error_r, jack_output_quark(), 0, + "Cannot register output port \"%s\"", + port_names[i]); + mpd_jack_disconnect(jd); + return false; + } + } + + return true; +} + +static bool +mpd_jack_test_default_device(void) +{ + return true; +} + static void * mpd_jack_init(G_GNUC_UNUSED const struct audio_format *audio_format, - const struct config_param *param, GError **error) + const struct config_param *param, GError **error_r) { struct jack_data *jd; const char *value; @@ -183,7 +221,7 @@ mpd_jack_init(G_GNUC_UNUSED const struct audio_format *audio_format, char **ports = g_strsplit(value, ",", 0); if (ports[0] == NULL || ports[1] == NULL || ports[2] != NULL) { - g_set_error(error, jack_output_quark(), 0, + g_set_error(error_r, jack_output_quark(), 0, "two port names expected in line %d", param->line); return NULL; @@ -210,47 +248,84 @@ mpd_jack_init(G_GNUC_UNUSED const struct audio_format *audio_format, return jd; } -static bool -mpd_jack_test_default_device(void) +static void +mpd_jack_finish(void *data) { - return true; + struct jack_data *jd = data; + + for (unsigned i = 0; i < G_N_ELEMENTS(jd->output_ports); ++i) + g_free(jd->output_ports[i]); + + g_free(jd); } static bool -mpd_jack_connect(struct jack_data *jd, GError **error) +mpd_jack_enable(void *data, GError **error_r) { - const char *output_ports[2], **jports; + struct jack_data *jd = (struct jack_data *)data; for (unsigned i = 0; i < G_N_ELEMENTS(jd->ringbuffer); ++i) - jd->ringbuffer[i] = - jack_ringbuffer_create(jd->ringbuffer_size); + jd->ringbuffer[i] = NULL; - jd->shutdown = false; + return mpd_jack_connect(jd, error_r); +} - if ((jd->client = jack_client_new(jd->name)) == NULL) { - g_set_error(error, jack_output_quark(), 0, - "Failed to connect to JACK server"); - return false; - } +static void +mpd_jack_disable(void *data) +{ + struct jack_data *jd = (struct jack_data *)data; - jack_set_process_callback(jd->client, mpd_jack_process, jd); - jack_on_shutdown(jd->client, mpd_jack_shutdown, jd); + if (jd->client != NULL) + mpd_jack_disconnect(jd); - for (unsigned i = 0; i < G_N_ELEMENTS(jd->ports); ++i) { - jd->ports[i] = jack_port_register(jd->client, port_names[i], - JACK_DEFAULT_AUDIO_TYPE, - JackPortIsOutput, 0); - if (jd->ports[i] == NULL) { - g_set_error(error, jack_output_quark(), 0, - "Cannot register output port \"%s\"", - port_names[i]); - return false; + for (unsigned i = 0; i < G_N_ELEMENTS(jd->ringbuffer); ++i) { + if (jd->ringbuffer[i] != NULL) { + jack_ringbuffer_free(jd->ringbuffer[i]); + jd->ringbuffer[i] = NULL; } } +} + +/** + * Stops the playback on the JACK connection. + */ +static void +mpd_jack_stop(struct jack_data *jd) +{ + assert(jd != NULL); + + if (jd->client == NULL) + return; + + if (jd->shutdown) + /* the connection has failed; close it */ + mpd_jack_disconnect(jd); + else + /* the connection is alive: just stop playback */ + jack_deactivate(jd->client); +} + +static bool +mpd_jack_start(struct jack_data *jd, GError **error_r) +{ + const char *output_ports[2], **jports; + + if (jd->client == NULL && !mpd_jack_connect(jd, error_r)) + return false; + + /* allocate the ring buffers on the first open(); these + persist until MPD exits. It's too unsafe to delete them + because we can never know when mpd_jack_process() gets + called */ + for (unsigned i = 0; i < G_N_ELEMENTS(jd->ringbuffer); ++i) + if (jd->ringbuffer[i] == NULL) + jd->ringbuffer[i] = + jack_ringbuffer_create(jd->ringbuffer_size); if ( jack_activate(jd->client) ) { - g_set_error(error, jack_output_quark(), 0, + g_set_error(error_r, jack_output_quark(), 0, "cannot activate client"); + mpd_jack_stop(jd); return false; } @@ -260,8 +335,9 @@ mpd_jack_connect(struct jack_data *jd, GError **error) jports = jack_get_ports(jd->client, NULL, NULL, JackPortIsPhysical | JackPortIsInput); if (jports == NULL) { - g_set_error(error, jack_output_quark(), 0, + g_set_error(error_r, jack_output_quark(), 0, "no ports found"); + mpd_jack_stop(jd); return false; } @@ -284,13 +360,14 @@ mpd_jack_connect(struct jack_data *jd, GError **error) ret = jack_connect(jd->client, jack_port_name(jd->ports[i]), output_ports[i]); if (ret != 0) { - g_set_error(error, jack_output_quark(), 0, + g_set_error(error_r, jack_output_quark(), 0, "Not a valid JACK port: %s", output_ports[i]); if (jports != NULL) free(jports); + mpd_jack_stop(jd); return false; } } @@ -302,16 +379,16 @@ mpd_jack_connect(struct jack_data *jd, GError **error) } static bool -mpd_jack_open(void *data, struct audio_format *audio_format, GError **error) +mpd_jack_open(void *data, struct audio_format *audio_format, GError **error_r) { struct jack_data *jd = data; assert(jd != NULL); - if (!mpd_jack_connect(jd, error)) { - mpd_jack_client_free(jd); + jd->pause = false; + + if (!mpd_jack_start(jd, error_r)) return false; - } set_audioformat(jd, audio_format); jd->audio_format = *audio_format; @@ -324,12 +401,7 @@ mpd_jack_close(G_GNUC_UNUSED void *data) { struct jack_data *jd = data; - mpd_jack_client_free(jd); -} - -static void -mpd_jack_cancel (G_GNUC_UNUSED void *data) -{ + mpd_jack_stop(jd); } static inline jack_default_audio_sample_t @@ -399,18 +471,20 @@ mpd_jack_write_samples(struct jack_data *jd, const void *src, } static size_t -mpd_jack_play(void *data, const void *chunk, size_t size, GError **error) +mpd_jack_play(void *data, const void *chunk, size_t size, GError **error_r) { struct jack_data *jd = data; const size_t frame_size = audio_format_frame_size(&jd->audio_format); size_t space = 0, space1; + jd->pause = false; + assert(size % frame_size == 0); size /= frame_size; while (true) { if (jd->shutdown) { - g_set_error(error, jack_output_quark(), 0, + g_set_error(error_r, jack_output_quark(), 0, "Refusing to play, because " "there is no client thread"); return 0; @@ -438,13 +512,32 @@ mpd_jack_play(void *data, const void *chunk, size_t size, GError **error) return size * frame_size; } -const struct audio_output_plugin jackPlugin = { +static bool +mpd_jack_pause(void *data) +{ + struct jack_data *jd = data; + + if (jd->shutdown) + return false; + + jd->pause = true; + + /* due to a MPD API limitation, we have to sleep a little bit + here, to avoid hogging the CPU */ + g_usleep(50000); + + return true; +} + +const struct audio_output_plugin jack_output_plugin = { .name = "jack", .test_default_device = mpd_jack_test_default_device, .init = mpd_jack_init, .finish = mpd_jack_finish, + .enable = mpd_jack_enable, + .disable = mpd_jack_disable, .open = mpd_jack_open, .play = mpd_jack_play, - .cancel = mpd_jack_cancel, + .pause = mpd_jack_pause, .close = mpd_jack_close, }; diff --git a/src/output/openal_plugin.c b/src/output/openal_plugin.c new file mode 100644 index 000000000..92ee82ef3 --- /dev/null +++ b/src/output/openal_plugin.c @@ -0,0 +1,273 @@ +/* + * Copyright (C) 2003-2009 The Music Player Daemon Project + * http://www.musicpd.org + * + * This program 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. + * + * This program 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 this program; if not, write to the Free Software Foundation, Inc., + * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. + */ + +#include "../output_api.h" +#include "../timer.h" +#include "config.h" + +#include <glib.h> + +#ifndef HAVE_OSX +#include <AL/al.h> +#include <AL/alc.h> +#else +#include <OpenAL/al.h> +#include <OpenAL/alc.h> +#endif + +#undef G_LOG_DOMAIN +#define G_LOG_DOMAIN "openal" + +/* should be enough for buffer size = 2048 */ +#define NUM_BUFFERS 16 + +struct openal_data { + const char *device_name; + ALCdevice *device; + ALCcontext *context; + Timer *timer; + ALuint buffers[NUM_BUFFERS]; + int filled; + ALuint source; + ALenum format; + ALuint frequency; +}; + +static inline GQuark +openal_output_quark(void) +{ + return g_quark_from_static_string("openal_output"); +} + +static ALenum +openal_audio_format(struct audio_format *audio_format) +{ + /* Only 8 and 16 bit samples are supported */ + if (audio_format->bits != 16 && audio_format->bits != 8) + audio_format->bits = 16; + + switch (audio_format->bits) + { + case 16: + if (audio_format->channels == 2) + return AL_FORMAT_STEREO16; + if (audio_format->channels == 1) + return AL_FORMAT_MONO16; + break; + + case 8: + if (audio_format->channels == 2) + return AL_FORMAT_STEREO8; + if (audio_format->channels == 1) + return AL_FORMAT_MONO8; + break; + } + + return 0; +} + +static bool +openal_setup_context(struct openal_data *od, + GError **error) +{ + od->device = alcOpenDevice(od->device_name); + + if (od->device == NULL) { + g_set_error(error, openal_output_quark(), 0, + "Error opening OpenAL device \"%s\"\n", + od->device_name); + return false; + } + + od->context = alcCreateContext(od->device, NULL); + + if (od->context == NULL) { + g_set_error(error, openal_output_quark(), 0, + "Error creating context for \"%s\"\n", + od->device_name); + alcCloseDevice(od->device); + return false; + } + + return true; +} + +static void +openal_unqueue_buffers(struct openal_data *od) +{ + ALint num; + ALuint buffer; + + alGetSourcei(od->source, AL_BUFFERS_QUEUED, &num); + + while (num--) { + alSourceUnqueueBuffers(od->source, 1, &buffer); + } +} + +static void * +openal_init(G_GNUC_UNUSED const struct audio_format *audio_format, + const struct config_param *param, + G_GNUC_UNUSED GError **error) +{ + const char *device_name = config_get_block_string(param, "device", NULL); + struct openal_data *od; + + if (device_name == NULL) { + device_name = alcGetString(NULL, ALC_DEFAULT_DEVICE_SPECIFIER); + } + + od = g_new(struct openal_data, 1); + od->device_name = device_name; + + return od; +} + +static void +openal_finish(void *data) +{ + struct openal_data *od = data; + + g_free(od); +} + +static bool +openal_open(void *data, struct audio_format *audio_format, + GError **error) +{ + struct openal_data *od = data; + + od->format = openal_audio_format(audio_format); + + if (!od->format) { + g_set_error(error, openal_output_quark(), 0, + "Unsupported audio format (%i channels, %i bps)", + audio_format->channels, + audio_format->bits); + return false; + } + + if (!openal_setup_context(od, error)) { + return false; + } + + alcMakeContextCurrent(od->context); + alGenBuffers(NUM_BUFFERS, od->buffers); + + if (alGetError() != AL_NO_ERROR) { + g_set_error(error, openal_output_quark(), 0, + "Failed to generate buffers"); + return false; + } + + alGenSources(1, &od->source); + + if (alGetError() != AL_NO_ERROR) { + g_set_error(error, openal_output_quark(), 0, + "Failed to generate source"); + alDeleteBuffers(NUM_BUFFERS, od->buffers); + return false; + } + + od->filled = 0; + od->timer = timer_new(audio_format); + od->frequency = audio_format->sample_rate; + + return true; +} + +static void +openal_close(void *data) +{ + struct openal_data *od = data; + + timer_free(od->timer); + alcMakeContextCurrent(od->context); + alDeleteSources(1, &od->source); + alDeleteBuffers(NUM_BUFFERS, od->buffers); + alcDestroyContext(od->context); + alcCloseDevice(od->device); +} + +static size_t +openal_play(void *data, const void *chunk, size_t size, + G_GNUC_UNUSED GError **error) +{ + struct openal_data *od = data; + ALuint buffer; + ALint num, state; + + if (alcGetCurrentContext() != od->context) { + alcMakeContextCurrent(od->context); + } + + alGetSourcei(od->source, AL_BUFFERS_PROCESSED, &num); + + if (od->filled < NUM_BUFFERS) { + /* fill all buffers */ + buffer = od->buffers[od->filled]; + od->filled++; + } else { + /* wait for processed buffer */ + while (num < 1) { + if (!od->timer->started) { + timer_start(od->timer); + } else { + timer_sync(od->timer); + } + + timer_add(od->timer, size); + + alGetSourcei(od->source, AL_BUFFERS_PROCESSED, &num); + } + + alSourceUnqueueBuffers(od->source, 1, &buffer); + } + + alBufferData(buffer, od->format, chunk, size, od->frequency); + alSourceQueueBuffers(od->source, 1, &buffer); + alGetSourcei(od->source, AL_SOURCE_STATE, &state); + + if (state != AL_PLAYING) { + alSourcePlay(od->source); + } + + return size; +} + +static void +openal_cancel(void *data) +{ + struct openal_data *od = data; + + od->filled = 0; + alcMakeContextCurrent(od->context); + alSourceStop(od->source); + openal_unqueue_buffers(od); +} + +const struct audio_output_plugin openal_output_plugin = { + .name = "openal", + .init = openal_init, + .finish = openal_finish, + .open = openal_open, + .close = openal_close, + .play = openal_play, + .cancel = openal_cancel, +}; diff --git a/src/output/oss_plugin.c b/src/output/oss_plugin.c index a66bc0598..258fbfa9d 100644 --- a/src/output/oss_plugin.c +++ b/src/output/oss_plugin.c @@ -601,5 +601,6 @@ const struct audio_output_plugin oss_output_plugin = { .close = oss_output_close, .play = oss_output_play, .cancel = oss_output_cancel, - .mixer_plugin = &oss_mixer, + + .mixer_plugin = &oss_mixer_plugin, }; diff --git a/src/output/pulse_output_plugin.c b/src/output/pulse_output_plugin.c new file mode 100644 index 000000000..13e1b6624 --- /dev/null +++ b/src/output/pulse_output_plugin.c @@ -0,0 +1,823 @@ +/* + * Copyright (C) 2003-2009 The Music Player Daemon Project + * http://www.musicpd.org + * + * This program 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. + * + * This program 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 this program; if not, write to the Free Software Foundation, Inc., + * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. + */ + +#include "pulse_output_plugin.h" +#include "output_api.h" +#include "mixer_list.h" +#include "mixer/pulse_mixer_plugin.h" + +#include <glib.h> + +#include <pulse/thread-mainloop.h> +#include <pulse/context.h> +#include <pulse/stream.h> +#include <pulse/introspect.h> +#include <pulse/subscribe.h> +#include <pulse/error.h> + +#include <assert.h> + +#define MPD_PULSE_NAME "Music Player Daemon" + +/** + * The quark used for GError.domain. + */ +static inline GQuark +pulse_output_quark(void) +{ + return g_quark_from_static_string("pulse_output"); +} + +void +pulse_output_set_mixer(struct pulse_output *po, struct pulse_mixer *pm) +{ + assert(po != NULL); + assert(po->mixer == NULL); + assert(pm != NULL); + + po->mixer = pm; + + if (po->mainloop == NULL) + return; + + pa_threaded_mainloop_lock(po->mainloop); + + if (po->context != NULL && + pa_context_get_state(po->context) == PA_CONTEXT_READY) { + pulse_mixer_on_connect(pm, po->context); + + if (po->stream != NULL && + pa_stream_get_state(po->stream) == PA_STREAM_READY) + pulse_mixer_on_change(pm, po->context, po->stream); + } + + pa_threaded_mainloop_unlock(po->mainloop); +} + +void +pulse_output_clear_mixer(struct pulse_output *po, struct pulse_mixer *pm) +{ + assert(po != NULL); + assert(pm != NULL); + assert(po->mixer == pm); + + po->mixer = NULL; +} + +bool +pulse_output_set_volume(struct pulse_output *po, + const struct pa_cvolume *volume, GError **error_r) +{ + pa_operation *o; + + if (po->context == NULL || po->stream == NULL || + pa_stream_get_state(po->stream) != PA_STREAM_READY) { + g_set_error(error_r, pulse_output_quark(), 0, "disconnected"); + return false; + } + + o = pa_context_set_sink_input_volume(po->context, + pa_stream_get_index(po->stream), + volume, NULL, NULL); + if (o == NULL) { + g_set_error(error_r, pulse_output_quark(), 0, + "failed to set PulseAudio volume: %s", + pa_strerror(pa_context_errno(po->context))); + return false; + } + + pa_operation_unref(o); + return true; +} + +/** + * \brief waits for a pulseaudio operation to finish, frees it and + * unlocks the mainloop + * \param operation the operation to wait for + * \return true if operation has finished normally (DONE state), + * false otherwise + */ +static bool +pulse_wait_for_operation(struct pa_threaded_mainloop *mainloop, + struct pa_operation *operation) +{ + pa_operation_state_t state; + + assert(mainloop != NULL); + assert(operation != NULL); + + state = pa_operation_get_state(operation); + while (state == PA_OPERATION_RUNNING) { + pa_threaded_mainloop_wait(mainloop); + state = pa_operation_get_state(operation); + } + + pa_operation_unref(operation); + + return state == PA_OPERATION_DONE; +} + +/** + * Callback function for stream operation. It just sends a signal to + * the caller thread, to wake pulse_wait_for_operation() up. + */ +static void +pulse_output_stream_success_cb(G_GNUC_UNUSED pa_stream *s, + G_GNUC_UNUSED int success, void *userdata) +{ + struct pulse_output *po = userdata; + + pa_threaded_mainloop_signal(po->mainloop, 0); +} + +static void +pulse_output_context_state_cb(struct pa_context *context, void *userdata) +{ + struct pulse_output *po = userdata; + + switch (pa_context_get_state(context)) { + case PA_CONTEXT_READY: + if (po->mixer != NULL) + pulse_mixer_on_connect(po->mixer, context); + + pa_threaded_mainloop_signal(po->mainloop, 0); + break; + + case PA_CONTEXT_TERMINATED: + case PA_CONTEXT_FAILED: + if (po->mixer != NULL) + pulse_mixer_on_disconnect(po->mixer); + + /* the caller thread might be waiting for these + states */ + pa_threaded_mainloop_signal(po->mainloop, 0); + break; + + case PA_CONTEXT_UNCONNECTED: + case PA_CONTEXT_CONNECTING: + case PA_CONTEXT_AUTHORIZING: + case PA_CONTEXT_SETTING_NAME: + break; + } +} + +static void +pulse_output_subscribe_cb(pa_context *context, + pa_subscription_event_type_t t, + uint32_t idx, void *userdata) +{ + struct pulse_output *po = userdata; + pa_subscription_event_type_t facility + = t & PA_SUBSCRIPTION_EVENT_FACILITY_MASK; + pa_subscription_event_type_t type + = t & PA_SUBSCRIPTION_EVENT_TYPE_MASK; + + if (po->mixer != NULL && + facility == PA_SUBSCRIPTION_EVENT_SINK_INPUT && + po->stream != NULL && + pa_stream_get_state(po->stream) == PA_STREAM_READY && + idx == pa_stream_get_index(po->stream) && + (type == PA_SUBSCRIPTION_EVENT_NEW || + type == PA_SUBSCRIPTION_EVENT_CHANGE)) + pulse_mixer_on_change(po->mixer, context, po->stream); +} + +/** + * Attempt to connect asynchronously to the PulseAudio server. + * + * @return true on success, false on error + */ +static bool +pulse_output_connect(struct pulse_output *po, GError **error_r) +{ + int error; + + error = pa_context_connect(po->context, po->server, + (pa_context_flags_t)0, NULL); + if (error < 0) { + g_set_error(error_r, pulse_output_quark(), 0, + "pa_context_connect() has failed: %s", + pa_strerror(pa_context_errno(po->context))); + return false; + } + + return true; +} + +/** + * Create, set up and connect a context. + * + * @return true on success, false on error + */ +static bool +pulse_output_setup_context(struct pulse_output *po, GError **error_r) +{ + po->context = pa_context_new(pa_threaded_mainloop_get_api(po->mainloop), + MPD_PULSE_NAME); + if (po->context == NULL) { + g_set_error(error_r, pulse_output_quark(), 0, + "pa_context_new() has failed"); + return false; + } + + pa_context_set_state_callback(po->context, + pulse_output_context_state_cb, po); + pa_context_set_subscribe_callback(po->context, + pulse_output_subscribe_cb, po); + + if (!pulse_output_connect(po, error_r)) { + pa_context_unref(po->context); + return false; + } + + return true; +} + +/** + * Frees and clears the context. + */ +static void +pulse_output_delete_context(struct pulse_output *po) +{ + pa_context_disconnect(po->context); + pa_context_unref(po->context); + po->context = NULL; +} + +static void * +pulse_output_init(G_GNUC_UNUSED const struct audio_format *audio_format, + const struct config_param *param, + G_GNUC_UNUSED GError **error_r) +{ + struct pulse_output *po; + + g_setenv("PULSE_PROP_media.role", "music", true); + + po = g_new(struct pulse_output, 1); + po->name = config_get_block_string(param, "name", "mpd_pulse"); + po->server = config_get_block_string(param, "server", NULL); + po->sink = config_get_block_string(param, "sink", NULL); + + po->mixer = NULL; + po->mainloop = NULL; + po->context = NULL; + po->stream = NULL; + + return po; +} + +static void +pulse_output_finish(void *data) +{ + struct pulse_output *po = data; + + g_free(po); +} + +static bool +pulse_output_enable(void *data, GError **error_r) +{ + struct pulse_output *po = data; + + assert(po->mainloop == NULL); + assert(po->context == NULL); + + /* create the libpulse mainloop and start the thread */ + + po->mainloop = pa_threaded_mainloop_new(); + if (po->mainloop == NULL) { + g_free(po); + + g_set_error(error_r, pulse_output_quark(), 0, + "pa_threaded_mainloop_new() has failed"); + return false; + } + + pa_threaded_mainloop_lock(po->mainloop); + + if (pa_threaded_mainloop_start(po->mainloop) < 0) { + pa_threaded_mainloop_unlock(po->mainloop); + pa_threaded_mainloop_free(po->mainloop); + g_free(po); + + g_set_error(error_r, pulse_output_quark(), 0, + "pa_threaded_mainloop_start() has failed"); + return false; + } + + pa_threaded_mainloop_unlock(po->mainloop); + + /* create the libpulse context and connect it */ + + pa_threaded_mainloop_lock(po->mainloop); + + if (!pulse_output_setup_context(po, error_r)) { + pa_threaded_mainloop_unlock(po->mainloop); + pa_threaded_mainloop_stop(po->mainloop); + pa_threaded_mainloop_free(po->mainloop); + g_free(po); + return false; + } + + pa_threaded_mainloop_unlock(po->mainloop); + + return true; +} + +static void +pulse_output_disable(void *data) +{ + struct pulse_output *po = data; + + pa_threaded_mainloop_stop(po->mainloop); + if (po->context != NULL) + pulse_output_delete_context(po); + pa_threaded_mainloop_free(po->mainloop); + po->mainloop = NULL; +} + +/** + * Check if the context is (already) connected, and waits if not. If + * the context has been disconnected, retry to connect. + * + * @return true on success, false on error + */ +static bool +pulse_output_wait_connection(struct pulse_output *po, GError **error_r) +{ + pa_context_state_t state; + + pa_threaded_mainloop_lock(po->mainloop); + + if (po->context == NULL && !pulse_output_setup_context(po, error_r)) + return false; + + while (true) { + state = pa_context_get_state(po->context); + switch (state) { + case PA_CONTEXT_READY: + /* nothing to do */ + pa_threaded_mainloop_unlock(po->mainloop); + return true; + + case PA_CONTEXT_UNCONNECTED: + case PA_CONTEXT_TERMINATED: + case PA_CONTEXT_FAILED: + /* failure */ + g_set_error(error_r, pulse_output_quark(), 0, + "failed to connect: %s", + pa_strerror(pa_context_errno(po->context))); + pulse_output_delete_context(po); + pa_threaded_mainloop_unlock(po->mainloop); + return false; + + case PA_CONTEXT_CONNECTING: + case PA_CONTEXT_AUTHORIZING: + case PA_CONTEXT_SETTING_NAME: + /* wait some more */ + pa_threaded_mainloop_wait(po->mainloop); + break; + } + } +} + +static void +pulse_output_stream_state_cb(pa_stream *stream, void *userdata) +{ + struct pulse_output *po = userdata; + + switch (pa_stream_get_state(stream)) { + case PA_STREAM_READY: + if (po->mixer != NULL) + pulse_mixer_on_change(po->mixer, po->context, stream); + + pa_threaded_mainloop_signal(po->mainloop, 0); + break; + + case PA_STREAM_FAILED: + case PA_STREAM_TERMINATED: + if (po->mixer != NULL) + pulse_mixer_on_disconnect(po->mixer); + + pa_threaded_mainloop_signal(po->mainloop, 0); + break; + + case PA_STREAM_UNCONNECTED: + case PA_STREAM_CREATING: + break; + } +} + +static void +pulse_output_stream_write_cb(G_GNUC_UNUSED pa_stream *stream, size_t nbytes, + void *userdata) +{ + struct pulse_output *po = userdata; + + po->writable = nbytes; + pa_threaded_mainloop_signal(po->mainloop, 0); +} + +static bool +pulse_output_open(void *data, struct audio_format *audio_format, + GError **error_r) +{ + struct pulse_output *po = data; + pa_sample_spec ss; + int error; + + if (po->context != NULL) { + switch (pa_context_get_state(po->context)) { + case PA_CONTEXT_UNCONNECTED: + case PA_CONTEXT_TERMINATED: + case PA_CONTEXT_FAILED: + /* the connection was closed meanwhile; delete + it, and pulse_output_wait_connection() will + reopen it */ + pulse_output_delete_context(po); + break; + + case PA_CONTEXT_READY: + case PA_CONTEXT_CONNECTING: + case PA_CONTEXT_AUTHORIZING: + case PA_CONTEXT_SETTING_NAME: + break; + } + } + + if (!pulse_output_wait_connection(po, error_r)) + return false; + + /* MPD doesn't support the other pulseaudio sample formats, so + we just force MPD to send us everything as 16 bit */ + audio_format->bits = 16; + + ss.format = PA_SAMPLE_S16NE; + ss.rate = audio_format->sample_rate; + ss.channels = audio_format->channels; + + pa_threaded_mainloop_lock(po->mainloop); + + /* create a stream .. */ + + po->stream = pa_stream_new(po->context, po->name, &ss, NULL); + if (po->stream == NULL) { + g_set_error(error_r, pulse_output_quark(), 0, + "pa_stream_new() has failed: %s", + pa_strerror(pa_context_errno(po->context))); + pa_threaded_mainloop_unlock(po->mainloop); + return false; + } + + pa_stream_set_state_callback(po->stream, + pulse_output_stream_state_cb, po); + pa_stream_set_write_callback(po->stream, + pulse_output_stream_write_cb, po); + + /* .. and connect it (asynchronously) */ + + error = pa_stream_connect_playback(po->stream, po->sink, + NULL, 0, NULL, NULL); + if (error < 0) { + pa_stream_unref(po->stream); + po->stream = NULL; + + g_set_error(error_r, pulse_output_quark(), 0, + "pa_stream_connect_playback() has failed: %s", + pa_strerror(pa_context_errno(po->context))); + pa_threaded_mainloop_unlock(po->mainloop); + return false; + } + + pa_threaded_mainloop_unlock(po->mainloop); + +#if !PA_CHECK_VERSION(0,9,11) + po->pause = false; +#endif + + return true; +} + +static void +pulse_output_close(void *data) +{ + struct pulse_output *po = data; + pa_operation *o; + + pa_threaded_mainloop_lock(po->mainloop); + + if (pa_stream_get_state(po->stream) == PA_STREAM_READY) { + o = pa_stream_drain(po->stream, + pulse_output_stream_success_cb, po); + if (o == NULL) { + g_warning("pa_stream_drain() has failed: %s", + pa_strerror(pa_context_errno(po->context))); + } else + pulse_wait_for_operation(po->mainloop, o); + } + + pa_stream_disconnect(po->stream); + pa_stream_unref(po->stream); + po->stream = NULL; + + if (po->context != NULL && + pa_context_get_state(po->context) != PA_CONTEXT_READY) + pulse_output_delete_context(po); + + pa_threaded_mainloop_unlock(po->mainloop); +} + +/** + * Check if the stream is (already) connected, and waits for a signal + * if not. The mainloop must be locked before calling this function. + * + * @return the current stream state + */ +static pa_stream_state_t +pulse_output_check_stream(struct pulse_output *po) +{ + pa_stream_state_t state = pa_stream_get_state(po->stream); + + switch (state) { + case PA_STREAM_READY: + case PA_STREAM_FAILED: + case PA_STREAM_TERMINATED: + case PA_STREAM_UNCONNECTED: + break; + + case PA_STREAM_CREATING: + pa_threaded_mainloop_wait(po->mainloop); + state = pa_stream_get_state(po->stream); + break; + } + + return state; +} + +/** + * Check if the stream is (already) connected, and waits if not. The + * mainloop must be locked before calling this function. + * + * @return true on success, false on error + */ +static bool +pulse_output_wait_stream(struct pulse_output *po, GError **error_r) +{ + pa_stream_state_t state = pa_stream_get_state(po->stream); + + switch (state) { + case PA_STREAM_READY: + return true; + + case PA_STREAM_FAILED: + case PA_STREAM_TERMINATED: + case PA_STREAM_UNCONNECTED: + g_set_error(error_r, pulse_output_quark(), 0, + "disconnected"); + return false; + + case PA_STREAM_CREATING: + break; + } + + do { + state = pulse_output_check_stream(po); + } while (state == PA_STREAM_CREATING); + + if (state != PA_STREAM_READY) { + g_set_error(error_r, pulse_output_quark(), 0, + "failed to connect the stream: %s", + pa_strerror(pa_context_errno(po->context))); + return false; + } + + return true; +} + +/** + * Determines whether the stream is paused. On libpulse older than + * 0.9.11, it uses a custom pause flag. + */ +static bool +pulse_output_stream_is_paused(struct pulse_output *po) +{ + assert(po->stream != NULL); + +#if !defined(PA_CHECK_VERSION) || !PA_CHECK_VERSION(0,9,11) + return po->pause; +#else + return pa_stream_is_corked(po->stream); +#endif +} + +/** + * Sets cork mode on the stream. + */ +static bool +pulse_output_stream_pause(struct pulse_output *po, bool pause, + GError **error_r) +{ + pa_operation *o; + + assert(po->stream != NULL); + + o = pa_stream_cork(po->stream, pause, + pulse_output_stream_success_cb, po); + if (o == NULL) { + g_set_error(error_r, pulse_output_quark(), 0, + "pa_stream_cork() has failed: %s", + pa_strerror(pa_context_errno(po->context))); + return false; + } + + if (!pulse_wait_for_operation(po->mainloop, o)) { + g_set_error(error_r, pulse_output_quark(), 0, + "pa_stream_cork() has failed: %s", + pa_strerror(pa_context_errno(po->context))); + return false; + } + +#if !PA_CHECK_VERSION(0,9,11) + po->pause = pause; +#endif + return true; +} + +static size_t +pulse_output_play(void *data, const void *chunk, size_t size, GError **error_r) +{ + struct pulse_output *po = data; + int error; + + assert(po->stream != NULL); + + pa_threaded_mainloop_lock(po->mainloop); + + /* check if the stream is (already) connected */ + + if (!pulse_output_wait_stream(po, error_r)) { + pa_threaded_mainloop_unlock(po->mainloop); + return 0; + } + + assert(po->context != NULL); + + /* unpause if previously paused */ + + if (pulse_output_stream_is_paused(po) && + !pulse_output_stream_pause(po, false, error_r)) + return 0; + + /* wait until the server allows us to write */ + + while (po->writable == 0) { + pa_threaded_mainloop_wait(po->mainloop); + + if (pa_stream_get_state(po->stream) != PA_STREAM_READY) { + pa_threaded_mainloop_unlock(po->mainloop); + g_set_error(error_r, pulse_output_quark(), 0, + "disconnected"); + return false; + } + } + + /* now write */ + + if (size > po->writable) + /* don't send more than possible */ + size = po->writable; + + po->writable -= size; + + error = pa_stream_write(po->stream, chunk, size, NULL, + 0, PA_SEEK_RELATIVE); + pa_threaded_mainloop_unlock(po->mainloop); + if (error < 0) { + g_set_error(error_r, pulse_output_quark(), error, + "%s", pa_strerror(error)); + return 0; + } + + return size; +} + +static void +pulse_output_cancel(void *data) +{ + struct pulse_output *po = data; + pa_operation *o; + + assert(po->stream != NULL); + + pa_threaded_mainloop_lock(po->mainloop); + + if (pa_stream_get_state(po->stream) != PA_STREAM_READY) { + /* no need to flush when the stream isn't connected + yet */ + pa_threaded_mainloop_unlock(po->mainloop); + return; + } + + assert(po->context != NULL); + + o = pa_stream_flush(po->stream, pulse_output_stream_success_cb, po); + if (o == NULL) { + g_warning("pa_stream_flush() has failed: %s", + pa_strerror(pa_context_errno(po->context))); + pa_threaded_mainloop_unlock(po->mainloop); + return; + } + + pulse_wait_for_operation(po->mainloop, o); + pa_threaded_mainloop_unlock(po->mainloop); +} + +static bool +pulse_output_pause(void *data) +{ + struct pulse_output *po = data; + GError *error = NULL; + + assert(po->stream != NULL); + + pa_threaded_mainloop_lock(po->mainloop); + + /* check if the stream is (already/still) connected */ + + if (!pulse_output_wait_stream(po, &error)) { + pa_threaded_mainloop_unlock(po->mainloop); + g_warning("%s", error->message); + g_error_free(error); + return false; + } + + assert(po->context != NULL); + + /* cork the stream */ + + if (pulse_output_stream_is_paused(po)) { + /* already paused; due to a MPD API limitation, we + have to sleep a little bit here, to avoid hogging + the CPU */ + + g_usleep(50000); + } else if (!pulse_output_stream_pause(po, true, &error)) { + pa_threaded_mainloop_unlock(po->mainloop); + g_warning("%s", error->message); + g_error_free(error); + return false; + } + + pa_threaded_mainloop_unlock(po->mainloop); + + return true; +} + +static bool +pulse_output_test_default_device(void) +{ + struct pulse_output *po; + bool success; + + po = pulse_output_init(NULL, NULL, NULL); + if (po == NULL) + return false; + + success = pulse_output_wait_connection(po, NULL); + pulse_output_finish(po); + + return success; +} + +const struct audio_output_plugin pulse_output_plugin = { + .name = "pulse", + + .test_default_device = pulse_output_test_default_device, + .init = pulse_output_init, + .finish = pulse_output_finish, + .enable = pulse_output_enable, + .disable = pulse_output_disable, + .open = pulse_output_open, + .play = pulse_output_play, + .cancel = pulse_output_cancel, + .pause = pulse_output_pause, + .close = pulse_output_close, + + .mixer_plugin = &pulse_mixer_plugin, +}; diff --git a/src/output/pulse_output_plugin.h b/src/output/pulse_output_plugin.h new file mode 100644 index 000000000..e6b37443f --- /dev/null +++ b/src/output/pulse_output_plugin.h @@ -0,0 +1,72 @@ +/* + * Copyright (C) 2003-2009 The Music Player Daemon Project + * http://www.musicpd.org + * + * This program 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. + * + * This program 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 this program; if not, write to the Free Software Foundation, Inc., + * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. + */ + +#ifndef MPD_PULSE_OUTPUT_PLUGIN_H +#define MPD_PULSE_OUTPUT_PLUGIN_H + +#include <stdbool.h> +#include <stddef.h> + +#include <glib.h> + +#include <pulse/version.h> + +#if !defined(PA_CHECK_VERSION) +/** + * This macro was implemented in libpulse 0.9.16. + */ +#define PA_CHECK_VERSION(a,b,c) false +#endif + +struct pa_operation; +struct pa_cvolume; + +struct pulse_output { + const char *name; + const char *server; + const char *sink; + + struct pulse_mixer *mixer; + + struct pa_threaded_mainloop *mainloop; + struct pa_context *context; + struct pa_stream *stream; + + size_t writable; + +#if !PA_CHECK_VERSION(0,9,11) + /** + * We need this variable because pa_stream_is_corked() wasn't + * added before 0.9.11. + */ + bool pause; +#endif +}; + +void +pulse_output_set_mixer(struct pulse_output *po, struct pulse_mixer *pm); + +void +pulse_output_clear_mixer(struct pulse_output *po, struct pulse_mixer *pm); + +bool +pulse_output_set_volume(struct pulse_output *po, + const struct pa_cvolume *volume, GError **error_r); + +#endif diff --git a/src/output/pulse_plugin.c b/src/output/pulse_plugin.c deleted file mode 100644 index ffc7abc8b..000000000 --- a/src/output/pulse_plugin.c +++ /dev/null @@ -1,179 +0,0 @@ -/* - * Copyright (C) 2003-2009 The Music Player Daemon Project - * http://www.musicpd.org - * - * This program 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. - * - * This program 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 this program; if not, write to the Free Software Foundation, Inc., - * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. - */ - -#include "../output_api.h" -#include "mixer_list.h" - -#include <glib.h> -#include <pulse/simple.h> -#include <pulse/error.h> - -#define MPD_PULSE_NAME "mpd" - -struct pulse_data { - const char *name; - const char *server; - const char *sink; - - pa_simple *s; -}; - -/** - * The quark used for GError.domain. - */ -static inline GQuark -pulse_output_quark(void) -{ - return g_quark_from_static_string("pulse_output"); -} - -static struct pulse_data *pulse_new_data(void) -{ - struct pulse_data *ret; - - ret = g_new(struct pulse_data, 1); - - ret->server = NULL; - ret->sink = NULL; - - return ret; -} - -static void pulse_free_data(struct pulse_data *pd) -{ - g_free(pd); -} - -static void * -pulse_init(G_GNUC_UNUSED const struct audio_format *audio_format, - const struct config_param *param, G_GNUC_UNUSED GError **error) -{ - struct pulse_data *pd; - - pd = pulse_new_data(); - pd->name = config_get_block_string(param, "name", "mpd_pulse"); - pd->server = config_get_block_string(param, "server", NULL); - pd->sink = config_get_block_string(param, "sink", NULL); - - return pd; -} - -static void pulse_finish(void *data) -{ - struct pulse_data *pd = data; - - pulse_free_data(pd); -} - -static bool pulse_test_default_device(void) -{ - pa_simple *s; - pa_sample_spec ss; - int error; - - ss.format = PA_SAMPLE_S16NE; - ss.rate = 44100; - ss.channels = 2; - - s = pa_simple_new(NULL, MPD_PULSE_NAME, PA_STREAM_PLAYBACK, NULL, - MPD_PULSE_NAME, &ss, NULL, NULL, &error); - if (!s) { - g_message("Cannot connect to default PulseAudio server: %s\n", - pa_strerror(error)); - return false; - } - - pa_simple_free(s); - - return true; -} - -static bool -pulse_open(void *data, struct audio_format *audio_format, GError **error_r) -{ - struct pulse_data *pd = data; - pa_sample_spec ss; - int error; - - /* MPD doesn't support the other pulseaudio sample formats, so - we just force MPD to send us everything as 16 bit */ - audio_format->bits = 16; - - ss.format = PA_SAMPLE_S16NE; - ss.rate = audio_format->sample_rate; - ss.channels = audio_format->channels; - - pd->s = pa_simple_new(pd->server, MPD_PULSE_NAME, PA_STREAM_PLAYBACK, - pd->sink, pd->name, - &ss, NULL, NULL, - &error); - if (!pd->s) { - g_set_error(error_r, pulse_output_quark(), error, - "Cannot connect to PulseAudio server: %s", - pa_strerror(error)); - return false; - } - - return true; -} - -static void pulse_cancel(void *data) -{ - struct pulse_data *pd = data; - int error; - - if (pa_simple_flush(pd->s, &error) < 0) - g_warning("Flush failed in PulseAudio output \"%s\": %s\n", - pd->name, pa_strerror(error)); -} - -static void pulse_close(void *data) -{ - struct pulse_data *pd = data; - - pa_simple_drain(pd->s, NULL); - pa_simple_free(pd->s); -} - -static size_t -pulse_play(void *data, const void *chunk, size_t size, GError **error_r) -{ - struct pulse_data *pd = data; - int error; - - if (pa_simple_write(pd->s, chunk, size, &error) < 0) { - g_set_error(error_r, pulse_output_quark(), error, - "%s", pa_strerror(error)); - return 0; - } - - return size; -} - -const struct audio_output_plugin pulse_plugin = { - .name = "pulse", - .test_default_device = pulse_test_default_device, - .init = pulse_init, - .finish = pulse_finish, - .open = pulse_open, - .play = pulse_play, - .cancel = pulse_cancel, - .close = pulse_close, - .mixer_plugin = &pulse_mixer, -}; diff --git a/src/output/recorder_output_plugin.c b/src/output/recorder_output_plugin.c new file mode 100644 index 000000000..413e5d0d1 --- /dev/null +++ b/src/output/recorder_output_plugin.c @@ -0,0 +1,214 @@ +/* + * Copyright (C) 2003-2009 The Music Player Daemon Project + * http://www.musicpd.org + * + * This program 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. + * + * This program 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 this program; if not, write to the Free Software Foundation, Inc., + * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. + */ + +#include "output_api.h" +#include "encoder_plugin.h" +#include "encoder_list.h" + +#include <assert.h> +#include <sys/types.h> +#include <sys/stat.h> +#include <fcntl.h> +#include <unistd.h> +#include <errno.h> + +#undef G_LOG_DOMAIN +#define G_LOG_DOMAIN "recorder" + +struct recorder_output { + /** + * The configured encoder plugin. + */ + struct encoder *encoder; + + /** + * The destination file name. + */ + const char *path; + + /** + * The destination file descriptor. + */ + int fd; + + /** + * The buffer for encoder_read(). + */ + char buffer[32768]; +}; + +/** + * The quark used for GError.domain. + */ +static inline GQuark +recorder_output_quark(void) +{ + return g_quark_from_static_string("recorder_output"); +} + +static void * +recorder_output_init(G_GNUC_UNUSED const struct audio_format *audio_format, + const struct config_param *param, GError **error_r) +{ + struct recorder_output *recorder = g_new(struct recorder_output, 1); + const char *encoder_name; + const struct encoder_plugin *encoder_plugin; + + /* read configuration */ + + encoder_name = config_get_block_string(param, "encoder", "vorbis"); + encoder_plugin = encoder_plugin_get(encoder_name); + if (encoder_plugin == NULL) { + g_set_error(error_r, recorder_output_quark(), 0, + "No such encoder: %s", encoder_name); + return NULL; + } + + recorder->path = config_get_block_string(param, "path", NULL); + if (recorder->path == NULL) { + g_set_error(error_r, recorder_output_quark(), 0, + "'path' not configured"); + return NULL; + } + + /* initialize encoder */ + + recorder->encoder = encoder_init(encoder_plugin, param, error_r); + if (recorder->encoder == NULL) + return NULL; + + return recorder; +} + +static void +recorder_output_finish(void *data) +{ + struct recorder_output *recorder = data; + + encoder_finish(recorder->encoder); + g_free(recorder); +} + +/** + * Writes pending data from the encoder to the output file. + */ +static bool +recorder_output_encoder_to_file(struct recorder_output *recorder, + GError **error_r) +{ + size_t size = 0, position, nbytes; + + assert(recorder->fd >= 0); + + /* read from the encoder */ + + size = encoder_read(recorder->encoder, recorder->buffer, + sizeof(recorder->buffer)); + if (size == 0) + return true; + + /* write everything into the file */ + + position = 0; + while (true) { + nbytes = write(recorder->fd, recorder->buffer + position, + size - position); + if (nbytes > 0) { + position += (size_t)nbytes; + if (position >= size) + return true; + } else if (nbytes == 0) { + /* shouldn't happen for files */ + g_set_error(error_r, recorder_output_quark(), 0, + "write() returned 0"); + return false; + } else if (errno != EINTR) { + g_set_error(error_r, recorder_output_quark(), 0, + "Failed to write to '%s': %s", + recorder->path, g_strerror(errno)); + return false; + } + } +} + +static bool +recorder_output_open(void *data, struct audio_format *audio_format, + GError **error_r) +{ + struct recorder_output *recorder = data; + bool success; + + /* create the output file */ + + recorder->fd = creat(recorder->path, 0666); + if (recorder->fd < 0) { + g_set_error(error_r, recorder_output_quark(), 0, + "Failed to create '%s': %s", + recorder->path, g_strerror(errno)); + return false; + } + + /* open the encoder */ + + success = encoder_open(recorder->encoder, audio_format, error_r); + if (!success) { + close(recorder->fd); + unlink(recorder->path); + return false; + } + + return true; +} + +static void +recorder_output_close(void *data) +{ + struct recorder_output *recorder = data; + + /* flush the encoder and write the rest to the file */ + + if (encoder_flush(recorder->encoder, NULL)) + recorder_output_encoder_to_file(recorder, NULL); + + /* now really close everything */ + + encoder_close(recorder->encoder); + + close(recorder->fd); +} + +static size_t +recorder_output_play(void *data, const void *chunk, size_t size, + GError **error_r) +{ + struct recorder_output *recorder = data; + + return encoder_write(recorder->encoder, chunk, size, error_r) && + recorder_output_encoder_to_file(recorder, error_r) + ? size : 0; +} + +const struct audio_output_plugin recorder_output_plugin = { + .name = "recorder", + .init = recorder_output_init, + .finish = recorder_output_finish, + .open = recorder_output_open, + .close = recorder_output_close, + .play = recorder_output_play, +}; diff --git a/src/output/shout_plugin.c b/src/output/shout_plugin.c index 4412d26ff..da90efd2d 100644 --- a/src/output/shout_plugin.c +++ b/src/output/shout_plugin.c @@ -126,6 +126,13 @@ my_shout_init_driver(const struct audio_format *audio_format, struct block_param *block_param; int public; + if (audio_format == NULL || + !audio_format_fully_defined(audio_format)) { + g_set_error(error, shout_output_quark(), 0, + "Need full audio format specification"); + return NULL; + } + sd = new_shout_data(); if (shout_init_count == 0) @@ -191,8 +198,6 @@ my_shout_init_driver(const struct audio_format *audio_format, } } - check_block_param("format"); - encoding = config_get_block_string(param, "encoding", "ogg"); encoder_plugin = shout_encoder_plugin_get(encoding); if (encoder_plugin == NULL) { @@ -471,10 +476,10 @@ shout_tag_to_metadata(const struct tag *tag, char *dest, size_t size) for (unsigned i = 0; i < tag->num_items; i++) { switch (tag->items[i]->type) { - case TAG_ITEM_ARTIST: + case TAG_ARTIST: strncpy(artist, tag->items[i]->value, size); break; - case TAG_ITEM_TITLE: + case TAG_TITLE: strncpy(title, tag->items[i]->value, size); break; |