/* * ALSA 0.9.x-1.x audio output driver * * Copyright (C) 2004 Alex Beregszaszi * Zsolt Barat * * modified for real ALSA 0.9.0 support by Zsolt Barat * additional AC-3 passthrough support by Andy Lo A Foe * 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 #include #include #include #include #include #include #include "config.h" #include "mpvcore/options.h" #include "mpvcore/m_option.h" #include "mpvcore/mp_msg.h" #define ALSA_PCM_NEW_HW_PARAMS_API #define ALSA_PCM_NEW_SW_PARAMS_API #include #include "ao.h" #include "audio/format.h" #include "audio/reorder_ch.h" struct priv { snd_pcm_t *alsa; snd_pcm_format_t alsa_fmt; int can_pause; snd_pcm_sframes_t prepause_frames; float delay_before_pause; int buffersize; // in frames int outburst; // in frames int cfg_block; char *cfg_device; char *cfg_mixer_device; char *cfg_mixer_name; int cfg_mixer_index; }; #define BUFFER_TIME 500000 // 0.5 s #define FRAGCOUNT 16 #define CHECK_ALSA_ERROR(message) \ do { \ if (err < 0) { \ MP_ERR(ao, "%s: %s\n", (message), snd_strerror(err)); \ goto alsa_error; \ } \ } while (0) static float get_delay(struct ao *ao); static void uninit(struct ao *ao, bool immed); 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, "alsa-lib: %s:%i:(%s) %s: %s\n", file, line, function, tmp, snd_strerror(err)); } else { mp_msg(MSGT_AO, MSGL_ERR, "alsa-lib: %s:%i:(%s) %s\n", file, line, function, tmp); } } /* to set/get/query special features/parameters */ static int control(struct ao *ao, enum aocontrol cmd, void *arg) { struct priv *p = ao->priv; snd_mixer_t *handle = NULL; switch (cmd) { case AOCONTROL_GET_MUTE: case AOCONTROL_SET_MUTE: case AOCONTROL_GET_VOLUME: case AOCONTROL_SET_VOLUME: { int err; snd_mixer_elem_t *elem; snd_mixer_selem_id_t *sid; long pmin, pmax; long get_vol, set_vol; float f_multi; if (AF_FORMAT_IS_IEC61937(ao->format)) return CONTROL_TRUE; //allocate simple id snd_mixer_selem_id_alloca(&sid); //sets simple-mixer index and name snd_mixer_selem_id_set_index(sid, p->cfg_mixer_index); snd_mixer_selem_id_set_name(sid, p->cfg_mixer_name); err = snd_mixer_open(&handle, 0); CHECK_ALSA_ERROR("Mixer open error"); err = snd_mixer_attach(handle, p->cfg_mixer_device); CHECK_ALSA_ERROR("Mixer attach error"); err = snd_mixer_selem_register(handle, NULL, NULL); CHECK_ALSA_ERROR("Mixer register error"); err = snd_mixer_load(handle); CHECK_ALSA_ERROR("Mixer load error"); elem = snd_mixer_find_selem(handle, sid); if (!elem) { MP_VERBOSE(ao, "Unable to find simple control '%s',%i.\n", snd_mixer_selem_id_get_name(sid), snd_mixer_selem_id_get_index(sid)); goto alsa_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 err = snd_mixer_selem_set_playback_volume (elem, SND_MIXER_SCHN_FRONT_LEFT, set_vol); CHECK_ALSA_ERROR("Error setting left channel"); MP_DBG(ao, "left=%li, ", set_vol); set_vol = vol->right / f_multi + pmin + 0.5; err = snd_mixer_selem_set_playback_volume (elem, SND_MIXER_SCHN_FRONT_RIGHT, set_vol); CHECK_ALSA_ERROR("Error setting right channel"); MP_DBG(ao, "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_DBG(ao, "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 alsa_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 alsa_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; } } //end switch return CONTROL_UNKNOWN; alsa_error: if (handle) snd_mixer_close(handle); return CONTROL_ERROR; } static const int mp_to_alsa_format[][2] = { {AF_FORMAT_S8, SND_PCM_FORMAT_S8}, {AF_FORMAT_U8, SND_PCM_FORMAT_U8}, {AF_FORMAT_U16_LE, SND_PCM_FORMAT_U16_LE}, {AF_FORMAT_U16_BE, SND_PCM_FORMAT_U16_BE}, {AF_FORMAT_S16_LE, SND_PCM_FORMAT_S16_LE}, {AF_FORMAT_S16_BE, SND_PCM_FORMAT_S16_BE}, {AF_FORMAT_U32_LE, SND_PCM_FORMAT_U32_LE}, {AF_FORMAT_U32_BE, SND_PCM_FORMAT_U32_BE}, {AF_FORMAT_S32_LE, SND_PCM_FORMAT_S32_LE}, {AF_FORMAT_S32_BE, SND_PCM_FORMAT_S32_BE}, {AF_FORMAT_U24_LE, SND_PCM_FORMAT_U24_3LE}, {AF_FORMAT_U24_BE, SND_PCM_FORMAT_U24_3BE}, {AF_FORMAT_S24_LE, SND_PCM_FORMAT_S24_3LE}, {AF_FORMAT_S24_BE, SND_PCM_FORMAT_S24_3BE}, {AF_FORMAT_FLOAT_LE, SND_PCM_FORMAT_FLOAT_LE}, {AF_FORMAT_FLOAT_BE, SND_PCM_FORMAT_FLOAT_BE}, {AF_FORMAT_AC3_LE, SND_PCM_FORMAT_S16_LE}, {AF_FORMAT_AC3_BE, SND_PCM_FORMAT_S16_BE}, {AF_FORMAT_IEC61937_LE, SND_PCM_FORMAT_S16_LE}, {AF_FORMAT_IEC61937_BE, SND_PCM_FORMAT_S16_BE}, {AF_FORMAT_MPEG2, SND_PCM_FORMAT_MPEG}, {AF_FORMAT_UNKNOWN, SND_PCM_FORMAT_UNKNOWN}, }; static int find_alsa_format(int af_format) { af_format = af_fmt_from_planar(af_format); for (int n = 0; mp_to_alsa_format[n][0] != AF_FORMAT_UNKNOWN; n++) { if (mp_to_alsa_format[n][0] == af_format) return mp_to_alsa_format[n][1]; } return SND_PCM_FORMAT_UNKNOWN; } // Lists device names and their implied channel map. // The second item must be resolvable with mp_chmap_from_str(). // Source: http://www.alsa-project.org/main/index.php/DeviceNames // (Speaker names are slightly different from mpv's.) static const char *device_channel_layouts[][2] = { {"default", "fc"}, {"default", "fl-fr"}, {"rear", "bl-br"}, {"center_lfe", "fc-lfe"}, {"side", "sl-sr"}, {"surround40", "fl-fr-bl-br"}, {"surround50", "fl-fr-bl-br-fc"}, {"surround41", "fl-fr-bl-br-lfe"}, {"surround51", "fl-fr-bl-br-fc-lfe"}, {"surround71", "fl-fr-bl-br-fc-lfe-sl-sr"}, }; #define ARRAY_LEN(x) (sizeof(x) / sizeof((x)[0])) #define NUM_ALSA_CHMAPS ARRAY_LEN(device_channel_layouts) static const char *select_chmap(struct ao *ao) { struct mp_chmap_sel sel = {0}; struct mp_chmap maps[NUM_ALSA_CHMAPS]; for (int n = 0; n < NUM_ALSA_CHMAPS; n++) { mp_chmap_from_str(&maps[n], bstr0(device_channel_layouts[n][1])); mp_chmap_sel_add_map(&sel, &maps[n]); }; if (!ao_chmap_sel_adjust(ao, &sel, &ao->channels)) return NULL; for (int n = 0; n < NUM_ALSA_CHMAPS; n++) { if (mp_chmap_equals(&ao->channels, &maps[n])) return device_channel_layouts[n][0]; } char *name = mp_chmap_to_str(&ao->channels); MP_ERR(ao, "channel layout %s (%d ch) not supported.\n", name, ao->channels.num); talloc_free(name); return "default"; } static int map_iec958_srate(int srate) { switch (srate) { case 44100: return IEC958_AES3_CON_FS_44100; case 48000: return IEC958_AES3_CON_FS_48000; case 32000: return IEC958_AES3_CON_FS_32000; case 22050: return IEC958_AES3_CON_FS_22050; case 24000: return IEC958_AES3_CON_FS_24000; case 88200: return IEC958_AES3_CON_FS_88200; case 768000: return IEC958_AES3_CON_FS_768000; case 96000: return IEC958_AES3_CON_FS_96000; case 176400: return IEC958_AES3_CON_FS_176400; case 192000: return IEC958_AES3_CON_FS_192000; default: return IEC958_AES3_CON_FS_NOTID; } } static int try_open_device(struct ao *ao, const char *device, int open_mode) { struct priv *p = ao->priv; if (AF_FORMAT_IS_IEC61937(ao->format)) { void *tmp = talloc_new(NULL); /* to set the non-audio bit, use AES0=6 */ char *params = talloc_asprintf(tmp, "AES0=%d,AES1=%d,AES2=0,AES3=%d", IEC958_AES0_NONAUDIO | IEC958_AES0_PRO_EMPHASIS_NONE, IEC958_AES1_CON_ORIGINAL | IEC958_AES1_CON_PCM_CODER, map_iec958_srate(ao->samplerate)); const char *ac3_device = device; int len = strlen(device); char *end = strchr(device, ':'); if (!end) { /* no existing parameters: add it behind device name */ ac3_device = talloc_asprintf(tmp, "%s:%s", device, params); } else if (end[1] == '\0') { /* ":" but no parameters */ ac3_device = talloc_asprintf(tmp, "%s%s", device, params); } else if (end[1] == '{' && device[len - 1] == '}') { /* parameters in config syntax: add it inside the { } block */ ac3_device = talloc_asprintf(tmp, "%.*s %s}", len - 1, device, params); } else { /* a simple list of parameters: add it at the end of the list */ ac3_device = talloc_asprintf(tmp, "%s,%s", device, params); } int err = snd_pcm_open (&p->alsa, ac3_device, SND_PCM_STREAM_PLAYBACK, open_mode); talloc_free(tmp); if (!err) return 0; } return snd_pcm_open(&p->alsa, device, SND_PCM_STREAM_PLAYBACK, open_mode); } /* open & setup audio device return: 0=success -1=fail */ static int init(struct ao *ao) { int err; snd_pcm_uframes_t chunk_size; snd_pcm_uframes_t bufsize; snd_pcm_uframes_t boundary; struct priv *p = ao->priv; p->prepause_frames = 0; p->delay_before_pause = 0; /* 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' */ const char *device; if (AF_FORMAT_IS_IEC61937(ao->format)) { device = "iec958"; MP_VERBOSE(ao, "playing AC3/iec61937/iec958, %i channels\n", ao->channels.num); } else { device = select_chmap(ao); if (strcmp(device, "default") != 0 && ao->format == AF_FORMAT_FLOAT) { // hack - use the converter plugin (why the heck?) device = talloc_asprintf(ao, "plug:%s", device); } } if (p->cfg_device && p->cfg_device[0]) device = p->cfg_device; MP_VERBOSE(ao, "using device: %s\n", device); p->can_pause = 1; MP_VERBOSE(ao, "using ALSA version: %s\n", snd_asoundlib_version()); snd_lib_error_set_handler(alsa_error_handler); int open_mode = p->cfg_block ? 0 : SND_PCM_NONBLOCK; //modes = 0, SND_PCM_NONBLOCK, SND_PCM_ASYNC err = try_open_device(ao, device, open_mode); if (err < 0) { if (err != -EBUSY && !p->cfg_block) { MP_WARN(ao, "Open in nonblock-mode " "failed, trying to open in block-mode.\n"); err = try_open_device(ao, device, 0); } CHECK_ALSA_ERROR("Playback open error"); } err = snd_pcm_nonblock(p->alsa, 0); if (err < 0) { MP_ERR(ao, "Error setting block-mode: %s.\n", snd_strerror(err)); } else { MP_VERBOSE(ao, "pcm opened in blocking mode\n"); } snd_pcm_hw_params_t *alsa_hwparams; snd_pcm_sw_params_t *alsa_swparams; snd_pcm_hw_params_alloca(&alsa_hwparams); snd_pcm_sw_params_alloca(&alsa_swparams); // setting hw-parameters err = snd_pcm_hw_params_any(p->alsa, alsa_hwparams); CHECK_ALSA_ERROR("Unable to get initial parameters"); p->alsa_fmt = find_alsa_format(ao->format); if (p->alsa_fmt == SND_PCM_FORMAT_UNKNOWN) { p->alsa_fmt = SND_PCM_FORMAT_S16; ao->format = AF_FORMAT_S16; } err = snd_pcm_hw_params_test_format(p->alsa, alsa_hwparams, p->alsa_fmt); if (err < 0) { MP_INFO(ao, "Format %s is not supported by hardware, trying default.\n", af_fmt_to_str(ao->format)); p->alsa_fmt = SND_PCM_FORMAT_S16_LE; if (AF_FORMAT_IS_AC3(ao->format)) ao->format = AF_FORMAT_AC3_LE; else if (AF_FORMAT_IS_IEC61937(ao->format)) ao->format = AF_FORMAT_IEC61937_LE; else ao->format = AF_FORMAT_S16_LE; } err = snd_pcm_hw_params_set_format(p->alsa, alsa_hwparams, p->alsa_fmt); CHECK_ALSA_ERROR("Unable to set format"); snd_pcm_access_t access = af_fmt_is_planar(ao->format) ? SND_PCM_ACCESS_RW_NONINTERLEAVED : SND_PCM_ACCESS_RW_INTERLEAVED; err = snd_pcm_hw_params_set_access(p->alsa, alsa_hwparams, access); if (err < 0 && af_fmt_is_planar(ao->format)) { ao->format = af_fmt_from_planar(ao->format); access = SND_PCM_ACCESS_RW_INTERLEAVED; err = snd_pcm_hw_params_set_access(p->alsa, alsa_hwparams, access); } CHECK_ALSA_ERROR("Unable to set access type"); int num_channels = ao->channels.num; err = snd_pcm_hw_params_set_channels_near (p->alsa, alsa_hwparams, &num_channels); CHECK_ALSA_ERROR("Unable to set channels"); if (num_channels != ao->channels.num) { MP_ERR(ao, "Couldn't get requested number of channels.\n"); mp_chmap_from_channels_alsa(&ao->channels, num_channels); } /* workaround for buggy rate plugin (should be fixed in ALSA 1.0.11) prefer our own resampler, since that allows users to choose the resampler, even per file if desired */ err = snd_pcm_hw_params_set_rate_resample(p->alsa, alsa_hwparams, 0); CHECK_ALSA_ERROR("Unable to disable resampling"); err = snd_pcm_hw_params_set_rate_near (p->alsa, alsa_hwparams, &ao->samplerate, NULL); CHECK_ALSA_ERROR("Unable to set samplerate-2"); err = snd_pcm_hw_params_set_buffer_time_near (p->alsa, alsa_hwparams, &(unsigned int){BUFFER_TIME}, NULL); CHECK_ALSA_ERROR("Unable to set buffer time near"); err = snd_pcm_hw_params_set_periods_near (p->alsa, alsa_hwparams, &(unsigned int){FRAGCOUNT}, NULL); CHECK_ALSA_ERROR("Unable to set periods"); /* finally install hardware parameters */ err = snd_pcm_hw_params(p->alsa, alsa_hwparams); CHECK_ALSA_ERROR("Unable to set hw-parameters"); // end setting hw-params // gets buffersize for control err = snd_pcm_hw_params_get_buffer_size(alsa_hwparams, &bufsize); CHECK_ALSA_ERROR("Unable to get buffersize"); p->buffersize = bufsize; MP_VERBOSE(ao, "got buffersize=%i samples\n", p->buffersize); err = snd_pcm_hw_params_get_period_size(alsa_hwparams, &chunk_size, NULL); CHECK_ALSA_ERROR("Unable to get period size"); MP_VERBOSE(ao, "got period size %li\n", chunk_size); p->outburst = chunk_size; /* setting software parameters */ err = snd_pcm_sw_params_current(p->alsa, alsa_swparams); CHECK_ALSA_ERROR("Unable to get sw-parameters"); err = snd_pcm_sw_params_get_boundary(alsa_swparams, &boundary); CHECK_ALSA_ERROR("Unable to get boundary"); /* start playing when one period has been written */ err = snd_pcm_sw_params_set_start_threshold (p->alsa, alsa_swparams, chunk_size); CHECK_ALSA_ERROR("Unable to set start threshold"); /* disable underrun reporting */ err = snd_pcm_sw_params_set_stop_threshold (p->alsa, alsa_swparams, boundary); CHECK_ALSA_ERROR("Unable to set stop threshold"); /* play silence when there is an underrun */ err = snd_pcm_sw_params_set_silence_size (p->alsa, alsa_swparams, boundary); CHECK_ALSA_ERROR("Unable to set silence size"); err = snd_pcm_sw_params(p->alsa, alsa_swparams); CHECK_ALSA_ERROR("Unable to get sw-parameters"); /* end setting sw-params */ p->can_pause = snd_pcm_hw_params_can_pause(alsa_hwparams); MP_VERBOSE(ao, "opened: %d Hz/%d channels/%d bps/%d samples buffer/%s\n", ao->samplerate, ao->channels.num, af_fmt2bits(ao->format), p->buffersize, snd_pcm_format_description(p->alsa_fmt)); return 0; alsa_error: uninit(ao, true); return -1; } // end init /* close audio device */ static void uninit(struct ao *ao, bool immed) { struct priv *p = ao->priv; if (p->alsa) { int err; if (!immed) snd_pcm_drain(p->alsa); err = snd_pcm_close(p->alsa); CHECK_ALSA_ERROR("pcm close error"); MP_VERBOSE(ao, "uninit: pcm closed\n"); } alsa_error: p->alsa = NULL; snd_lib_error_set_handler(NULL); } static void audio_pause(struct ao *ao) { struct priv *p = ao->priv; int err; if (p->can_pause) { p->delay_before_pause = get_delay(ao); err = snd_pcm_pause(p->alsa, 1); CHECK_ALSA_ERROR("pcm pause error"); } else { MP_VERBOSE(ao, "pause not supported by hardware\n"); if (snd_pcm_delay(p->alsa, &p->prepause_frames) < 0 || p->prepause_frames < 0) p->prepause_frames = 0; p->delay_before_pause = p->prepause_frames / (float)ao->samplerate; err = snd_pcm_drop(p->alsa); CHECK_ALSA_ERROR("pcm drop error"); } alsa_error: ; } static void audio_resume(struct ao *ao) { struct priv *p = ao->priv; int err; if (snd_pcm_state(p->alsa) == SND_PCM_STATE_SUSPENDED) { MP_INFO(ao, "PCM in suspend mode, trying to resume.\n"); while ((err = snd_pcm_resume(p->alsa)) == -EAGAIN) sleep(1); } if (p->can_pause) { err = snd_pcm_pause(p->alsa, 0); CHECK_ALSA_ERROR("pcm resume error"); } else { MP_VERBOSE(ao, "resume not supported by hardware\n"); err = snd_pcm_prepare(p->alsa); CHECK_ALSA_ERROR("pcm prepare error"); if (p->prepause_frames) ao_play_silence(ao, p->prepause_frames); } alsa_error: ; } /* stop playing and empty buffers (for seeking/pause) */ static void reset(struct ao *ao) { struct priv *p = ao->priv; int err; p->prepause_frames = 0; p->delay_before_pause = 0; err = snd_pcm_drop(p->alsa); CHECK_ALSA_ERROR("pcm prepare error"); err = snd_pcm_prepare(p->alsa); CHECK_ALSA_ERROR("pcm prepare error"); alsa_error: ; } static int play(struct ao *ao, void **data, int samples, int flags) { struct priv *p = ao->priv; snd_pcm_sframes_t res = 0; if (!(flags & AOPLAY_FINAL_CHUNK)) samples = samples / p->outburst * p->outburst; if (!p->alsa) { MP_ERR(ao, "Device configuration error."); return -1; } if (samples == 0) return 0; do { if (af_fmt_is_planar(ao->format)) { res = snd_pcm_writen(p->alsa, data, samples); } else { res = snd_pcm_writei(p->alsa, data[0], samples); } if (res == -EINTR) { /* nothing to do */ res = 0; } else if (res == -ESTRPIPE) { /* suspend */ MP_INFO(ao, "PCM in suspend mode, trying to resume.\n"); while ((res = snd_pcm_resume(p->alsa)) == -EAGAIN) sleep(1); } if (res < 0) { MP_ERR(ao, "Write error: %s\n", snd_strerror(res)); res = snd_pcm_prepare(p->alsa); int err = res; CHECK_ALSA_ERROR("pcm prepare error"); res = 0; } } while (res == 0); return res < 0 ? -1 : res; alsa_error: return -1; } /* how many byes are free in the buffer */ static int get_space(struct ao *ao) { struct priv *p = ao->priv; snd_pcm_status_t *status; int err; snd_pcm_status_alloca(&status); err = snd_pcm_status(p->alsa, status); CHECK_ALSA_ERROR("cannot get pcm status"); unsigned space = snd_pcm_status_get_avail(status); if (space > p->buffersize) // Buffer underrun? space = p->buffersize; return space; alsa_error: return 0; } /* delay in seconds between first and last sample in buffer */ static float get_delay(struct ao *ao) { struct priv *p = ao->priv; if (p->alsa) { snd_pcm_sframes_t delay; if (snd_pcm_state(p->alsa) == SND_PCM_STATE_PAUSED) return p->delay_before_pause; if (snd_pcm_delay(p->alsa, &delay) < 0) return 0; if (delay < 0) { /* underrun - move the application pointer forward to catch up */ snd_pcm_forward(p->alsa, -delay); delay = 0; } return (float)delay / (float)ao->samplerate; } else return 0; } #define OPT_BASE_STRUCT struct priv const struct ao_driver audio_out_alsa = { .description = "ALSA-0.9.x-1.x audio output", .name = "alsa", .init = init, .uninit = uninit, .control = control, .get_space = get_space, .play = play, .get_delay = get_delay, .pause = audio_pause, .resume = audio_resume, .reset = reset, .priv_size = sizeof(struct priv), .priv_defaults = &(const struct priv) { .cfg_block = 1, .cfg_mixer_device = "default", .cfg_mixer_name = "Master", .cfg_mixer_index = 0, }, .options = (const struct m_option[]) { OPT_STRING("device", cfg_device, 0), OPT_FLAG("block", cfg_block, 0), OPT_STRING("mixer-device", cfg_mixer_device, 0), OPT_STRING("mixer-name", cfg_mixer_name, 0), OPT_INTRANGE("mixer-index", cfg_mixer_index, 0, 0, 99), {0} }, };