/* * 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 mpv. * * mpv 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. * * mpv 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 mpv. If not, see . */ #include #include #include #include #include #include #include "config.h" #include "options/options.h" #include "options/m_option.h" #include "common/msg.h" #include "osdep/endian.h" #include #define HAVE_CHMAP_API \ (defined(SND_CHMAP_API_VERSION) && SND_CHMAP_API_VERSION >= (1 << 16)) #include "ao.h" #include "internal.h" #include "audio/format.h" struct priv { snd_pcm_t *alsa; snd_pcm_format_t alsa_fmt; int can_pause; snd_pcm_sframes_t prepause_frames; double delay_before_pause; int buffersize; // in frames int outburst; // in frames char *cfg_device; char *cfg_mixer_device; char *cfg_mixer_name; int cfg_mixer_index; int cfg_resample; int cfg_ni; int cfg_ignore_chmap; }; #define BUFFER_TIME 250000 // 250ms #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) #define CHECK_ALSA_WARN(message) \ do { \ if (err < 0) \ MP_WARN(ao, "%s: %s\n", (message), snd_strerror(err)); \ } while (0) 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_SPECIAL(ao->format)) return CONTROL_FALSE; snd_mixer_selem_id_alloca(&sid); 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; 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_U8, SND_PCM_FORMAT_U8}, {AF_FORMAT_S16, SND_PCM_FORMAT_S16}, {AF_FORMAT_S32, SND_PCM_FORMAT_S32}, {AF_FORMAT_S24, MP_SELECT_LE_BE(SND_PCM_FORMAT_S24_3LE, SND_PCM_FORMAT_S24_3BE)}, {AF_FORMAT_FLOAT, SND_PCM_FORMAT_FLOAT}, {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; } #if HAVE_CHMAP_API static const int alsa_to_mp_channels[][2] = { {SND_CHMAP_FL, MP_SP(FL)}, {SND_CHMAP_FR, MP_SP(FR)}, {SND_CHMAP_RL, MP_SP(BL)}, {SND_CHMAP_RR, MP_SP(BR)}, {SND_CHMAP_FC, MP_SP(FC)}, {SND_CHMAP_LFE, MP_SP(LFE)}, {SND_CHMAP_SL, MP_SP(SL)}, {SND_CHMAP_SR, MP_SP(SR)}, {SND_CHMAP_RC, MP_SP(BC)}, {SND_CHMAP_FLC, MP_SP(FLC)}, {SND_CHMAP_FRC, MP_SP(FRC)}, {SND_CHMAP_FLW, MP_SP(WL)}, {SND_CHMAP_FRW, MP_SP(WR)}, {SND_CHMAP_TC, MP_SP(TC)}, {SND_CHMAP_TFL, MP_SP(TFL)}, {SND_CHMAP_TFR, MP_SP(TFR)}, {SND_CHMAP_TFC, MP_SP(TFC)}, {SND_CHMAP_TRL, MP_SP(TBL)}, {SND_CHMAP_TRR, MP_SP(TBR)}, {SND_CHMAP_TRC, MP_SP(TBC)}, {SND_CHMAP_RRC, MP_SP(SDR)}, {SND_CHMAP_RLC, MP_SP(SDL)}, {SND_CHMAP_MONO, MP_SP(FC)}, {SND_CHMAP_NA, MP_SPEAKER_ID_NA}, {SND_CHMAP_LAST, MP_SPEAKER_ID_COUNT} }; static int find_mp_channel(int alsa_channel) { for (int i = 0; alsa_to_mp_channels[i][1] != MP_SPEAKER_ID_COUNT; i++) { if (alsa_to_mp_channels[i][0] == alsa_channel) return alsa_to_mp_channels[i][1]; } return MP_SPEAKER_ID_COUNT; } static int find_alsa_channel(int mp_channel) { for (int i = 0; alsa_to_mp_channels[i][1] != MP_SPEAKER_ID_COUNT; i++) { if (alsa_to_mp_channels[i][1] == mp_channel) return alsa_to_mp_channels[i][0]; } return SND_CHMAP_UNKNOWN; } static int mp_chmap_from_alsa(struct mp_chmap *dst, snd_pcm_chmap_t *src) { *dst = (struct mp_chmap) {0}; if (src->channels > MP_NUM_CHANNELS) return -1; dst->num = src->channels; for (int c = 0; c < dst->num; c++) dst->speaker[c] = find_mp_channel(src->pos[c]); return 0; } static bool query_chmaps(struct ao *ao, struct mp_chmap *chmap) { struct priv *p = ao->priv; struct mp_chmap_sel chmap_sel = {.tmp = p}; snd_pcm_chmap_query_t **maps = snd_pcm_query_chmaps(p->alsa); if (!maps) return false; for (int i = 0; maps[i] != NULL; i++) { struct mp_chmap entry; mp_chmap_from_alsa(&entry, &maps[i]->map); if (mp_chmap_is_valid(&entry)) { if (maps[i]->type == SND_CHMAP_TYPE_VAR) mp_chmap_reorder_norm(&entry); MP_VERBOSE(ao, "Got supported channel map: %s (type %s)\n", mp_chmap_to_str(&entry), snd_pcm_chmap_type_name(maps[i]->type)); mp_chmap_sel_add_map(&chmap_sel, &entry); } else { char tmp[128]; if (snd_pcm_chmap_print(&maps[i]->map, sizeof(tmp), tmp) > 0) MP_VERBOSE(ao, "skipping unknown ALSA channel map: %s\n", tmp); } } snd_pcm_free_chmaps(maps); return ao_chmap_sel_adjust(ao, &chmap_sel, chmap); } #else /* HAVE_CHMAP_API */ static bool query_chmaps(struct ao *ao, struct mp_chmap *chmap) { return false; } #endif /* else HAVE_CHMAP_API */ 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; } } // ALSA device strings can have parameters. They are usually appended to the // device name. Since there can be various forms, and we (sometimes) want to // append them to unknown device strings, which possibly already include params. static char *append_params(void *ta_parent, const char *device, const char *p) { if (!p || !p[0]) return talloc_strdup(ta_parent, device); int len = strlen(device); char *end = strchr(device, ':'); if (!end) { /* no existing parameters: add it behind device name */ return talloc_asprintf(ta_parent, "%s:%s", device, p); } else if (end[1] == '\0') { /* ":" but no parameters */ return talloc_asprintf(ta_parent, "%s%s", device, p); } else if (end[1] == '{' && device[len - 1] == '}') { /* parameters in config syntax: add it inside the { } block */ return talloc_asprintf(ta_parent, "%.*s %s}", len - 1, device, p); } else { /* a simple list of parameters: add it at the end of the list */ return talloc_asprintf(ta_parent, "%s,%s", device, p); } abort(); } static int try_open_device(struct ao *ao, const char *device) { struct priv *p = ao->priv; int err; if (AF_FORMAT_IS_IEC61937(ao->format)) { void *tmp = talloc_new(NULL); 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 = append_params(tmp, device, params); MP_VERBOSE(ao, "opening device '%s' => '%s'\n", device, ac3_device); err = snd_pcm_open(&p->alsa, ac3_device, SND_PCM_STREAM_PLAYBACK, 0); if (err < 0) { // Some spdif-capable devices do not accept the AES0 parameter, // and instead require the iec958 pseudo-device (they will play // noise otherwise). Unfortunately, ALSA gives us no way to map // these devices, so try it for the default device only. bstr dev; bstr_split_tok(bstr0(device), ":", &dev, &(bstr){0}); if (bstr_equals0(dev, "default")) { ac3_device = append_params(tmp, "iec958", params); MP_VERBOSE(ao, "got error %d; opening iec fallback device '%s'\n", err, ac3_device); err = snd_pcm_open (&p->alsa, ac3_device, SND_PCM_STREAM_PLAYBACK, 0); } } talloc_free(tmp); } else { MP_VERBOSE(ao, "opening device '%s'\n", device); err = snd_pcm_open(&p->alsa, device, SND_PCM_STREAM_PLAYBACK, 0); } return err; } static void uninit(struct ao *ao) { struct priv *p = ao->priv; if (p->alsa) { int err; err = snd_pcm_close(p->alsa); CHECK_ALSA_ERROR("pcm close error"); } alsa_error: ; } #define INIT_OK 0 #define INIT_ERROR -1 #define INIT_BRAINDEATH -2 static int init_device(struct ao *ao, bool second_try) { struct priv *p = ao->priv; int err; const char *device = "default"; if (ao->device) device = ao->device; if (p->cfg_device && p->cfg_device[0]) device = p->cfg_device; err = try_open_device(ao, device); CHECK_ALSA_ERROR("Playback open error"); err = snd_pcm_nonblock(p->alsa, 0); CHECK_ALSA_WARN("Unable to set blocking mode"); 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); err = snd_pcm_hw_params_any(p->alsa, alsa_hwparams); CHECK_ALSA_ERROR("Unable to get initial parameters"); if (AF_FORMAT_IS_IEC61937(ao->format)) { if (ao->format == AF_FORMAT_S_MP3) { p->alsa_fmt = SND_PCM_FORMAT_MPEG; } else { p->alsa_fmt = SND_PCM_FORMAT_S16; } } else { 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) { if (AF_FORMAT_IS_IEC61937(ao->format)) CHECK_ALSA_ERROR("Unable to set IEC61937 format"); 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; ao->format = AF_FORMAT_S16; } 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_FORMAT_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_FORMAT_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"); struct mp_chmap dev_chmap = ao->channels; if (AF_FORMAT_IS_IEC61937(ao->format) || p->cfg_ignore_chmap) { dev_chmap.num = 0; // disable chmap API } else if (dev_chmap.num == 1 && dev_chmap.speaker[0] == MP_SPEAKER_ID_FC) { // As yet another ALSA API inconsistency, mono is not reported correctly. dev_chmap.num = 0; } else if (query_chmaps(ao, &dev_chmap)) { ao->channels = dev_chmap; } else { // Assume only stereo and mono are supported. mp_chmap_from_channels(&ao->channels, MPMIN(2, dev_chmap.num)); dev_chmap.num = 0; } 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 > MP_NUM_CHANNELS) { MP_FATAL(ao, "Too many audio channels (%d).\n", num_channels); goto alsa_error; } if (num_channels != ao->channels.num) { int req = ao->channels.num; mp_chmap_from_channels_alsa(&ao->channels, num_channels); if (!mp_chmap_is_valid(&ao->channels)) mp_chmap_from_channels(&ao->channels, 2); MP_ERR(ao, "Asked for %d channels, got %d - fallback to %s.\n", req, num_channels, mp_chmap_to_str(&ao->channels)); } // Some ALSA drivers have broken delay reporting, so disable the ALSA // resampling plugin by default. if (!p->cfg_resample) { 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_WARN("Unable to set buffer time near"); err = snd_pcm_hw_params_set_periods_near (p->alsa, alsa_hwparams, &(unsigned int){FRAGCOUNT}, NULL); CHECK_ALSA_WARN("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 */ #if HAVE_CHMAP_API if (mp_chmap_is_valid(&dev_chmap)) { snd_pcm_chmap_t *alsa_chmap = calloc(1, sizeof(*alsa_chmap) + sizeof(alsa_chmap->pos[0]) * dev_chmap.num); if (!alsa_chmap) goto alsa_error; alsa_chmap->channels = dev_chmap.num; for (int c = 0; c < dev_chmap.num; c++) alsa_chmap->pos[c] = find_alsa_channel(dev_chmap.speaker[c]); // mpv and ALSA use different conventions for mono if (dev_chmap.num == 1 && dev_chmap.speaker[0] == MP_SP(FC)) alsa_chmap->pos[0] = SND_CHMAP_MONO; char tmp[128]; if (snd_pcm_chmap_print(alsa_chmap, sizeof(tmp), tmp) > 0) MP_VERBOSE(ao, "trying to set ALSA channel map: %s\n", tmp); err = snd_pcm_set_chmap(p->alsa, alsa_chmap); if (err == -ENXIO) { // A device my not be able to set any channel map, even channel maps // that were reported as supported. This is either because the ALSA // device is broken (dmix), or because the driver has only 1 // channel map per channel count, and setting the map is not needed. MP_VERBOSE(ao, "device returned ENXIO when setting channel map %s\n", mp_chmap_to_str(&dev_chmap)); } else { CHECK_ALSA_WARN("Channel map setup failed"); } free(alsa_chmap); } snd_pcm_chmap_t *alsa_chmap = snd_pcm_get_chmap(p->alsa); if (alsa_chmap) { char tmp[128]; if (snd_pcm_chmap_print(alsa_chmap, sizeof(tmp), tmp) > 0) MP_VERBOSE(ao, "channel map reported by ALSA: %s\n", tmp); struct mp_chmap chmap; mp_chmap_from_alsa(&chmap, alsa_chmap); MP_VERBOSE(ao, "which we understand as: %s\n", mp_chmap_to_str(&chmap)); if (p->cfg_ignore_chmap) { MP_VERBOSE(ao, "user set ignore-chmap; ignoring the channel map.\n"); } else if (AF_FORMAT_IS_IEC61937(ao->format)) { MP_VERBOSE(ao, "using spdif passthrough; ignoring the channel map.\n"); } else if (mp_chmap_is_valid(&chmap)) { // Is it one that contains NA channels? struct mp_chmap without_na = chmap; mp_chmap_remove_na(&without_na); if (mp_chmap_is_valid(&without_na) && !mp_chmap_equals(&without_na, &chmap) && !mp_chmap_equals(&chmap, &ao->channels) && !second_try) { // Sometimes, ALSA will advertise certain chmaps, but it's not // possible to set them. This can happen with dmix: as of // alsa 1.0.28, dmix can do stereo only, but advertises the // surround chmaps of the underlying device. In this case, // e.g. setting 6 channels will succeed, but requesting 5.1 // afterwards will fail. Then it will return something like // "FL FR NA NA NA NA" as channel map. This means we would // have to pad stereo output to 6 channels with silence, which // would require lots of extra processing. You can't change // the number of channels to 2 either, because the hw params // are already set! So just fuck it and reopen the device with // the chmap "cleaned out" of NA entries. MP_VERBOSE(ao, "Working around braindead ALSA behavior.\n"); err = snd_pcm_close(p->alsa); p->alsa = NULL; CHECK_ALSA_ERROR("pcm close error"); ao->channels = without_na; return INIT_BRAINDEATH; } if (mp_chmap_equals(&chmap, &ao->channels)) { MP_VERBOSE(ao, "which is what we requested.\n"); } else if (chmap.num == ao->channels.num) { MP_VERBOSE(ao, "using the ALSA channel map.\n"); ao->channels = chmap; } else { MP_WARN(ao, "ALSA channel map conflicts with channel count!\n"); } } else { MP_WARN(ao, "Got unknown channel map from ALSA.\n"); } // mpv and ALSA use different conventions for mono if (ao->channels.num == 1) { MP_VERBOSE(ao, "assuming we actually got MONO from ALSA.\n"); ao->channels.speaker[0] = MP_SP(FC); } free(alsa_chmap); } #endif snd_pcm_uframes_t bufsize; 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); snd_pcm_uframes_t chunk_size; 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"); snd_pcm_uframes_t boundary; 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 set sw-parameters"); /* end setting sw-params */ p->can_pause = snd_pcm_hw_params_can_pause(alsa_hwparams); return INIT_OK; alsa_error: uninit(ao); return INIT_ERROR; } static int init(struct ao *ao) { struct priv *p = ao->priv; if (!p->cfg_ni) ao->format = af_fmt_from_planar(ao->format); MP_VERBOSE(ao, "using ALSA version: %s\n", snd_asoundlib_version()); int r = init_device(ao, false); if (r == INIT_BRAINDEATH) r = init_device(ao, true); // retry with normalized channel layout return r == INIT_OK ? 0 : -1; } static void drain(struct ao *ao) { struct priv *p = ao->priv; snd_pcm_drain(p->alsa); } 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 / p->outburst * p->outburst; alsa_error: return 0; } /* delay in seconds between first and last sample in buffer */ static double get_delay(struct ao *ao) { struct priv *p = ao->priv; 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 delay / (double)ao->samplerate; } static void audio_pause(struct ao *ao) { struct priv *p = ao->priv; int err; if (p->can_pause) { if (snd_pcm_state(p->alsa) == SND_PCM_STATE_RUNNING) { 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 / (double)ao->samplerate; err = snd_pcm_drop(p->alsa); CHECK_ALSA_ERROR("pcm drop error"); } alsa_error: ; } static void resume_device(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); } } static void audio_resume(struct ao *ao) { struct priv *p = ao->priv; int err; resume_device(ao); if (p->can_pause) { if (snd_pcm_state(p->alsa) == SND_PCM_STATE_PAUSED) { 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: ; } 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 (samples == 0) return 0; do { if (AF_FORMAT_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 || res == -EAGAIN) { /* retry */ res = 0; } else if (res == -ENODEV) { MP_WARN(ao, "Device lost, trying to recover...\n"); ao_request_reload(ao); } else if (res < 0) { if (res == -ESTRPIPE) { /* suspend */ resume_device(ao); } else { 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; } #define MAX_POLL_FDS 20 static int audio_wait(struct ao *ao, pthread_mutex_t *lock) { struct priv *p = ao->priv; int err; int num_fds = snd_pcm_poll_descriptors_count(p->alsa); if (num_fds <= 0 || num_fds >= MAX_POLL_FDS) goto alsa_error; struct pollfd fds[MAX_POLL_FDS]; err = snd_pcm_poll_descriptors(p->alsa, fds, num_fds); CHECK_ALSA_ERROR("cannot get pollfds"); while (1) { int r = ao_wait_poll(ao, fds, num_fds, lock); if (r) return r; unsigned short revents; snd_pcm_poll_descriptors_revents(p->alsa, fds, num_fds, &revents); CHECK_ALSA_ERROR("cannot read poll events"); if (revents & POLLERR) return -1; if (revents & POLLOUT) return 0; } return 0; alsa_error: return -1; } static void list_devs(struct ao *ao, struct ao_device_list *list) { void **hints; if (snd_device_name_hint(-1, "pcm", &hints) < 0) return; for (int n = 0; hints[n]; n++) { char *name = snd_device_name_get_hint(hints[n], "NAME"); char *desc = snd_device_name_get_hint(hints[n], "DESC"); char *io = snd_device_name_get_hint(hints[n], "IOID"); if (io && strcmp(io, "Output") != 0) continue; char desc2[1024]; snprintf(desc2, sizeof(desc2), "%s", desc ? desc : ""); for (int i = 0; desc2[i]; i++) { if (desc2[i] == '\n') desc2[i] = '/'; } ao_device_list_add(list, ao, &(struct ao_device_desc){name, desc2}); } snd_device_name_free_hint(hints); } #define OPT_BASE_STRUCT struct priv const struct ao_driver audio_out_alsa = { .description = "ALSA 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, .drain = drain, .wait = audio_wait, .wakeup = ao_wakeup_poll, .list_devs = list_devs, .priv_size = sizeof(struct priv), .priv_defaults = &(const struct priv) { .cfg_mixer_device = "default", .cfg_mixer_name = "Master", .cfg_mixer_index = 0, .cfg_ni = 0, }, .options = (const struct m_option[]) { OPT_STRING("device", cfg_device, 0), OPT_FLAG("resample", cfg_resample, 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), OPT_FLAG("non-interleaved", cfg_ni, 0), OPT_FLAG("ignore-chmap", cfg_ignore_chmap, 0), {0} }, };