May 2015 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] ALSA: hda - Set patch_ops before calling auto-parser
by Takashi Iwai 29 May '15

29 May '15

... so that the parser code can overwrite some optional ops. For Realtek and others, it can be set solely in the spec allocator, so it results in a good code cleanup, too. With this change, we can finally remove the local stream_pm setup and rely on the generic parser's automatic setting. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_analog.c | 3 +-- sound/pci/hda/patch_ca0110.c | 3 +-- sound/pci/hda/patch_ca0132.c | 8 +++--- sound/pci/hda/patch_cirrus.c | 12 ++++----- sound/pci/hda/patch_cmedia.c | 4 +-- sound/pci/hda/patch_conexant.c | 3 +-- sound/pci/hda/patch_realtek.c | 61 ++++++++++++++++-------------------------- sound/pci/hda/patch_sigmatel.c | 20 +------------- sound/pci/hda/patch_via.c | 21 +++++---------- 9 files changed, 44 insertions(+), 91 deletions(-) diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index 231f89029779..c033a4ee6547 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -205,8 +205,6 @@ static int ad198x_parse_auto_config(struct hda_codec *codec, bool indep_hp) if (err < 0) return err; - codec->patch_ops = ad198x_auto_patch_ops; - return 0; } @@ -223,6 +221,7 @@ static int alloc_ad_spec(struct hda_codec *codec) return -ENOMEM; codec->spec = spec; snd_hda_gen_spec_init(&spec->gen); + codec->patch_ops = ad198x_auto_patch_ops; return 0; } diff --git a/sound/pci/hda/patch_ca0110.c b/sound/pci/hda/patch_ca0110.c index 447302695195..484bbf4134cd 100644 --- a/sound/pci/hda/patch_ca0110.c +++ b/sound/pci/hda/patch_ca0110.c @@ -63,6 +63,7 @@ static int patch_ca0110(struct hda_codec *codec) return -ENOMEM; snd_hda_gen_spec_init(spec); codec->spec = spec; + codec->patch_ops = ca0110_patch_ops; spec->multi_cap_vol = 1; codec->bus->needs_damn_long_delay = 1; @@ -71,8 +72,6 @@ static int patch_ca0110(struct hda_codec *codec) if (err < 0) goto error; - codec->patch_ops = ca0110_patch_ops; - return 0; error: diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c index 8dec790dae45..0f039abe9673 100644 --- a/sound/pci/hda/patch_ca0132.c +++ b/sound/pci/hda/patch_ca0132.c @@ -4709,6 +4709,10 @@ static int patch_ca0132(struct hda_codec *codec) codec->spec = spec; spec->codec = codec; + codec->patch_ops = ca0132_patch_ops; + codec->pcm_format_first = 1; + codec->no_sticky_stream = 1; + /* Detect codec quirk */ quirk = snd_pci_quirk_lookup(codec->bus->pci, ca0132_quirks); if (quirk) @@ -4737,10 +4741,6 @@ static int patch_ca0132(struct hda_codec *codec) if (err < 0) return err; - codec->patch_ops = ca0132_patch_ops; - codec->pcm_format_first = 1; - codec->no_sticky_stream = 1; - return 0; } diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c index 50e9dd675579..25ccf781fbe7 100644 --- a/sound/pci/hda/patch_cirrus.c +++ b/sound/pci/hda/patch_cirrus.c @@ -584,6 +584,7 @@ static int patch_cs420x(struct hda_codec *codec) if (!spec) return -ENOMEM; + codec->patch_ops = cs_patch_ops; spec->gen.automute_hook = cs_automute; codec->single_adc_amp = 1; @@ -595,8 +596,6 @@ static int patch_cs420x(struct hda_codec *codec) if (err < 0) goto error; - codec->patch_ops = cs_patch_ops; - snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PROBE); return 0; @@ -738,6 +737,7 @@ static int patch_cs4208(struct hda_codec *codec) if (!spec) return -ENOMEM; + codec->patch_ops = cs_patch_ops; spec->gen.automute_hook = cs_automute; /* exclude NID 0x10 (HP) from output volumes due to different steps */ spec->gen.out_vol_mask = 1ULL << 0x10; @@ -756,8 +756,6 @@ static int patch_cs4208(struct hda_codec *codec) if (err < 0) goto error; - codec->patch_ops = cs_patch_ops; - snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PROBE); return 0; @@ -1150,6 +1148,7 @@ static int patch_cs4210(struct hda_codec *codec) if (!spec) return -ENOMEM; + codec->patch_ops = cs421x_patch_ops; spec->gen.automute_hook = cs_automute; snd_hda_pick_fixup(codec, cs421x_models, cs421x_fixup_tbl, @@ -1167,8 +1166,6 @@ static int patch_cs4210(struct hda_codec *codec) if (err < 0) goto error; - codec->patch_ops = cs421x_patch_ops; - snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PROBE); return 0; @@ -1187,11 +1184,12 @@ static int patch_cs4213(struct hda_codec *codec) if (!spec) return -ENOMEM; + codec->patch_ops = cs421x_patch_ops; + err = cs421x_parse_auto_config(codec); if (err < 0) goto error; - codec->patch_ops = cs421x_patch_ops; return 0; error: diff --git a/sound/pci/hda/patch_cmedia.c b/sound/pci/hda/patch_cmedia.c index 617d9012e78a..f5ed078710f8 100644 --- a/sound/pci/hda/patch_cmedia.c +++ b/sound/pci/hda/patch_cmedia.c @@ -57,6 +57,7 @@ static int patch_cmi9880(struct hda_codec *codec) return -ENOMEM; codec->spec = spec; + codec->patch_ops = cmi_auto_patch_ops; cfg = &spec->gen.autocfg; snd_hda_gen_spec_init(&spec->gen); @@ -67,7 +68,6 @@ static int patch_cmi9880(struct hda_codec *codec) if (err < 0) goto error; - codec->patch_ops = cmi_auto_patch_ops; return 0; error: @@ -86,6 +86,7 @@ static int patch_cmi8888(struct hda_codec *codec) return -ENOMEM; codec->spec = spec; + codec->patch_ops = cmi_auto_patch_ops; cfg = &spec->gen.autocfg; snd_hda_gen_spec_init(&spec->gen); @@ -112,7 +113,6 @@ static int patch_cmi8888(struct hda_codec *codec) } } - codec->patch_ops = cmi_auto_patch_ops; return 0; error: diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 78b719b5b34d..f788a91b544a 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -850,6 +850,7 @@ static int patch_conexant_auto(struct hda_codec *codec) return -ENOMEM; snd_hda_gen_spec_init(&spec->gen); codec->spec = spec; + codec->patch_ops = cx_auto_patch_ops; cx_auto_parse_beep(codec); cx_auto_parse_eapd(codec); @@ -908,8 +909,6 @@ static int patch_conexant_auto(struct hda_codec *codec) if (err < 0) goto error; - codec->patch_ops = cx_auto_patch_ops; - /* Some laptops with Conexant chips show stalls in S3 resume, * which falls into the single-cmd mode. * Better to make reset, then. diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index da2d37158d70..376444a9f11e 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -1003,6 +1003,7 @@ static int alc_alloc_spec(struct hda_codec *codec, hda_nid_t mixer_nid) codec->single_adc_amp = 1; /* FIXME: do we need this for all Realtek codec models? */ codec->spdif_status_reset = 1; + codec->patch_ops = alc_patch_ops; err = alc_codec_rename_from_preset(codec); if (err < 0) { @@ -1447,6 +1448,8 @@ static int patch_alc880(struct hda_codec *codec) spec->gen.need_dac_fix = 1; spec->gen.beep_nid = 0x01; + codec->patch_ops.unsol_event = alc880_unsol_event; + snd_hda_pick_fixup(codec, alc880_fixup_models, alc880_fixup_tbl, alc880_fixups); snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PRE_PROBE); @@ -1459,10 +1462,6 @@ static int patch_alc880(struct hda_codec *codec) if (!spec->gen.no_analog) set_beep_amp(spec, 0x0b, 0x05, HDA_INPUT); - codec->patch_ops = alc_patch_ops; - codec->patch_ops.unsol_event = alc880_unsol_event; - - snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PROBE); return 0; @@ -1699,6 +1698,8 @@ static int patch_alc260(struct hda_codec *codec) spec->gen.prefer_hp_amp = 1; spec->gen.beep_nid = 0x01; + spec->shutup = alc_eapd_shutup; + snd_hda_pick_fixup(codec, alc260_fixup_models, alc260_fixup_tbl, alc260_fixups); snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PRE_PROBE); @@ -1711,9 +1712,6 @@ static int patch_alc260(struct hda_codec *codec) if (!spec->gen.no_analog) set_beep_amp(spec, 0x07, 0x05, HDA_INPUT); - codec->patch_ops = alc_patch_ops; - spec->shutup = alc_eapd_shutup; - snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PROBE); return 0; @@ -2298,8 +2296,6 @@ static int patch_alc882(struct hda_codec *codec) if (!spec->gen.no_analog && spec->gen.beep_nid) set_beep_amp(spec, 0x0b, 0x05, HDA_INPUT); - codec->patch_ops = alc_patch_ops; - snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PROBE); return 0; @@ -2435,6 +2431,8 @@ static int patch_alc262(struct hda_codec *codec) spec = codec->spec; spec->gen.shared_mic_vref_pin = 0x18; + spec->shutup = alc_eapd_shutup; + #if 0 /* pshou 07/11/05 set a zero PCM sample to DAC when FIFO is * under-run @@ -2460,9 +2458,6 @@ static int patch_alc262(struct hda_codec *codec) if (!spec->gen.no_analog && spec->gen.beep_nid) set_beep_amp(spec, 0x0b, 0x05, HDA_INPUT); - codec->patch_ops = alc_patch_ops; - spec->shutup = alc_eapd_shutup; - snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PROBE); return 0; @@ -2566,6 +2561,8 @@ static int patch_alc268(struct hda_codec *codec) spec = codec->spec; spec->gen.beep_nid = 0x01; + spec->shutup = alc_eapd_shutup; + snd_hda_pick_fixup(codec, alc268_fixup_models, alc268_fixup_tbl, alc268_fixups); snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PRE_PROBE); @@ -2587,9 +2584,6 @@ static int patch_alc268(struct hda_codec *codec) (0 << AC_AMPCAP_MUTE_SHIFT)); } - codec->patch_ops = alc_patch_ops; - spec->shutup = alc_eapd_shutup; - snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PROBE); return 0; @@ -5672,6 +5666,12 @@ static int patch_alc269(struct hda_codec *codec) if (codec->core.vendor_id != 0x10ec0292) codec->power_save_node = 1; +#ifdef CONFIG_PM + codec->patch_ops.suspend = alc269_suspend; + codec->patch_ops.resume = alc269_resume; +#endif + spec->shutup = alc269_shutup; + snd_hda_pick_fixup(codec, alc269_fixup_models, alc269_fixup_tbl, alc269_fixups); snd_hda_pick_pin_fixup(codec, alc269_pin_fixup_tbl, alc269_fixups); @@ -5768,15 +5768,6 @@ static int patch_alc269(struct hda_codec *codec) if (!spec->gen.no_analog && spec->gen.beep_nid && spec->gen.mixer_nid) set_beep_amp(spec, spec->gen.mixer_nid, 0x04, HDA_INPUT); - codec->patch_ops = alc_patch_ops; - codec->patch_ops.stream_pm = snd_hda_gen_stream_pm; -#ifdef CONFIG_PM - codec->patch_ops.suspend = alc269_suspend; - codec->patch_ops.resume = alc269_resume; -#endif - if (!spec->shutup) - spec->shutup = alc269_shutup; - snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PROBE); return 0; @@ -5892,6 +5883,10 @@ static int patch_alc861(struct hda_codec *codec) spec = codec->spec; spec->gen.beep_nid = 0x23; +#ifdef CONFIG_PM + spec->power_hook = alc_power_eapd; +#endif + snd_hda_pick_fixup(codec, NULL, alc861_fixup_tbl, alc861_fixups); snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PRE_PROBE); @@ -5903,11 +5898,6 @@ static int patch_alc861(struct hda_codec *codec) if (!spec->gen.no_analog) set_beep_amp(spec, 0x23, 0, HDA_OUTPUT); - codec->patch_ops = alc_patch_ops; -#ifdef CONFIG_PM - spec->power_hook = alc_power_eapd; -#endif - snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PROBE); return 0; @@ -5984,6 +5974,8 @@ static int patch_alc861vd(struct hda_codec *codec) spec = codec->spec; spec->gen.beep_nid = 0x23; + spec->shutup = alc_eapd_shutup; + snd_hda_pick_fixup(codec, NULL, alc861vd_fixup_tbl, alc861vd_fixups); snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PRE_PROBE); @@ -5995,10 +5987,6 @@ static int patch_alc861vd(struct hda_codec *codec) if (!spec->gen.no_analog) set_beep_amp(spec, 0x0b, 0x05, HDA_INPUT); - codec->patch_ops = alc_patch_ops; - - spec->shutup = alc_eapd_shutup; - snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PROBE); return 0; @@ -6599,6 +6587,8 @@ static int patch_alc662(struct hda_codec *codec) spec = codec->spec; + spec->shutup = alc_eapd_shutup; + /* handle multiple HPs as is */ spec->parse_flags = HDA_PINCFG_NO_HP_FIXUP; @@ -6650,9 +6640,6 @@ static int patch_alc662(struct hda_codec *codec) } } - codec->patch_ops = alc_patch_ops; - spec->shutup = alc_eapd_shutup; - snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PROBE); return 0; @@ -6689,8 +6676,6 @@ static int patch_alc680(struct hda_codec *codec) return err; } - codec->patch_ops = alc_patch_ops; - return 0; } diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 6ff92aba0f57..bdcda6a41032 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -4417,6 +4417,7 @@ static int alloc_stac_spec(struct hda_codec *codec) codec->spec = spec; codec->no_trigger_sense = 1; /* seems common with STAC/IDT codecs */ spec->gen.dac_min_mute = true; + codec->patch_ops = stac_patch_ops; return 0; } @@ -4433,7 +4434,6 @@ static int patch_stac9200(struct hda_codec *codec) spec->linear_tone_beep = 1; spec->gen.own_eapd_ctl = 1; - codec->patch_ops = stac_patch_ops; codec->power_filter = snd_hda_codec_eapd_power_filter; snd_hda_add_verbs(codec, stac9200_eapd_init); @@ -4466,8 +4466,6 @@ static int patch_stac925x(struct hda_codec *codec) spec->linear_tone_beep = 1; spec->gen.own_eapd_ctl = 1; - codec->patch_ops = stac_patch_ops; - snd_hda_add_verbs(codec, stac925x_core_init); snd_hda_pick_fixup(codec, stac925x_models, stac925x_fixup_tbl, @@ -4537,8 +4535,6 @@ static int patch_stac92hd73xx(struct hda_codec *codec) spec->gen.own_eapd_ctl = 1; spec->gen.power_down_unused = 1; - codec->patch_ops = stac_patch_ops; - snd_hda_pick_fixup(codec, stac92hd73xx_models, stac92hd73xx_fixup_tbl, stac92hd73xx_fixups); snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PRE_PROBE); @@ -4614,8 +4610,6 @@ static int patch_stac92hd83xxx(struct hda_codec *codec) spec->num_pwrs = ARRAY_SIZE(stac92hd83xxx_pwr_nids); spec->default_polarity = -1; /* no default cfg */ - codec->patch_ops = stac_patch_ops; - snd_hda_add_verbs(codec, stac92hd83xxx_core_init); snd_hda_pick_fixup(codec, stac92hd83xxx_models, stac92hd83xxx_fixup_tbl, @@ -4664,8 +4658,6 @@ static int patch_stac92hd95(struct hda_codec *codec) spec->num_pwrs = ARRAY_SIZE(stac92hd95_pwr_nids); spec->default_polarity = 0; - codec->patch_ops = stac_patch_ops; - snd_hda_pick_fixup(codec, stac92hd95_models, stac92hd95_fixup_tbl, stac92hd95_fixups); snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PRE_PROBE); @@ -4704,8 +4696,6 @@ static int patch_stac92hd71bxx(struct hda_codec *codec) spec->gen.mixer_nid = 0x17; spec->have_spdif_mux = 1; - codec->patch_ops = stac_patch_ops; - /* GPIO0 = EAPD */ spec->gpio_mask = 0x01; spec->gpio_dir = 0x01; @@ -4784,8 +4774,6 @@ static int patch_stac922x(struct hda_codec *codec) spec->linear_tone_beep = 1; spec->gen.own_eapd_ctl = 1; - codec->patch_ops = stac_patch_ops; - snd_hda_add_verbs(codec, stac922x_core_init); /* Fix Mux capture level; max to 2 */ @@ -4841,8 +4829,6 @@ static int patch_stac927x(struct hda_codec *codec) spec->aloopback_shift = 0; spec->eapd_switch = 1; - codec->patch_ops = stac_patch_ops; - snd_hda_pick_fixup(codec, stac927x_models, stac927x_fixup_tbl, stac927x_fixups); snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PRE_PROBE); @@ -4904,8 +4890,6 @@ static int patch_stac9205(struct hda_codec *codec) /* Turn on/off EAPD per HP plugging */ spec->eapd_switch = 1; - codec->patch_ops = stac_patch_ops; - snd_hda_pick_fixup(codec, stac9205_models, stac9205_fixup_tbl, stac9205_fixups); snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PRE_PROBE); @@ -4977,8 +4961,6 @@ static int patch_stac9872(struct hda_codec *codec) spec->linear_tone_beep = 1; spec->gen.own_eapd_ctl = 1; - codec->patch_ops = stac_patch_ops; - snd_hda_add_verbs(codec, stac9872_core_init); snd_hda_pick_fixup(codec, stac9872_models, stac9872_fixup_tbl, diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index 31a95cca015d..2c6c6727ca93 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -117,6 +117,8 @@ static void via_playback_pcm_hook(struct hda_pcm_stream *hinfo, struct snd_pcm_substream *substream, int action); +static const struct hda_codec_ops via_patch_ops; /* defined below */ + static struct via_spec *via_new_spec(struct hda_codec *codec) { struct via_spec *spec; @@ -137,6 +139,7 @@ static struct via_spec *via_new_spec(struct hda_codec *codec) spec->gen.add_stereo_mix_input = HDA_HINT_STEREO_MIX_AUTO; codec->power_save_node = 1; spec->gen.power_down_unused = 1; + codec->patch_ops = via_patch_ops; return spec; } @@ -472,7 +475,6 @@ static const struct hda_codec_ops via_patch_ops = { .init = via_init, .free = via_free, .unsol_event = snd_hda_jack_unsol_event, - .stream_pm = snd_hda_gen_stream_pm, #ifdef CONFIG_PM .suspend = via_suspend, .check_power_status = via_check_power_status, @@ -651,6 +653,9 @@ static int patch_vt1708(struct hda_codec *codec) if (spec == NULL) return -ENOMEM; + /* override some patch_ops */ + codec->patch_ops.build_controls = vt1708_build_controls; + codec->patch_ops.build_pcms = vt1708_build_pcms; spec->gen.mixer_nid = 0x17; /* set jackpoll_interval while parsing the codec */ @@ -679,10 +684,6 @@ static int patch_vt1708(struct hda_codec *codec) spec->init_verbs[spec->num_iverbs++] = vt1708_init_verbs; - codec->patch_ops = via_patch_ops; - codec->patch_ops.build_controls = vt1708_build_controls; - codec->patch_ops.build_pcms = vt1708_build_pcms; - /* clear jackpoll_interval again; it's set dynamically */ codec->jackpoll_interval = 0; @@ -707,8 +708,6 @@ static int patch_vt1709(struct hda_codec *codec) return err; } - codec->patch_ops = via_patch_ops; - return 0; } @@ -735,7 +734,6 @@ static int patch_vt1708B(struct hda_codec *codec) return err; } - codec->patch_ops = via_patch_ops; return 0; } @@ -800,7 +798,6 @@ static int patch_vt1708S(struct hda_codec *codec) spec->init_verbs[spec->num_iverbs++] = vt1708S_init_verbs; - codec->patch_ops = via_patch_ops; return 0; } @@ -842,7 +839,6 @@ static int patch_vt1702(struct hda_codec *codec) spec->init_verbs[spec->num_iverbs++] = vt1702_init_verbs; - codec->patch_ops = via_patch_ops; return 0; } @@ -915,7 +911,6 @@ static int patch_vt1718S(struct hda_codec *codec) spec->init_verbs[spec->num_iverbs++] = vt1718S_init_verbs; - codec->patch_ops = via_patch_ops; return 0; } @@ -1015,7 +1010,6 @@ static int patch_vt1716S(struct hda_codec *codec) spec->mixers[spec->num_mixers++] = vt1716s_dmic_mixer; spec->mixers[spec->num_mixers++] = vt1716S_mono_out_mixer; - codec->patch_ops = via_patch_ops; return 0; } @@ -1123,7 +1117,6 @@ static int patch_vt2002P(struct hda_codec *codec) else spec->init_verbs[spec->num_iverbs++] = vt2002P_init_verbs; - codec->patch_ops = via_patch_ops; return 0; } @@ -1162,7 +1155,6 @@ static int patch_vt1812(struct hda_codec *codec) spec->init_verbs[spec->num_iverbs++] = vt1812_init_verbs; - codec->patch_ops = via_patch_ops; return 0; } @@ -1200,7 +1192,6 @@ static int patch_vt3476(struct hda_codec *codec) spec->init_verbs[spec->num_iverbs++] = vt3476_init_verbs; - codec->patch_ops = via_patch_ops; return 0; } -- 2.4.2

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[alsa-devel] [PATCH] ASoC: DAPM: Add APIs to create individual DAPM controls.
by Liam Girdwood 29 May '15

29 May '15

The topology core needs to be able to create individual widget controls at runtime and driver init. Add a regular locked and unlocked API calls to facilitate this requirement. The unlocked call is used by the topology core during component driver probing where the card dapm_mutex is held by the ASoC core and the locked version at non component driver probe time. Signed-off-by: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> --- sound/soc/soc-dapm.c | 34 ++++++++++++++++++++++++++-------- 1 file changed, 26 insertions(+), 8 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index defe0f0..62406d8 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -52,8 +52,8 @@ static int snd_soc_dapm_add_path(struct snd_soc_dapm_context *dapm, const char *control, int (*connected)(struct snd_soc_dapm_widget *source, struct snd_soc_dapm_widget *sink)); -static struct snd_soc_dapm_widget * -snd_soc_dapm_new_control(struct snd_soc_dapm_context *dapm, +struct snd_soc_dapm_widget * +snd_soc_dapm_new_control_unlocked(struct snd_soc_dapm_context *dapm, const struct snd_soc_dapm_widget *widget); /* dapm power sequences - make this per codec in the future */ @@ -342,7 +342,8 @@ static int dapm_kcontrol_data_alloc(struct snd_soc_dapm_widget *widget, data->value = template.on_val; - data->widget = snd_soc_dapm_new_control(widget->dapm, + data->widget = + snd_soc_dapm_new_control_unlocked(widget->dapm, &template); if (!data->widget) { kfree(data); @@ -3053,8 +3054,25 @@ int snd_soc_dapm_put_pin_switch(struct snd_kcontrol *kcontrol, } EXPORT_SYMBOL_GPL(snd_soc_dapm_put_pin_switch); -static struct snd_soc_dapm_widget * +struct snd_soc_dapm_widget * snd_soc_dapm_new_control(struct snd_soc_dapm_context *dapm, + const struct snd_soc_dapm_widget *widget) +{ + struct snd_soc_dapm_widget *w; + + mutex_lock_nested(&dapm->card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME); + w = snd_soc_dapm_new_control_unlocked(dapm, widget); + if (!w) + dev_err(dapm->dev, + "ASoC: Failed to create DAPM control %s\n", + widget->name); + + mutex_unlock(&dapm->card->dapm_mutex); + return w; +} + +struct snd_soc_dapm_widget * +snd_soc_dapm_new_control_unlocked(struct snd_soc_dapm_context *dapm, const struct snd_soc_dapm_widget *widget) { struct snd_soc_dapm_widget *w; @@ -3199,7 +3217,7 @@ int snd_soc_dapm_new_controls(struct snd_soc_dapm_context *dapm, mutex_lock_nested(&dapm->card->dapm_mutex, SND_SOC_DAPM_CLASS_INIT); for (i = 0; i < num; i++) { - w = snd_soc_dapm_new_control(dapm, widget); + w = snd_soc_dapm_new_control_unlocked(dapm, widget); if (!w) { dev_err(dapm->dev, "ASoC: Failed to create DAPM control %s\n", @@ -3437,7 +3455,7 @@ int snd_soc_dapm_new_pcm(struct snd_soc_card *card, dev_dbg(card->dev, "ASoC: adding %s widget\n", link_name); - w = snd_soc_dapm_new_control(&card->dapm, &template); + w = snd_soc_dapm_new_control_unlocked(&card->dapm, &template); if (!w) { dev_err(card->dev, "ASoC: Failed to create %s widget\n", link_name); @@ -3488,7 +3506,7 @@ int snd_soc_dapm_new_dai_widgets(struct snd_soc_dapm_context *dapm, dev_dbg(dai->dev, "ASoC: adding %s widget\n", template.name); - w = snd_soc_dapm_new_control(dapm, &template); + w = snd_soc_dapm_new_control_unlocked(dapm, &template); if (!w) { dev_err(dapm->dev, "ASoC: Failed to create %s widget\n", dai->driver->playback.stream_name); @@ -3507,7 +3525,7 @@ int snd_soc_dapm_new_dai_widgets(struct snd_soc_dapm_context *dapm, dev_dbg(dai->dev, "ASoC: adding %s widget\n", template.name); - w = snd_soc_dapm_new_control(dapm, &template); + w = snd_soc_dapm_new_control_unlocked(dapm, &template); if (!w) { dev_err(dapm->dev, "ASoC: Failed to create %s widget\n", dai->driver->capture.stream_name); -- 2.1.4

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[alsa-devel] [PATCH v3] ASoC: wm_adsp: Dump scratch registers on DSP shutdown
by Richard Fitzgerald 29 May '15

29 May '15

The SCRATCH registers are used by firmwares to hold diagnostic information. Log this during shutdown to assist analysis and debug of firmwares. Signed-off-by: Richard Fitzgerald <rf(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm_adsp.c | 27 +++++++++++++++++++++++++++ 1 files changed, 27 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index f6642c1..477390a 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -121,6 +121,11 @@ #define ADSP2_WDMA_CONFIG_2 0x31 #define ADSP2_RDMA_CONFIG_1 0x34 +#define ADSP2_SCRATCH0 0x40 +#define ADSP2_SCRATCH1 0x41 +#define ADSP2_SCRATCH2 0x42 +#define ADSP2_SCRATCH3 0x43 + /* * ADSP2 Control */ @@ -364,6 +369,25 @@ static unsigned int wm_adsp_region_to_reg(struct wm_adsp_region const *mem, } } +static void wm_adsp2_show_fw_status(struct wm_adsp *dsp) +{ + u16 scratch[4]; + int ret; + + ret = regmap_raw_read(dsp->regmap, dsp->base + ADSP2_SCRATCH0, + scratch, sizeof(scratch)); + if (ret) { + adsp_err(dsp, "Failed to read SCRATCH regs: %d\n", ret); + return; + } + + adsp_dbg(dsp, "FW SCRATCH 0:0x%x 1:0x%x 2:0x%x 3:0x%x\n", + be16_to_cpu(scratch[0]), + be16_to_cpu(scratch[1]), + be16_to_cpu(scratch[2]), + be16_to_cpu(scratch[3])); +} + static int wm_coeff_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { @@ -1898,6 +1922,9 @@ int wm_adsp2_event(struct snd_soc_dapm_widget *w, break; case SND_SOC_DAPM_PRE_PMD: + /* Log firmware state, it can be useful for analysis */ + wm_adsp2_show_fw_status(dsp); + dsp->running = false; regmap_update_bits(dsp->regmap, dsp->base + ADSP2_CONTROL, -- 1.7.2.5

2 1

Re: [alsa-devel] ASoC: PATCH: wm8731 - reinitialize regmap cache after hardware reset
by Mark Brown 29 May '15

29 May '15

On Thu, May 28, 2015 at 02:55:48PM -0700, Sergey Kiselev wrote: Please fix your mail client to word wrap within paragraphs, it makes your mail more readable. > On Thu, May 28, 2015 at 12:26 PM, Mark Brown wrote: > OK, I was trying to move the request supplies and reset code to > wm8731_{i2c|spi}_probe(). And I noticed the following weirdness with > the supplies: The current code in wm8731_probe() requests and enables > regulators, resets the codec hardware, tweaks some registers, and then > disables all the regulators: That's totally normal, the new register values will be rewritten as part of resyncing the register cache when the device is powered back up.

1 0

[alsa-devel] [PATCH] ALSA: fix building with CONFIG_SND_PROC_FS disabled
by Arnd Bergmann 29 May '15

29 May '15

A recent change to split the procfs support for ALSA turned out to be incomplete, leading to build errors: sound/core/seq/seq_info.c:61:12: error: redefinition of 'snd_seq_info_init' sound/core/seq/seq_device.c: In function 'seq_dev_proc_init': sound/core/seq/seq_device.c:276:2: error: 'info_entry' undeclared (first use in this function) sound/core/seq/seq_device.c: In function 'alsa_seq_device_exit': sound/core/seq/seq_device.c:309:22: error: 'info_entry' undeclared (first use in this function) This replaces some more instances of CONFIG_PROC_FS with CONFIG_SND_PROC_FS, as the original patch started. It is enough to avoid all randconfig regressions I saw on ARM because of the CONFIG_SND_PROC_FS introduction. Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> Fixes: cd6a65036f0e ("ALSA: replace CONFIG_PROC_FS with CONFIG_SND_PROC_FS") --- I assume this was already reported. If not, please apply my patch or fold it into the original commit. diff --git a/sound/core/seq/Makefile b/sound/core/seq/Makefile index b29ffe835205..b65fa5a1943b 100644 --- a/sound/core/seq/Makefile +++ b/sound/core/seq/Makefile @@ -7,7 +7,7 @@ snd-seq-device-objs := seq_device.o snd-seq-objs := seq.o seq_lock.o seq_clientmgr.o seq_memory.o seq_queue.o \ seq_fifo.o seq_prioq.o seq_timer.o \ seq_system.o seq_ports.o -snd-seq-$(CONFIG_PROC_FS) += seq_info.o +snd-seq-$(CONFIG_SND_PROC_FS) += seq_info.o snd-seq-midi-objs := seq_midi.o snd-seq-midi-emul-objs := seq_midi_emul.o snd-seq-midi-event-objs := seq_midi_event.o diff --git a/sound/core/seq/seq_device.c b/sound/core/seq/seq_device.c index 288af56a43cd..c4acf17e9f5e 100644 --- a/sound/core/seq/seq_device.c +++ b/sound/core/seq/seq_device.c @@ -272,7 +272,7 @@ EXPORT_SYMBOL_GPL(snd_seq_driver_unregister); static int __init seq_dev_proc_init(void) { -#ifdef CONFIG_PROC_FS +#ifdef CONFIG_SND_PROC_FS info_entry = snd_info_create_module_entry(THIS_MODULE, "drivers", snd_seq_root); if (info_entry == NULL) @@ -305,7 +305,7 @@ static void __exit alsa_seq_device_exit(void) #ifdef CONFIG_MODULES cancel_work_sync(&autoload_work); #endif -#ifdef CONFIG_PROC_FS +#ifdef CONFIG_SND_PROC_FS snd_info_free_entry(info_entry); #endif bus_unregister(&snd_seq_bus_type);

3 2

[alsa-devel] ASoC: codec register dump in debugfs
by Sergej Sawazki 29 May '15

29 May '15

Dear all, please correct me if I am wrong. Since the codec drivers have been converted to direct regmap API usage, it is not common to set the following snd_soc_codec_driver struct members: .reg_cache_size .reg_word_size However, the functions soc_codec_reg_show() and format_register_str() (responsible for formating and printing the codec register dumps to debugfs) depend on these members. I'm writing a ASoC machine driver. The board has two wm8741 codecs. I would like to see the codec register dumps in debugfs. eg: cat /sys/kernel/debug/asoc/<card_name>/codec\:wm8741.1-001a/codec_reg cat /sys/kernel/debug/asoc/<card_name>/codec\:wm8741.1-001b/codec_reg Is it OK to set the above mentioned members in a codec driver to "enable" the register dump output? Or should we refactor the above mentioned functions to use the regmap API? Or am I missing something? Thanks, Sergej

2 1

[alsa-devel] [BUG, bisect] ALSA: hda - No sound output
by Jeremiah Mahler 29 May '15

29 May '15

Takashi, all, On several of my machines there is no longer any sound output when running the latest linux-next (20150528). Activity can be seen in a mixer such as pavucontrol suggesting that there should be sound, but both the speakers and head phones are silent. I have bisected the kernel and found that the following patch introduced the bug. From 49fb18972581a781658a4637de76e6069ed5964e Mon Sep 17 00:00:00 2001 From: Takashi Iwai <tiwai(a)suse.de> Date: Wed, 27 May 2015 08:37:19 +0200 Subject: [PATCH] ALSA: hda - Set stream_pm ops automatically by generic parser This allows user to test power_save_node feature via sysfs or patch firmware even on the codecs that don't specify it. It'll also save a few lines. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_generic.c | 5 ++++- sound/pci/hda/patch_realtek.c | 1 - sound/pci/hda/patch_sigmatel.c | 1 - sound/pci/hda/patch_via.c | 1 - 4 files changed, 4 insertions(+), 4 deletions(-) -- - Jeremiah Mahler

2 2

[alsa-devel] [PATCH 00/10] Leftover CONFIG_PROC_FS fixes & cleanups
by Takashi Iwai 29 May '15

29 May '15

Hi, here is a patch series to fix / clean up the leftover CONFIG_PROC_FS after Keyon's patch to introduce CONFIG_SND_PROC_FS. Takashi === Takashi Iwai (10): ALSA: aloop: Drop unnecessary ifdef CONFIG_PROC_FS ALSA: dummy: Replace CONFIG_PROC_FS with CONFIG_SND_PROC_FS ALSA: opl4: Fix / cleanup ifdef CONFIG_PROC_FS ALSA: ak4xxx-adda: Drop unnecessary ifdef CONFIG_PROC_FS ALSA: ac97: Fix ifdef CONFIG_PROC_FS ALSA: ca0106: Fix/cleanup ifdef CONFIG_PROC_FS ALSA: cs46xx: Fix old ifdef CONFIG_PROC_FS ALSA: emu10k1: Fix/cleanup ifdef CONFIG_PROC_FS ALSA: emux: Fix/cleanup old ifdef CONFIG_PROC_FS ALSA: pci: Drop superfluous ifdef CONFIG_PROC_FS include/sound/emux_synth.h | 2 +- sound/drivers/aloop.c | 8 -------- sound/drivers/dummy.c | 4 ++-- sound/drivers/opl4/Makefile | 3 ++- sound/drivers/opl4/opl4_lib.c | 4 ---- sound/drivers/opl4/opl4_local.h | 7 +++++-- sound/drivers/opl4/opl4_proc.c | 4 ---- sound/i2c/other/ak4xxx-adda.c | 4 ---- sound/pci/ac97/Makefile | 2 +- sound/pci/ac97/ac97_local.h | 2 +- sound/pci/ak4531_codec.c | 6 ------ sound/pci/atiixp.c | 4 ---- sound/pci/atiixp_modem.c | 4 ---- sound/pci/ca0106/Makefile | 3 ++- sound/pci/ca0106/ca0106_main.c | 2 +- sound/pci/ca0106/ca0106_proc.c | 4 ---- sound/pci/cmipci.c | 5 ----- sound/pci/cs46xx/cs46xx_lib.c | 4 ++-- sound/pci/cs46xx/cs46xx_lib.h | 4 ++-- sound/pci/cs46xx/dsp_spos.c | 4 ++-- sound/pci/cs46xx/dsp_spos_scb_lib.c | 6 +++--- sound/pci/emu10k1/Makefile | 3 ++- sound/pci/emu10k1/emu10k1_main.c | 2 +- sound/pci/emu10k1/emuproc.c | 2 -- sound/pci/ice1712/quartet.c | 7 +------ sound/pci/intel8x0.c | 4 ---- sound/pci/intel8x0m.c | 5 ----- sound/pci/oxygen/oxygen_lib.c | 4 ---- sound/synth/emux/Makefile | 5 +++-- sound/synth/emux/emux.c | 4 ---- sound/synth/emux/emux_proc.c | 4 ---- sound/synth/emux/emux_voice.h | 6 +++++- 32 files changed, 36 insertions(+), 96 deletions(-) -- 2.4.2

2 11

[alsa-devel] [PATCH] ALSA: core: Fix randconfig build wrt CONFIG_PROC_FS
by Takashi Iwai 29 May '15

29 May '15

There are a few leftover CONFIG_PROC_FS forgotten to replace with CONFIG_SND_PROC_FS. Fixes: cd6a65036f0e ('ALSA: replace CONFIG_PROC_FS with CONFIG_SND_PROC_FS') Reported-by: Jim Davis <jim.epost(a)gmail.com> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/seq/Makefile | 2 +- sound/core/seq/seq_device.c | 4 ++-- 2 files changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/core/seq/Makefile b/sound/core/seq/Makefile index b29ffe835205..b65fa5a1943b 100644 --- a/sound/core/seq/Makefile +++ b/sound/core/seq/Makefile @@ -7,7 +7,7 @@ snd-seq-device-objs := seq_device.o snd-seq-objs := seq.o seq_lock.o seq_clientmgr.o seq_memory.o seq_queue.o \ seq_fifo.o seq_prioq.o seq_timer.o \ seq_system.o seq_ports.o -snd-seq-$(CONFIG_PROC_FS) += seq_info.o +snd-seq-$(CONFIG_SND_PROC_FS) += seq_info.o snd-seq-midi-objs := seq_midi.o snd-seq-midi-emul-objs := seq_midi_emul.o snd-seq-midi-event-objs := seq_midi_event.o diff --git a/sound/core/seq/seq_device.c b/sound/core/seq/seq_device.c index 288af56a43cd..c4acf17e9f5e 100644 --- a/sound/core/seq/seq_device.c +++ b/sound/core/seq/seq_device.c @@ -272,7 +272,7 @@ EXPORT_SYMBOL_GPL(snd_seq_driver_unregister); static int __init seq_dev_proc_init(void) { -#ifdef CONFIG_PROC_FS +#ifdef CONFIG_SND_PROC_FS info_entry = snd_info_create_module_entry(THIS_MODULE, "drivers", snd_seq_root); if (info_entry == NULL) @@ -305,7 +305,7 @@ static void __exit alsa_seq_device_exit(void) #ifdef CONFIG_MODULES cancel_work_sync(&autoload_work); #endif -#ifdef CONFIG_PROC_FS +#ifdef CONFIG_SND_PROC_FS snd_info_free_entry(info_entry); #endif bus_unregister(&snd_seq_bus_type); -- 2.4.2

1 0

[alsa-devel] [PATCH v2 1/3] ASoC: topology: Add topology UAPI header
by Liam Girdwood 29 May '15

29 May '15

The ASoC topology UAPI header defines the structures required to define any DSP firmware audio topology and control objects from userspace. The following objects are supported :- o kcontrols including TLV controls. o DAPM widgets and graph elements o Vendor bespoke objects. o Coefficient data o FE PCM capabilities and config. o BE link capabilities and config. o Codec <-> codec link capabilities and config. o Topology object manifest. The file format is simple and divided into blocks for each object type and each block has a header that defines it's size and type. Blocks can be in any order of type and can either all be in a single file or spread across more than one file. Blocks also have a group identifier ID so that they can be loaded and unloaded by ID. Signed-off-by: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> --- include/sound/soc-dapm.h | 1 + include/uapi/sound/asoc.h | 388 ++++++++++++++++++++++++++++++++++++++++++++++ 2 files changed, 389 insertions(+) create mode 100644 include/uapi/sound/asoc.h diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index 0bc8364..4f1b417 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -15,6 +15,7 @@ #include <linux/types.h> #include <sound/control.h> +#include <sound/asoc.h> struct device; diff --git a/include/uapi/sound/asoc.h b/include/uapi/sound/asoc.h new file mode 100644 index 0000000..1221520 --- /dev/null +++ b/include/uapi/sound/asoc.h @@ -0,0 +1,388 @@ +/* + * uapi/sound/asoc.h -- ALSA SoC Firmware Controls and DAPM + * + * Copyright (C) 2012 Texas Instruments Inc. + * Copyright (C) 2015 Intel Corporation. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + * + * Simple file API to load FW that includes mixers, coefficients, DAPM graphs, + * algorithms, equalisers, DAIs, widgets etc. +*/ + +#ifndef __LINUX_UAPI_SND_ASOC_H +#define __LINUX_UAPI_SND_ASOC_H + +#include <linux/types.h> +#include <sound/asound.h> + +/* + * Maximum number of channels topology kcontrol can represent. + */ +#define SND_SOC_TPLG_MAX_CHAN 8 + +/* + * Maximum number of PCM formats capability + */ +#define SND_SOC_TPLG_MAX_FORMATS 16 + +/* + * Maximum number of PCM stream configs + */ +#define SND_SOC_TPLG_STREAM_CONFIG_MAX 8 + +/* individual kcontrol info types - can be mixed with other types */ +#define SND_SOC_TPLG_CTL_VOLSW 1 +#define SND_SOC_TPLG_CTL_VOLSW_SX 2 +#define SND_SOC_TPLG_CTL_VOLSW_XR_SX 3 +#define SND_SOC_TPLG_CTL_ENUM 4 +#define SND_SOC_TPLG_CTL_BYTES 5 +#define SND_SOC_TPLG_CTL_ENUM_VALUE 6 +#define SND_SOC_TPLG_CTL_RANGE 7 +#define SND_SOC_TPLG_CTL_STROBE 8 + + +/* individual widget kcontrol info types - can be mixed with other types */ +#define SND_SOC_TPLG_DAPM_CTL_VOLSW 64 +#define SND_SOC_TPLG_DAPM_CTL_ENUM_DOUBLE 65 +#define SND_SOC_TPLG_DAPM_CTL_ENUM_VIRT 66 +#define SND_SOC_TPLG_DAPM_CTL_ENUM_VALUE 67 +#define SND_SOC_TPLG_DAPM_CTL_PIN 68 + +/* DAPM widget types - add new items to the end */ +#define SND_SOC_TPLG_DAPM_INPUT 0 +#define SND_SOC_TPLG_DAPM_OUTPUT 1 +#define SND_SOC_TPLG_DAPM_MUX 2 +#define SND_SOC_TPLG_DAPM_MIXER 3 +#define SND_SOC_TPLG_DAPM_PGA 4 +#define SND_SOC_TPLG_DAPM_OUT_DRV 5 +#define SND_SOC_TPLG_DAPM_ADC 6 +#define SND_SOC_TPLG_DAPM_DAC 7 +#define SND_SOC_TPLG_DAPM_SWITCH 8 +#define SND_SOC_TPLG_DAPM_PRE 9 +#define SND_SOC_TPLG_DAPM_POST 10 +#define SND_SOC_TPLG_DAPM_AIF_IN 11 +#define SND_SOC_TPLG_DAPM_AIF_OUT 12 +#define SND_SOC_TPLG_DAPM_DAI_IN 13 +#define SND_SOC_TPLG_DAPM_DAI_OUT 14 +#define SND_SOC_TPLG_DAPM_DAI_LINK 15 +#define SND_SOC_TPLG_DAPM_LAST SND_SOC_TPLG_DAPM_DAI_LINK + +/* Header magic number and string sizes */ +#define SND_SOC_TPLG_MAGIC 0x41536F43 /* ASoC */ + +/* string sizes */ +#define SND_SOC_TPLG_NUM_TEXTS 16 + +/* ABI version */ +#define SND_SOC_TPLG_ABI_VERSION 0x2 + +/* Max size of TLV data */ +#define SND_SOC_TPLG_TLV_SIZE 32 + +/* + * File and Block header data types. + * Add new generic and vendor types to end of list. + * Generic types are handled by the core whilst vendors types are passed + * to the component drivers for handling. + */ +#define SND_SOC_TPLG_TYPE_MIXER 1 +#define SND_SOC_TPLG_TYPE_BYTES 2 +#define SND_SOC_TPLG_TYPE_ENUM 3 +#define SND_SOC_TPLG_TYPE_DAPM_GRAPH 4 +#define SND_SOC_TPLG_TYPE_DAPM_WIDGET 5 +#define SND_SOC_TPLG_TYPE_DAI_LINK 6 +#define SND_SOC_TPLG_TYPE_PCM 7 +#define SND_SOC_TPLG_TYPE_MANIFEST 8 +#define SND_SOC_TPLG_TYPE_CODEC_LINK 9 +#define SND_SOC_TPLG_TYPE_MAX SND_SOC_TPLG_TYPE_CODEC_LINK + +/* vendor block IDs - please add new vendor types to end */ +#define SND_SOC_TPLG_TYPE_VENDOR_FW 1000 +#define SND_SOC_TPLG_TYPE_VENDOR_CONFIG 1001 +#define SND_SOC_TPLG_TYPE_VENDOR_COEFF 1002 +#define SND_SOC_TPLG_TYPEVENDOR_CODEC 1003 + +#define SND_SOC_TPLG_STREAM_PLAYBACK 0 +#define SND_SOC_TPLG_STREAM_CAPTURE 1 + +/* + * Block Header. + * This header preceeds all object and object arrays below. + */ +struct snd_soc_tplg_hdr { + __le32 magic; /* magic number */ + __le32 abi; /* ABI version */ + __le32 version; /* optional vendor specific version details */ + __le32 type; /* SND_SOC_TPLG_TYPE_ */ + __le32 size; /* size of this structure */ + __le32 vendor_type; /* optional vendor specific type info */ + __le32 payload_size; /* data bytes, excluding this header */ + __le32 index; /* identifier for block */ + __le32 count; /* number of elements in block */ +} __attribute__((packed)); + +/* + * Private data. + * All topology objects may have private data that can be used by the driver or + * firmware. Core will ignore this data. + */ +struct snd_soc_tplg_private { + __le32 size; /* in bytes of private data */ + char data[0]; +} __attribute__((packed)); + +/* + * Kcontrol TLV data. + */ +struct snd_soc_tplg_ctl_tlv { + __le32 size; /* in bytes aligned to 4 */ + __le32 numid; /* control element numeric identification */ + __le32 count; /* number of elem in data array */ + __le32 data[SND_SOC_TPLG_TLV_SIZE]; +} __attribute__((packed)); + +/* + * Kcontrol channel data + */ +struct snd_soc_tplg_channel { + __le32 size; /* in bytes of this structure */ + __le32 reg; + __le32 shift; + __le32 id; /* ID maps to Left, Right, LFE etc */ +} __attribute__((packed)); + +/* + * Kcontrol Operations IDs + */ +struct snd_soc_tplg_kcontrol_ops_id { + __le32 get; + __le32 put; + __le32 info; +} __attribute__((packed)); + +/* + * kcontrol header + */ +struct snd_soc_tplg_ctl_hdr { + __le32 size; /* in bytes of this structure */ + __le32 type; + char name[SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; + __le32 access; + struct snd_soc_tplg_kcontrol_ops_id ops; + __le32 tlv_size; /* non zero means control has TLV data */ +} __attribute__((packed)); + +/* + * Stream Capabilities + */ +struct snd_soc_tplg_stream_caps { + __le32 size; /* in bytes of this structure */ + char name[SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; + __le64 formats[SND_SOC_TPLG_MAX_FORMATS]; /* supported formats SNDRV_PCM_FMTBIT_* */ + __le32 rates; /* supported rates SNDRV_PCM_RATE_* */ + __le32 rate_min; /* min rate */ + __le32 rate_max; /* max rate */ + __le32 channels_min; /* min channels */ + __le32 channels_max; /* max channels */ + __le32 periods_min; /* min number of periods */ + __le32 periods_max; /* max number of periods */ + __le32 period_size_min; /* min period size bytes */ + __le32 period_size_max; /* max period size bytes */ + __le32 buffer_size_min; /* min buffer size bytes */ + __le32 buffer_size_max; /* max buffer size bytes */ +} __attribute__((packed)); + +/* + * FE or BE Stream configuration supported by SW/FW + */ +struct snd_soc_tplg_stream { + __le32 size; /* in bytes of this structure */ + __le64 format; /* SNDRV_PCM_FMTBIT_* */ + __le32 rate; /* SNDRV_PCM_RATE_* */ + __le32 period_bytes; /* size of period in bytes */ + __le32 buffer_bytes; /* size of buffer in bytes */ + __le32 channels; /* channels */ + __le32 tdm_slot; /* optional BE bitmask of supported TDM slots */ + __le32 dai_fmt; /* SND_SOC_DAIFMT_ */ +} __attribute__((packed)); + +/* + * Duplex stream configuration supported by SW/FW. + */ +struct snd_soc_tplg_stream_config { + __le32 size; /* in bytes of this structure */ + char name[SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; + struct snd_soc_tplg_stream playback; + struct snd_soc_tplg_stream capture; +} __attribute__((packed)); + +/* + * Manifest. List totals for each payload type. Not used in parsing, but will + * be passed to the component driver before any other objects in order for any + * global componnent resource allocations. + * + * File block representation for manifest :- + * +-----------------------------------+----+ + * | struct snd_soc_tplg_hdr | 1 | + * +-----------------------------------+----+ + * | struct snd_soc_tplg_manifest | 1 | + * +-----------------------------------+----+ + */ +struct snd_soc_tplg_manifest { + __le32 size; /* in bytes of this structure */ + __le32 control_elems; /* number of control elements */ + __le32 widget_elems; /* number of widget elements */ + __le32 graph_elems; /* number of graph elements */ + __le32 dai_elems; /* number of DAI elements */ + __le32 dai_link_elems; /* number of DAI link elements */ +} __attribute__((packed)); + +/* + * Mixer kcontrol. + * + * File block representation for mixer kcontrol :- + * +-----------------------------------+----+ + * | struct snd_soc_tplg_hdr | 1 | + * +-----------------------------------+----+ + * | struct snd_soc_tplg_mixer_control | N | + * +-----------------------------------+----+ + */ +struct snd_soc_tplg_mixer_control { + struct snd_soc_tplg_ctl_hdr hdr; + __le32 size; /* in bytes of this structure */ + __le32 min; + __le32 max; + __le32 platform_max; + __le32 invert; + __le32 num_channels; + struct snd_soc_tplg_channel channel[SND_SOC_TPLG_MAX_CHAN]; + struct snd_soc_tplg_ctl_tlv tlv; + struct snd_soc_tplg_private priv; +} __attribute__((packed)); + +/* + * Enumerated kcontrol + * + * File block representation for enum kcontrol :- + * +-----------------------------------+----+ + * | struct snd_soc_tplg_hdr | 1 | + * +-----------------------------------+----+ + * | struct snd_soc_tplg_enum_control | N | + * +-----------------------------------+----+ + */ +struct snd_soc_tplg_enum_control { + struct snd_soc_tplg_ctl_hdr hdr; + __le32 size; /* in bytes of this structure */ + __le32 num_channels; + struct snd_soc_tplg_channel channel[SND_SOC_TPLG_MAX_CHAN]; + __le32 items; + __le32 mask; + __le32 count; + char texts[SND_SOC_TPLG_NUM_TEXTS][SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; + __le32 values[SND_SOC_TPLG_NUM_TEXTS * SNDRV_CTL_ELEM_ID_NAME_MAXLEN / 4]; + struct snd_soc_tplg_private priv; +} __attribute__((packed)); + +/* + * Bytes kcontrol + * + * File block representation for bytes kcontrol :- + * +-----------------------------------+----+ + * | struct snd_soc_tplg_hdr | 1 | + * +-----------------------------------+----+ + * | struct snd_soc_tplg_bytes_control | N | + * +-----------------------------------+----+ + */ +struct snd_soc_tplg_bytes_control { + struct snd_soc_tplg_ctl_hdr hdr; + __le32 size; /* in bytes of this structure */ + __le32 max; + __le32 mask; + __le32 base; + __le32 num_regs; + struct snd_soc_tplg_private priv; +} __attribute__((packed)); + +/* + * DAPM Graph Element + * + * File block representation for DAPM graph elements :- + * +-------------------------------------+----+ + * | struct snd_soc_tplg_hdr | 1 | + * +-------------------------------------+----+ + * | struct snd_soc_tplg_dapm_graph_elem | N | + * +-------------------------------------+----+ + */ +struct snd_soc_tplg_dapm_graph_elem { + char sink[SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; + char control[SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; + char source[SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; +} __attribute__((packed)); + +/* + * DAPM Widget. + * + * File block representation for DAPM widget :- + * +-------------------------------------+-----+ + * | struct snd_soc_tplg_hdr | 1 | + * +-------------------------------------+-----+ + * | struct snd_soc_tplg_dapm_widget | N | + * +-------------------------------------+-----+ + * | struct snd_soc_tplg_enum_control | 0|1 | + * | struct snd_soc_tplg_mixer_control | 0|N | + * +-------------------------------------+-----+ + * + * Optional enum or mixer control can be appended to the end of each widget + * in the block. + */ +struct snd_soc_tplg_dapm_widget { + __le32 size; /* in bytes of this structure */ + __le32 id; /* SND_SOC_DAPM_CTL */ + char name[SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; + char sname[SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; + + __le32 reg; /* negative reg = no direct dapm */ + __le32 shift; /* bits to shift */ + __le32 mask; /* non-shifted mask */ + __u32 invert; /* invert the power bit */ + __u32 ignore_suspend; /* kept enabled over suspend */ + __u16 event_flags; + __u16 event_type; + __u16 num_kcontrols; + struct snd_soc_tplg_private priv; + /* + * kcontrols that relate to this widget + * follow here after widget private data + */ +} __attribute__((packed)); + +struct snd_soc_tplg_pcm_cfg_caps { + struct snd_soc_tplg_stream_caps caps; + struct snd_soc_tplg_stream_config configs[SND_SOC_TPLG_STREAM_CONFIG_MAX]; + __le32 num_configs; /* number of configs */ +} __attribute__((packed)); + +/* + * Describes SW/FW specific features of PCM or DAI link. + * + * File block representation for PCM/DAI-Link :- + * +-----------------------------------+-----+ + * | struct snd_soc_tplg_hdr | 1 | + * +-----------------------------------+-----+ + * | struct snd_soc_tplg_dapm_pcm_dai | N | + * +-----------------------------------+-----+ + */ +struct snd_soc_tplg_pcm_dai { + __le32 size; /* in bytes of this structure */ + char name[SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; + __le32 id; /* unique ID - used to match */ + __le32 playback; /* supports playback mode */ + __le32 capture; /* supports capture mode */ + __le32 compress; /* 1 = compressed; 0 = PCM */ + struct snd_soc_tplg_pcm_cfg_caps capconf[2]; /* capabilities and configs */ +} __attribute__((packed)); + +#endif -- 2.1.4

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