July 2011 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] ALSA: asihpi - bug fix pa use before init.
by linux＠audioscience.com 27 Jul '11

27 Jul '11

From: Eliot Blennerhassett <eblennerhassett(a)audioscience.com> Fixes bug introduced by 1c073b67. Also declare pa local to block in which it is used. Signed-off-by: Eliot Blennerhassett <eblennerhassett(a)audioscience.com> --- sound/pci/asihpi/hpioctl.c | 11 ++++------- 1 files changed, 4 insertions(+), 7 deletions(-) diff --git a/sound/pci/asihpi/hpioctl.c b/sound/pci/asihpi/hpioctl.c index 65fcf47..e0cff0c 100644 --- a/sound/pci/asihpi/hpioctl.c +++ b/sound/pci/asihpi/hpioctl.c @@ -107,7 +107,6 @@ long asihpi_hpi_ioctl(struct file *file, unsigned int cmd, unsigned long arg) union hpi_response_buffer_v1 *hr; u16 res_max_size; u32 uncopied_bytes; - struct hpi_adapter *pa = NULL; int err = 0; if (cmd != HPI_IOCTL_LINUX) @@ -182,6 +181,7 @@ long asihpi_hpi_ioctl(struct file *file, unsigned int cmd, unsigned long arg) /* -1=no data 0=read from user mem, 1=write to user mem */ int wrflag = -1; u32 adapter = hm->h.adapter_index; + struct hpi_adapter *pa = &adapters[adapter]; if ((adapter > HPI_MAX_ADAPTERS) || (!pa->type)) { hpi_init_response(&hr->r0, HPI_OBJ_ADAPTER, @@ -197,9 +197,7 @@ long asihpi_hpi_ioctl(struct file *file, unsigned int cmd, unsigned long arg) goto out; } - pa = &adapters[adapter]; - - if (mutex_lock_interruptible(&adapters[adapter].mutex)) { + if (mutex_lock_interruptible(&pa->mutex)) { err = -EINTR; goto out; } @@ -235,8 +233,7 @@ long asihpi_hpi_ioctl(struct file *file, unsigned int cmd, unsigned long arg) "stream buffer size %d\n", size); - mutex_unlock(&adapters - [adapter].mutex); + mutex_unlock(&pa->mutex); err = -EINVAL; goto out; } @@ -277,7 +274,7 @@ long asihpi_hpi_ioctl(struct file *file, unsigned int cmd, unsigned long arg) uncopied_bytes, size); } - mutex_unlock(&adapters[adapter].mutex); + mutex_unlock(&pa->mutex); } /* on return response size must be set */ -- 1.7.0.4

2 1

Re: [alsa-devel] Bug#635362: Realtek ALC889: HDA Intel and kernel 3.0 gives choppy sound
by Julian Sikorski 27 Jul '11

27 Jul '11

W dniu 26.07.2011 23:14, Colomban Wendling pisze: > Hi, > > Following a bug report started in Debian BTS, in bug 635362 [1], I was > asked to post here after testing upstream kernels and reproducing the bug. > > So, here's the problem: my HDA audio chip [2] started to output very > choppy sound with Linux kernel 3.0, while it worked perfectly well in > previous versions -- latest two I used being 2.6.38 and 2.6.39 from > Debian. Please note that simply booting on 2.6.39 works as usual, so > it's IMHO definitely a driver regression. > > I tried Debian's 3.0 as well as mainline 3.0 and git5 snapshots, and all > 3 were affected by the problem -- though the kernel log isn't exactly > the same. > > I attach three dmesg outputs, always took after trying to play a sound > (and hearing it choppy, it happens every time): > > dmesg-3.0.0-mainline: dmesg when running 3.0 mainline > dmesg-3.0.0-git5: when running 3.0-git5 snapshot > dmesg-3.0.0-git5-2: same as dmesg-3.0.0-git5 but configured without > CONFIG_SND_HDA_ENABLE_REALTEK_QUIRKS (doesn't change anything) > > If you want to see dmesg from the 2.6.39 kernel (where sound works well) > or Debian's 3.0, see respectively [3] and [4]. > > > Thanks by advance, and if you need any further details or testing, > please let me know. > > Regards, > Colomban > > > [1] http://bugs.debian.org/cgi-bin/bugreport.cgi?bug=635362 > [2] Realtek ALC889 on an MSI H55M-E33, see lspci in Debian bug [1] for > details > [3] > http://bugs.debian.org/cgi-bin/bugreport.cgi?msg=15;filename=dmesg-2.6.39-2… > [4] > http://bugs.debian.org/cgi-bin/bugreport.cgi?msg=15;filename=dmesg-3.0.0-1-… I am probably wrong, but did you try setting CONFIG_SND_HDA_BUFFER_SIZE to a larger value? Julian

2 2

[alsa-devel] [PATCH v3] ASoC: dapm - Add API call to query the widgets of valid DAPM paths.
by Liam Girdwood 26 Jul '11

26 Jul '11

In preparation for Dynamic PCM (AKA DSP) support. Add a DAPM API call to determine whether a DAPM audio path is valid between source and sink widgets and return the path widgets. This also takes into account all kcontrol mux and mixer settings in between the source and sink widgets to validate the audio path. This will be used by the Dynamic PCM core to determine the runtime DAI mappings between FE and BE DAIs in order to run PCM operations. This modifies the existing is_connected_input_ep() and is_connected_output_ep() by adding a new parameter to store any valid widgets in a list. We also add some extra debug to provide further insight into the DAPM path and widget discovery. Signed-off-by: Liam Girdwood <lrg(a)ti.com> --- include/sound/soc-dapm.h | 6 ++ sound/soc/soc-dapm.c | 184 ++++++++++++++++++++++++++++++++++++++++++---- 2 files changed, 176 insertions(+), 14 deletions(-) diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index 5833f01..a56ead2 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -310,6 +310,7 @@ struct snd_soc_dapm_path; struct snd_soc_dapm_pin; struct snd_soc_dapm_route; struct snd_soc_dapm_context; +struct snd_soc_dapm_widget_list; int dapm_reg_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event); @@ -375,6 +376,11 @@ int snd_soc_dapm_force_enable_pin(struct snd_soc_dapm_context *dapm, int snd_soc_dapm_ignore_suspend(struct snd_soc_dapm_context *dapm, const char *pin); +/* dapm path query */ +int snd_soc_dapm_get_connected_widgets(struct snd_soc_dapm_context *dapm, + const char *stream_name, struct snd_soc_dapm_widget_list **list, + int stream); + /* dapm widget types */ enum snd_soc_dapm_type { snd_soc_dapm_input = 0, /* input pin */ diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 80f97b2..120df5b 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -605,11 +605,51 @@ static int snd_soc_dapm_suspend_check(struct snd_soc_dapm_widget *widget) } } +/* add widget to list if it's not already in the list */ +static int dapm_list_add_widget(struct snd_soc_dapm_widget_list **list, + struct snd_soc_dapm_widget *w) +{ + struct snd_soc_dapm_widget_list *wlist; + int wlistsize, wlistentries, i; + + if (*list == NULL) + return -EINVAL; + + wlist = *list; + + /* is this widget already in the list */ + for (i = 0; i < wlist->num_widgets; i++) { + if (wlist->widgets[i] == w) + return 0; + } + + /* allocate some new space */ + wlistentries = wlist->num_widgets + 1; + wlistsize = sizeof(struct snd_soc_dapm_widget_list) + + wlistentries * sizeof(struct snd_soc_dapm_widget *); + *list = krealloc(wlist, wlistsize, GFP_KERNEL); + if (*list == NULL) { + dev_err(w->dapm->dev, "can't allocate widget list for %s\n", + w->name); + return -ENOMEM; + } + wlist = *list; + + /* insert the widget */ + dev_dbg(w->dapm->dev, "added %s in widget list pos %d\n", + w->name, wlist->num_widgets); + + wlist->widgets[wlist->num_widgets] = w; + wlist->num_widgets++; + return 1; +} + /* * Recursively check for a completed path to an active or physically connected * output widget. Returns number of complete paths. */ -static int is_connected_output_ep(struct snd_soc_dapm_widget *widget) +static int is_connected_output_ep(struct snd_soc_dapm_widget *widget, + struct snd_soc_dapm_widget_list **list) { struct snd_soc_dapm_path *path; int con = 0; @@ -638,15 +678,35 @@ static int is_connected_output_ep(struct snd_soc_dapm_widget *widget) } list_for_each_entry(path, &widget->sinks, list_source) { - if (path->weak) - continue; if (path->walked) continue; + dev_dbg(widget->dapm->dev," %c : %s -> %s -> %s : %c\n", + path->sink && path->connect ? '*' : ' ', + widget->name, path->name, path->sink->name, + path->weak ? 'w': ' '); + + if (path->weak) + continue; + if (path->sink && path->connect) { + path->walked = 1; - con += is_connected_output_ep(path->sink); + + /* do we need to add this widget to the list ? */ + if (list) { + int err; + + err = dapm_list_add_widget(list, path->sink); + if (err < 0) { + dev_err(widget->dapm->dev, "could not add widget %s\n", + widget->name); + return con; + } + } + + con += is_connected_output_ep(path->sink, list); } } @@ -657,7 +717,8 @@ static int is_connected_output_ep(struct snd_soc_dapm_widget *widget) * Recursively check for a completed path to an active or physically connected * input widget. Returns number of complete paths. */ -static int is_connected_input_ep(struct snd_soc_dapm_widget *widget) +static int is_connected_input_ep(struct snd_soc_dapm_widget *widget, + struct snd_soc_dapm_widget_list **list) { struct snd_soc_dapm_path *path; int con = 0; @@ -691,21 +752,116 @@ static int is_connected_input_ep(struct snd_soc_dapm_widget *widget) } list_for_each_entry(path, &widget->sources, list_sink) { - if (path->weak) - continue; if (path->walked) continue; + if (path->source) + dev_vdbg(widget->dapm->dev," %c : %s <- %s <- %s : %c\n", + path->source && path->connect ? '*' : ' ', + widget->name, path->name, path->source->name, + path->weak ? 'w': ' '); + + if (path->weak) + continue; + if (path->source && path->connect) { + path->walked = 1; - con += is_connected_input_ep(path->source); + + /* do we need to add this widget to the list ? */ + if (list) { + int err; + + err = dapm_list_add_widget(list, path->source); + if (err < 0) { + dev_err(widget->dapm->dev, "could not add widget %s\n", + widget->name); + return con; + } + } + + con += is_connected_input_ep(path->source, list); } } return con; } +static int dapm_get_playback_paths(struct snd_soc_dapm_context *dapm, + struct snd_soc_dapm_widget *root, + struct snd_soc_dapm_widget_list **list) +{ + int paths; + + dev_dbg(dapm->dev, "Playback: checking paths from %s\n",root->name); + paths = is_connected_output_ep(root, list); + dev_dbg(dapm->dev, "Playback: found %d paths from %s\n", paths, root->name); + + return paths; +} + +static int dapm_get_capture_paths(struct snd_soc_dapm_context *dapm, + struct snd_soc_dapm_widget *root, + struct snd_soc_dapm_widget_list **list) +{ + int paths; + + dev_dbg(dapm->dev, "Capture: checking paths from %s\n",root->name); + paths = is_connected_input_ep(root, list); + dev_dbg(dapm->dev, "Capture: found %d paths from %s\n", paths, root->name); + + return paths; +} + +/** + * snd_soc_dapm_get_connected_widgets - query audio path and it's widgets. + * @dapm: the dapm context. + * @stream_name: stream name. + * @list: list of active widgets for this stream. + * @stream: stream direction. + * + * Queries DAPM graph as to whether an valid audio stream path exists for + * the initial stream specified by name. This takes into account + * current mixer and mux kcontrol settings. Creates list of valid widgets. + * + * Returns the number of valid paths or negative error. + */ +int snd_soc_dapm_get_connected_widgets(struct snd_soc_dapm_context *dapm, + const char *stream_name, struct snd_soc_dapm_widget_list **list, + int stream) +{ + struct snd_soc_dapm_widget *w; + enum snd_soc_dapm_type type; + int paths; + + if (stream == SNDRV_PCM_STREAM_PLAYBACK) + type = snd_soc_dapm_aif_in; + else + type = snd_soc_dapm_aif_out; + + /* get stream root widget AIF from stream string and direction */ + list_for_each_entry(w, &dapm->card->widgets, list) { + + if (w->id != type) + continue; + + if (!strcmp(w->sname, stream_name)) + goto found; + } + dev_err(dapm->dev, "root widget for %s not found\n", stream_name); + return 0; + +found: + if (stream == SNDRV_PCM_STREAM_PLAYBACK) + paths = dapm_get_playback_paths(dapm, w, list); + else + paths = dapm_get_capture_paths(dapm, w, list); + + dapm_clear_walk(dapm); + return paths; +} + /* * Handler for generic register modifier widget. */ @@ -732,9 +888,9 @@ static int dapm_generic_check_power(struct snd_soc_dapm_widget *w) { int in, out; - in = is_connected_input_ep(w); + in = is_connected_input_ep(w, NULL); dapm_clear_walk(w->dapm); - out = is_connected_output_ep(w); + out = is_connected_output_ep(w, NULL); dapm_clear_walk(w->dapm); return out != 0 && in != 0; } @@ -745,7 +901,7 @@ static int dapm_adc_check_power(struct snd_soc_dapm_widget *w) int in; if (w->active) { - in = is_connected_input_ep(w); + in = is_connected_input_ep(w, NULL); dapm_clear_walk(w->dapm); return in != 0; } else { @@ -759,7 +915,7 @@ static int dapm_dac_check_power(struct snd_soc_dapm_widget *w) int out; if (w->active) { - out = is_connected_output_ep(w); + out = is_connected_output_ep(w, NULL); dapm_clear_walk(w->dapm); return out != 0; } else { @@ -1315,9 +1471,9 @@ static ssize_t dapm_widget_power_read_file(struct file *file, if (!buf) return -ENOMEM; - in = is_connected_input_ep(w); + in = is_connected_input_ep(w, NULL); dapm_clear_walk(w->dapm); - out = is_connected_output_ep(w); + out = is_connected_output_ep(w, NULL); dapm_clear_walk(w->dapm); ret = snprintf(buf, PAGE_SIZE, "%s: %s in %d out %d", -- 1.7.4.1

2 5

[alsa-devel] CX20590 - problems
by wouerner 26 Jul '11

26 Jul '11

hi, i'm from brazil, sorry bad english I have sony vaio vpceg15fb, internal mic - not work headphone jack - not work front speaker - work fine mic jack - work fine shell _> head -n 1 /proc/asound/card0/codec* ==> /proc/asound/card0/codec#0 <== Codec: Conexant CX20590 ==> /proc/asound/card0/codec#3 <== Codec: Intel CougarPoint HDMI not supported yet? i tried all tutorials in internet. thanks.

1 0

[alsa-devel] [PATCH] ASoC: core - make IO calls return a valid errno on failure.
by Liam Girdwood 26 Jul '11

26 Jul '11

IO call failures should return an errno instead of -1 Signed-off-by: Liam Girdwood <lrg(a)ti.com> --- sound/soc/soc-core.c | 4 ++-- sound/soc/soc-dapm.c | 4 ++-- sound/soc/soc-io.c | 6 +++--- 3 files changed, 7 insertions(+), 7 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 159f144..fcf69f4 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1661,7 +1661,7 @@ int snd_soc_platform_read(struct snd_soc_platform *platform, if (!platform->driver->read) { dev_err(platform->dev, "platform has no read back\n"); - return -1; + return -EINVAL; } ret = platform->driver->read(platform, reg); @@ -1677,7 +1677,7 @@ int snd_soc_platform_write(struct snd_soc_platform *platform, { if (!platform->driver->write) { dev_err(platform->dev, "platform has no write back\n"); - return -1; + return -EINVAL; } dev_dbg(platform->dev, "write %x = %x\n", reg, val); diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 3956670..45141fa 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -132,7 +132,7 @@ static int soc_widget_read(struct snd_soc_dapm_widget *w, int reg) return snd_soc_platform_read(w->platform, reg); dev_err(w->dapm->dev, "no valid widget read method\n"); - return -1; + return -EINVAL; } static int soc_widget_write(struct snd_soc_dapm_widget *w, int reg, int val) @@ -143,7 +143,7 @@ static int soc_widget_write(struct snd_soc_dapm_widget *w, int reg, int val) return snd_soc_platform_write(w->platform, reg, val); dev_err(w->dapm->dev, "no valid widget write method\n"); - return -1; + return -EINVAL; } static int soc_widget_update_bits(struct snd_soc_dapm_widget *w, diff --git a/sound/soc/soc-io.c b/sound/soc/soc-io.c index cca490c..166186c 100644 --- a/sound/soc/soc-io.c +++ b/sound/soc/soc-io.c @@ -41,7 +41,7 @@ static int do_hw_write(struct snd_soc_codec *codec, unsigned int reg, !codec->cache_bypass) { ret = snd_soc_cache_write(codec, reg, value); if (ret < 0) - return -1; + return ret; } if (codec->cache_only) { @@ -67,7 +67,7 @@ static unsigned int hw_read(struct snd_soc_codec *codec, unsigned int reg) snd_soc_codec_volatile_register(codec, reg) || codec->cache_bypass) { if (codec->cache_only) - return -1; + return -EINVAL; BUG_ON(!codec->hw_read); return codec->hw_read(codec, reg); @@ -75,7 +75,7 @@ static unsigned int hw_read(struct snd_soc_codec *codec, unsigned int reg) ret = snd_soc_cache_read(codec, reg, &val); if (ret < 0) - return -1; + return ret; return val; } -- 1.7.4.1

2 3

[alsa-devel] [PATCH] Cirrus Logic CS421x support
by Tim Howe 26 Jul '11

26 Jul '11

This update includes the changes necessary for supporting the CS421x family of codecs. Previously this file only supported the CS420x family of codecs. This file also contains init verbs to correct several issues in the CS421x hardware. Behavior between the CS421x and CS420x codec families is similar, so several functions have been reused with "if" statements to determine which codec family (CS421x or CS420x) is present. Also, this file will be updated sometime in the near future in order to add support for a system using CS421x that requires mono mix on the speaker output only. Signed-off-by: Tim Howe <tim.howe(a)cirrus.com> --- sound/pci/hda/patch_cirrus.c | 786 +++++++++++++++++++++++++++++++++++++++--- 1 files changed, 745 insertions(+), 41 deletions(-) diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c index 7f93739..27dbd5c 100644 --- a/sound/pci/hda/patch_cirrus.c +++ b/sound/pci/hda/patch_cirrus.c @@ -25,6 +25,7 @@ #include <sound/core.h> #include "hda_codec.h" #include "hda_local.h" +#include <sound/tlv.h> /* */ @@ -61,9 +62,15 @@ struct cs_spec { unsigned int hp_detect:1; unsigned int mic_detect:1; + /* CS421x */ + unsigned int spdif_detect:1; + unsigned int sense_b:1; + hda_nid_t vendor_nid; + struct hda_input_mux input_mux; + unsigned int last_input; }; -/* available models */ +/* available models with CS420x */ enum { CS420X_MBP53, CS420X_MBP55, @@ -72,6 +79,12 @@ enum { CS420X_MODELS }; +/* CS421x boards */ +enum { + CS421X_CDB4210, + CS421X_MODELS +}; + /* Vendor-specific processing widget */ #define CS420X_VENDOR_NID 0x11 #define CS_DIG_OUT1_PIN_NID 0x10 @@ -111,21 +124,42 @@ enum { /* 0x0009 - 0x0014 -> 12 test regs */ /* 0x0015 - visibility reg */ +/* + * Cirrus Logic CS4210 + * + * 1 DAC => HP(sense) / Speakers, + * 1 ADC <= LineIn(sense) / MicIn / DMicIn, + * 1 SPDIF OUT => SPDIF Trasmitter(sense) +*/ +#define CS4210_DAC_NID 0x02 +#define CS4210_ADC_NID 0x03 +#define CS421X_VENDOR_NID 0x0B +#define CS421X_DMIC_PIN_NID 0x09 /* Port E */ +#define CS421X_SPDIF_PIN_NID 0x0A /* Port H */ + +#define CS421X_IDX_DEV_CFG 0x01 +#define CS421X_IDX_ADC_CFG 0x02 +#define CS421X_IDX_DAC_CFG 0x03 +#define CS421X_IDX_SPK_CTL 0x04 + +#define SPDIF_EVENT 0x04 static inline int cs_vendor_coef_get(struct hda_codec *codec, unsigned int idx) { - snd_hda_codec_write(codec, CS420X_VENDOR_NID, 0, + struct cs_spec *spec = codec->spec; + snd_hda_codec_write(codec, spec->vendor_nid, 0, AC_VERB_SET_COEF_INDEX, idx); - return snd_hda_codec_read(codec, CS420X_VENDOR_NID, 0, + return snd_hda_codec_read(codec, spec->vendor_nid, 0, AC_VERB_GET_PROC_COEF, 0); } static inline void cs_vendor_coef_set(struct hda_codec *codec, unsigned int idx, unsigned int coef) { - snd_hda_codec_write(codec, CS420X_VENDOR_NID, 0, + struct cs_spec *spec = codec->spec; + snd_hda_codec_write(codec, spec->vendor_nid, 0, AC_VERB_SET_COEF_INDEX, idx); - snd_hda_codec_write(codec, CS420X_VENDOR_NID, 0, + snd_hda_codec_write(codec, spec->vendor_nid, 0, AC_VERB_SET_PROC_COEF, coef); } @@ -343,18 +377,44 @@ static hda_nid_t get_adc(struct hda_codec *codec, hda_nid_t pin, { int i; hda_nid_t nid; + struct cs_spec *spec = codec->spec; nid = codec->start_nid; for (i = 0; i < codec->num_nodes; i++, nid++) { unsigned int type; - int idx; - type = get_wcaps_type(get_wcaps(codec, nid)); - if (type != AC_WID_AUD_IN) - continue; - idx = snd_hda_get_conn_index(codec, nid, pin, 0); - if (idx >= 0) { - *idxp = idx; - return nid; + int j, nums; + /* specific to CS421x */ + if (spec->vendor_nid == CS421X_VENDOR_NID) { + hda_nid_t pins[HDA_MAX_NUM_INPUTS]; + type = (get_wcaps(codec, nid) & AC_WCAP_TYPE) + >> AC_WCAP_TYPE_SHIFT; + if (type != AC_WID_AUD_IN) + continue; + nums = snd_hda_get_connections(codec, nid, pins, + ARRAY_SIZE(pins)); + if (nums <= 0) + continue; + for (j = 0; j < nums; j++) { + if (pins[j] == pin) { + *idxp = j; + return nid; + } + } + } else { + hda_nid_t pins[2]; + type = get_wcaps_type(get_wcaps(codec, nid)); + if (type != AC_WID_AUD_IN) + continue; + nums = snd_hda_get_connections(codec, nid, pins, + ARRAY_SIZE(pins)); + if (nums <= 0) + continue; + for (j = 0; j < nums; j++) { + if (pins[j] == pin) { + *idxp = j; + return nid; + } + } } } return 0; @@ -835,6 +895,8 @@ static int build_digital_input(struct hda_codec *codec) /* * auto-mute and auto-mic switching + * CS421x auto-output redirecting + * HP/SPK/SPDIF */ static void cs_automute(struct hda_codec *codec) @@ -842,9 +904,25 @@ static void cs_automute(struct hda_codec *codec) struct cs_spec *spec = codec->spec; struct auto_pin_cfg *cfg = &spec->autocfg; unsigned int hp_present; + unsigned int spdif_present; hda_nid_t nid; int i; + spdif_present = 0; + if (cfg->dig_outs) { + nid = cfg->dig_out_pins[0]; + if (is_jack_detectable(codec, nid)) { + /* + TODO: SPDIF output redirect when SENSE_B is enabled. + Shared (SENSE_A) jack (e.g HP/mini-TOSLINK) + assumed. + */ + if (snd_hda_jack_detect(codec, nid) + /* && spec->sense_b */) + spdif_present = 1; + } + } + hp_present = 0; for (i = 0; i < cfg->hp_outs; i++) { nid = cfg->hp_pins[i]; @@ -854,11 +932,19 @@ static void cs_automute(struct hda_codec *codec) if (hp_present) break; } + + /* mute speakers if spdif or hp jack is plugged in */ for (i = 0; i < cfg->speaker_outs; i++) { nid = cfg->speaker_pins[i]; snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, hp_present ? 0 : PIN_OUT); + /* detect on spdif is specific to CS421x */ + if (spec->vendor_nid == CS421X_VENDOR_NID) { + snd_hda_codec_write(codec, nid, 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, + spdif_present ? 0 : PIN_OUT); + } } if (spec->board_config == CS420X_MBP53 || spec->board_config == CS420X_MBP55 || @@ -867,21 +953,62 @@ static void cs_automute(struct hda_codec *codec) snd_hda_codec_write(codec, 0x01, 0, AC_VERB_SET_GPIO_DATA, gpio); } + + /* specific to CS421x */ + if (spec->vendor_nid == CS421X_VENDOR_NID) { + /* mute HPs if spdif jack (SENSE_B) is present */ + for (i = 0; i < cfg->hp_outs; i++) { + nid = cfg->hp_pins[i]; + snd_hda_codec_write(codec, nid, 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, + (spdif_present && spec->sense_b) ? 0 : PIN_HP); + } + + /* SPDIF TX on/off */ + if (cfg->dig_outs) { + nid = cfg->dig_out_pins[0]; + snd_hda_codec_write(codec, nid, 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, + spdif_present ? PIN_OUT : 0); + + } + /* Update board GPIOs if neccessary ... */ + } } +/* + * Auto-input redirect for CS421x + * Switch max 3 inputs of a single ADC (nid 3) +*/ + static void cs_automic(struct hda_codec *codec) { struct cs_spec *spec = codec->spec; struct auto_pin_cfg *cfg = &spec->autocfg; hda_nid_t nid; unsigned int present; - + nid = cfg->inputs[spec->automic_idx].pin; present = snd_hda_jack_detect(codec, nid); - if (present) - change_cur_input(codec, spec->automic_idx, 0); - else - change_cur_input(codec, !spec->automic_idx, 0); + + /* specific to CS421x, single ADC */ + if (spec->vendor_nid == CS421X_VENDOR_NID) { + if (present) { + spec->last_input = spec->cur_input; + spec->cur_input = spec->automic_idx; + } else { + spec->cur_input = spec->last_input; + } + + snd_hda_codec_write_cache(codec, spec->cur_adc, 0, + AC_VERB_SET_CONNECT_SEL, + spec->adc_idx[spec->cur_input]); + } else { + if (present) + change_cur_input(codec, spec->automic_idx, 0); + else + change_cur_input(codec, !spec->automic_idx, 0); + } } /* @@ -911,23 +1038,28 @@ static void init_output(struct hda_codec *codec) for (i = 0; i < cfg->line_outs; i++) snd_hda_codec_write(codec, cfg->line_out_pins[i], 0, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT); + /* HP */ for (i = 0; i < cfg->hp_outs; i++) { hda_nid_t nid = cfg->hp_pins[i]; snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP); if (!cfg->speaker_outs) continue; - if (is_jack_detectable(codec, nid)) { + if (get_wcaps(codec, nid) & AC_WCAP_UNSOL_CAP) { snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | HP_EVENT); spec->hp_detect = 1; } } + + /* Speaker */ for (i = 0; i < cfg->speaker_outs; i++) snd_hda_codec_write(codec, cfg->speaker_pins[i], 0, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT); - if (spec->hp_detect) + + /* SPDIF is enabled on presence detect for CS421x */ + if (spec->hp_detect || spec->spdif_detect) cs_automute(codec); } @@ -961,19 +1093,31 @@ static void init_input(struct hda_codec *codec) AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | MIC_EVENT); } - change_cur_input(codec, spec->cur_input, 1); - if (spec->mic_detect) - cs_automic(codec); - - coef = 0x000a; /* ADC1/2 - Digital and Analog Soft Ramp */ - if (is_active_pin(codec, CS_DMIC2_PIN_NID)) - coef |= 0x0500; /* DMIC2 enable 2 channels, disable GPIO1 */ - if (is_active_pin(codec, CS_DMIC1_PIN_NID)) - coef |= 0x1800; /* DMIC1 enable 2 channels, disable GPIO0 - * No effect if SPDIF_OUT2 is selected in - * IDX_SPDIF_CTL. - */ - cs_vendor_coef_set(codec, IDX_ADC_CFG, coef); + /* specific to CS421x */ + if (spec->vendor_nid == CS421X_VENDOR_NID) { + if (spec->mic_detect) + cs_automic(codec); + else { + spec->cur_adc = spec->adc_nid[spec->cur_input]; + snd_hda_codec_write(codec, spec->cur_adc, 0, + AC_VERB_SET_CONNECT_SEL, + spec->adc_idx[spec->cur_input]); + } + } else { + change_cur_input(codec, spec->cur_input, 1); + if (spec->mic_detect) + cs_automic(codec); + + coef = 0x000a; /* ADC1/2 - Digital and Analog Soft Ramp */ + if (is_active_pin(codec, CS_DMIC2_PIN_NID)) + coef |= 0x0500; /* DMIC2 2 chan on, GPIO1 off */ + if (is_active_pin(codec, CS_DMIC1_PIN_NID)) + coef |= 0x1800; /* DMIC1 2 chan on, GPIO0 off + * No effect if SPDIF_OUT2 is + * selected in IDX_SPDIF_CTL. + */ + cs_vendor_coef_set(codec, IDX_ADC_CFG, coef); + } } static const struct hda_verb cs_coef_init_verbs[] = { @@ -1173,7 +1317,7 @@ struct cs_pincfg { u32 val; }; -static const struct cs_pincfg mbp53_pincfgs[] = { +static struct cs_pincfg mbp53_pincfgs[] = { { 0x09, 0x012b4050 }, { 0x0a, 0x90100141 }, { 0x0b, 0x90100140 }, @@ -1187,7 +1331,7 @@ static const struct cs_pincfg mbp53_pincfgs[] = { {} /* terminator */ }; -static const struct cs_pincfg mbp55_pincfgs[] = { +static struct cs_pincfg mbp55_pincfgs[] = { { 0x09, 0x012b4030 }, { 0x0a, 0x90100121 }, { 0x0b, 0x90100120 }, @@ -1201,7 +1345,7 @@ static const struct cs_pincfg mbp55_pincfgs[] = { {} /* terminator */ }; -static const struct cs_pincfg imac27_pincfgs[] = { +static struct cs_pincfg imac27_pincfgs[] = { { 0x09, 0x012b4050 }, { 0x0a, 0x90100140 }, { 0x0b, 0x90100142 }, @@ -1215,22 +1359,22 @@ static const struct cs_pincfg imac27_pincfgs[] = { {} /* terminator */ }; -static const struct cs_pincfg *cs_pincfgs[CS420X_MODELS] = { +static struct cs_pincfg *cs_pincfgs[CS420X_MODELS] = { [CS420X_MBP53] = mbp53_pincfgs, [CS420X_MBP55] = mbp55_pincfgs, [CS420X_IMAC27] = imac27_pincfgs, }; -static void fix_pincfg(struct hda_codec *codec, int model) +static void fix_pincfg(struct hda_codec *codec, int model, + struct cs_pincfg **pin_configs) { - const struct cs_pincfg *cfg = cs_pincfgs[model]; + const struct cs_pincfg *cfg = pin_configs[model]; if (!cfg) return; for (; cfg->nid; cfg++) snd_hda_codec_set_pincfg(codec, cfg->nid, cfg->val); } - static int patch_cs420x(struct hda_codec *codec) { struct cs_spec *spec; @@ -1241,11 +1385,13 @@ static int patch_cs420x(struct hda_codec *codec) return -ENOMEM; codec->spec = spec; + spec->vendor_nid = CS420X_VENDOR_NID; + spec->board_config = snd_hda_check_board_config(codec, CS420X_MODELS, cs420x_models, cs420x_cfg_tbl); if (spec->board_config >= 0) - fix_pincfg(codec, spec->board_config); + fix_pincfg(codec, spec->board_config, cs_pincfgs); switch (spec->board_config) { case CS420X_IMAC27: @@ -1272,6 +1418,562 @@ static int patch_cs420x(struct hda_codec *codec) return err; } +/* + * Cirrus Logic CS4210 + * + * 1 DAC => HP(sense) / Speakers, + * 1 ADC <= LineIn(sense) / MicIn / DMicIn, + * 1 SPDIF OUT => SPDIF Trasmitter(sense) +*/ + +/* CS4210 board names */ +static const char *cs421x_models[CS421X_MODELS] = { + [CS421X_CDB4210] = "cdb4210", +}; + +static struct snd_pci_quirk cs421x_cfg_tbl[] = { + /* Test Intel board + CDB2410 */ + SND_PCI_QUIRK(0x8086, 0x5001, "DP45SG/CDB4210", CS421X_CDB4210), + {} /* terminator */ +}; + +/* CS4210 board pinconfigs */ +/* Default CS4210 (CDB4210)*/ +static struct cs_pincfg cdb4210_pincfgs[] = { + { 0x05, 0x0321401f }, + { 0x06, 0x90170010 }, + { 0x07, 0x03813031 }, + { 0x08, 0xb7a70037 }, + { 0x09, 0xb7a6003e }, + { 0x0a, 0x034510f0 }, + {} /* terminator */ +}; + +static struct cs_pincfg *cs421x_pincfgs[CS421X_MODELS] = { + [CS421X_CDB4210] = cdb4210_pincfgs, +}; + +static struct hda_verb cs421x_coef_init_verbs[] = { + {0x0B, AC_VERB_SET_PROC_STATE, 1}, + {0x0B, AC_VERB_SET_COEF_INDEX, CS421X_IDX_DEV_CFG}, + /* + Disable Coefficient Index Auto-Increment(DAI)=1, + PDREF=0 + */ + {0x0B, AC_VERB_SET_PROC_COEF, 0x0001 }, + + {0x0B, AC_VERB_SET_COEF_INDEX, CS421X_IDX_ADC_CFG}, + /* ADC SZCMode = Digital Soft Ramp */ + {0x0B, AC_VERB_SET_PROC_COEF, 0x0002 }, + + {0x0B, AC_VERB_SET_COEF_INDEX, CS421X_IDX_DAC_CFG}, + {0x0B, AC_VERB_SET_PROC_COEF, + (0x0002 /* DAC SZCMode = Digital Soft Ramp */ + | 0x0004 /* Mute DAC on FIFO error */ + | 0x0008 /* Enable DAC High Pass Filter */ + )}, + {} /* terminator */ +}; + +/* Errata: CS4210 rev A1 Silicon + * + * http://www.cirrus.com/en/pubs/errata/ + * + * Description: + * 1. Performance degredation is present in the ADC. + * 2. Speaker output is not completely muted upon HP detect. + * 3. Noise is present when clipping occurs on the amplified + * speaker outputs. + * + * Workaround: + * The following verb sequence written to the registers during + * initialization will correct the issues listed above. + */ + +static struct hda_verb cs421x_coef_init_verbs_A1_silicon_fixes[] = { + {0x0B, AC_VERB_SET_PROC_STATE, 0x01}, /* VPW: processing on */ + + {0x0B, AC_VERB_SET_COEF_INDEX, 0x0006}, + {0x0B, AC_VERB_SET_PROC_COEF, 0x9999}, /* Test mode: on */ + + {0x0B, AC_VERB_SET_COEF_INDEX, 0x000A}, + {0x0B, AC_VERB_SET_PROC_COEF, 0x14CB}, /* Chop double */ + + {0x0B, AC_VERB_SET_COEF_INDEX, 0x0011}, + {0x0B, AC_VERB_SET_PROC_COEF, 0xA2D0}, /* Increase ADC current */ + + {0x0B, AC_VERB_SET_COEF_INDEX, 0x001A}, + {0x0B, AC_VERB_SET_PROC_COEF, 0x02A9}, /* Mute speaker */ + + {0x0B, AC_VERB_SET_COEF_INDEX, 0x001B}, + {0x0B, AC_VERB_SET_PROC_COEF, 0X1006}, /* Remove noise */ + + {} /* terminator */ +}; + +/* Speaker Amp Gain is controlled by the vendor widget's coef 4 */ +static const DECLARE_TLV_DB_SCALE(cs421x_speaker_boost_db_scale, 900, 300, 0); + +static int cs421x_boost_vol_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 1; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = 3; + return 0; +} + +static int cs421x_boost_vol_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + + ucontrol->value.integer.value[0] = + cs_vendor_coef_get(codec, CS421X_IDX_SPK_CTL) & 0x0003; + return 0; +} + +static int cs421x_boost_vol_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + + unsigned int vol = ucontrol->value.integer.value[0]; + unsigned int coef = + cs_vendor_coef_get(codec, CS421X_IDX_SPK_CTL); + unsigned int original_coef = coef; + + coef &= ~0x0003; + coef |= (vol & 0x0003); + if (original_coef == coef) + return 0; + else { + cs_vendor_coef_set(codec, CS421X_IDX_SPK_CTL, coef); + return 1; + } +} + +static const struct snd_kcontrol_new cs421x_speaker_bost_ctl = { + + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE | + SNDRV_CTL_ELEM_ACCESS_TLV_READ), + .name = "Speaker Boost Playback Volume", + .info = cs421x_boost_vol_info, + .get = cs421x_boost_vol_get, + .put = cs421x_boost_vol_put, + .tlv = { .p = cs421x_speaker_boost_db_scale }, +}; + +static void cs421x_pinmux_init(struct hda_codec *codec) +{ + struct cs_spec *spec = codec->spec; + unsigned int def_conf, coef; + + /* GPIO, DMIC_SCL, DMIC_SDA and SENSE_B are multiplexed */ + coef = cs_vendor_coef_get(codec, CS421X_IDX_DEV_CFG); + + if (spec->gpio_mask) + coef |= 0x0008; /* B1,B2 are GPIOs */ + else + coef &= ~0x0008; + + if (spec->sense_b) + coef |= 0x0010; /* B2 is SENSE_B, not inverted */ + else + coef &= ~0x0010; + + cs_vendor_coef_set(codec, CS421X_IDX_DEV_CFG, coef); + + if ((spec->gpio_mask || spec->sense_b) && + is_active_pin(codec, CS421X_DMIC_PIN_NID)) { + + /* + GPIO or SENSE_B forced - disconnect the DMIC pin. + */ + def_conf = snd_hda_codec_get_pincfg(codec, CS421X_DMIC_PIN_NID); + def_conf &= ~AC_DEFCFG_PORT_CONN; + def_conf |= (AC_JACK_PORT_NONE << AC_DEFCFG_PORT_CONN_SHIFT); + snd_hda_codec_set_pincfg(codec, CS421X_DMIC_PIN_NID, def_conf); + } +} + +static void init_cs421x_digital(struct hda_codec *codec) +{ + struct cs_spec *spec = codec->spec; + struct auto_pin_cfg *cfg = &spec->autocfg; + int i; + + + for (i = 0; i < cfg->dig_outs; i++) { + hda_nid_t nid = cfg->dig_out_pins[i]; + if (!cfg->speaker_outs) + continue; + if (get_wcaps(codec, nid) & AC_WCAP_UNSOL_CAP) { + + snd_hda_codec_write(codec, nid, 0, + AC_VERB_SET_UNSOLICITED_ENABLE, + AC_USRSP_EN | SPDIF_EVENT); + spec->spdif_detect = 1; + } + } +} + +static int cs421x_init(struct hda_codec *codec) +{ + struct cs_spec *spec = codec->spec; + + snd_hda_sequence_write(codec, cs421x_coef_init_verbs); + snd_hda_sequence_write(codec, cs421x_coef_init_verbs_A1_silicon_fixes); + + cs421x_pinmux_init(codec); + + if (spec->gpio_mask) { + snd_hda_codec_write(codec, 0x01, 0, AC_VERB_SET_GPIO_MASK, + spec->gpio_mask); + snd_hda_codec_write(codec, 0x01, 0, AC_VERB_SET_GPIO_DIRECTION, + spec->gpio_dir); + snd_hda_codec_write(codec, 0x01, 0, AC_VERB_SET_GPIO_DATA, + spec->gpio_data); + } + + init_output(codec); + init_input(codec); + init_cs421x_digital(codec); + + return 0; +} + +/* + * CS4210 Input MUX (1 ADC) + */ +static int cs421x_mux_enum_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct cs_spec *spec = codec->spec; + + return snd_hda_input_mux_info(&spec->input_mux, uinfo); +} + +static int cs421x_mux_enum_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct cs_spec *spec = codec->spec; + + ucontrol->value.enumerated.item[0] = spec->cur_input; + return 0; +} + +static int cs421x_mux_enum_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct cs_spec *spec = codec->spec; + + return snd_hda_input_mux_put(codec, &spec->input_mux, ucontrol, + spec->adc_nid[0], &spec->cur_input); + +} + +static struct snd_kcontrol_new cs421x_capture_source = { + + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Capture Source", + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .info = cs421x_mux_enum_info, + .get = cs421x_mux_enum_get, + .put = cs421x_mux_enum_put, +}; + +static int cs421x_add_input_volume_control(struct hda_codec *codec, int item) +{ + struct cs_spec *spec = codec->spec; + struct auto_pin_cfg *cfg = &spec->autocfg; + const struct hda_input_mux *imux = &spec->input_mux; + hda_nid_t pin = cfg->inputs[item].pin; + struct snd_kcontrol *kctl; + u32 caps; + + if (!(get_wcaps(codec, pin) & AC_WCAP_IN_AMP)) + return 0; + + caps = query_amp_caps(codec, pin, HDA_INPUT); + caps = (caps & AC_AMPCAP_NUM_STEPS) >> AC_AMPCAP_NUM_STEPS_SHIFT; + if (caps <= 1) + return 0; + + return add_volume(codec, imux->items[item].label, 0, + HDA_COMPOSE_AMP_VAL(pin, 3, 0, HDA_INPUT), 1, &kctl); +} + +/* add a (input-boost) volume control to the given input pin */ +static int build_cs421x_input(struct hda_codec *codec) +{ + struct cs_spec *spec = codec->spec; + struct auto_pin_cfg *cfg = &spec->autocfg; + struct hda_input_mux *imux = &spec->input_mux; + int i, err, type_idx; + const char *label; + + if (!spec->num_inputs) + return 0; + + /* make bind-capture */ + spec->capture_bind[0] = make_bind_capture(codec, &snd_hda_bind_sw); + spec->capture_bind[1] = make_bind_capture(codec, &snd_hda_bind_vol); + for (i = 0; i < 2; i++) { + struct snd_kcontrol *kctl; + int n; + if (!spec->capture_bind[i]) + return -ENOMEM; + kctl = snd_ctl_new1(&cs_capture_ctls[i], codec); + if (!kctl) + return -ENOMEM; + kctl->private_value = (long)spec->capture_bind[i]; + err = snd_hda_ctl_add(codec, 0, kctl); + if (err < 0) + return err; + for (n = 0; n < AUTO_PIN_LAST; n++) { + if (!spec->adc_nid[n]) + continue; + err = snd_hda_add_nid(codec, kctl, 0, spec->adc_nid[n]); + if (err < 0) + return err; + } + } + + /* Add Input MUX Items + Capture Volume/Switch */ + for (i = 0; i < spec->num_inputs; i++) { + label = hda_get_autocfg_input_label(codec, cfg, i); + snd_hda_add_imux_item(imux, label, spec->adc_idx[i], &type_idx); + + err = cs421x_add_input_volume_control(codec, i); + if (err < 0) + return err; + } + + /* + Add 'Capture Source' Switch if + * 2 inputs and no mic detec + * 3 inputs + */ + if ((spec->num_inputs == 2 && !spec->mic_detect) || + (spec->num_inputs == 3)) { + + err = snd_hda_ctl_add(codec, spec->adc_nid[0], + snd_ctl_new1(&cs421x_capture_source, codec)); + if (err < 0) + return err; + } + + return 0; +} + +/* Single DAC (Mute/Gain) */ +static int build_cs421x_output(struct hda_codec *codec) +{ + hda_nid_t dac = CS4210_DAC_NID; + struct cs_spec *spec = codec->spec; + struct auto_pin_cfg *cfg = &spec->autocfg; + struct snd_kcontrol *kctl; + int err; + char *name = "HP/Speakers"; + + fix_volume_caps(codec, dac); + if (!spec->vmaster_sw) { + err = add_vmaster(codec, dac); + if (err < 0) + return err; + } + + err = add_mute(codec, name, 0, + HDA_COMPOSE_AMP_VAL(dac, 3, 0, HDA_OUTPUT), 0, &kctl); + if (err < 0) + return err; + err = snd_ctl_add_slave(spec->vmaster_sw, kctl); + if (err < 0) + return err; + + err = add_volume(codec, name, 0, + HDA_COMPOSE_AMP_VAL(dac, 3, 0, HDA_OUTPUT), 0, &kctl); + if (err < 0) + return err; + err = snd_ctl_add_slave(spec->vmaster_vol, kctl); + if (err < 0) + return err; + + if (cfg->speaker_outs) { + err = snd_hda_ctl_add(codec, 0, + snd_ctl_new1(&cs421x_speaker_bost_ctl, codec)); + if (err < 0) + return err; + } + return err; +} + +static int cs421x_build_controls(struct hda_codec *codec) +{ + int err; + + err = build_cs421x_output(codec); + if (err < 0) + return err; + err = build_cs421x_input(codec); + if (err < 0) + return err; + err = build_digital_output(codec); + if (err < 0) + return err; + return cs421x_init(codec); +} + +static void cs421x_unsol_event(struct hda_codec *codec, unsigned int res) +{ + switch ((res >> 26) & 0x3f) { + case HP_EVENT: + case SPDIF_EVENT: + cs_automute(codec); + break; + + case MIC_EVENT: + cs_automic(codec); + break; + } +} + +static int parse_cs421x_input(struct hda_codec *codec) +{ + struct cs_spec *spec = codec->spec; + struct auto_pin_cfg *cfg = &spec->autocfg; + int i; + + for (i = 0; i < cfg->num_inputs; i++) { + hda_nid_t pin = cfg->inputs[i].pin; + spec->adc_nid[i] = get_adc(codec, pin, &spec->adc_idx[i]); + spec->cur_input = spec->last_input = i; + spec->num_inputs++; + + /* check whether the automatic mic switch is available */ + if (is_ext_mic(codec, i) && cfg->num_inputs >= 2) { + spec->mic_detect = 1; + spec->automic_idx = i; + } + } + return 0; +} + +static int cs421x_parse_auto_config(struct hda_codec *codec) +{ + struct cs_spec *spec = codec->spec; + int err; + + err = snd_hda_parse_pin_def_config(codec, &spec->autocfg, NULL); + if (err < 0) + return err; + err = parse_output(codec); + if (err < 0) + return err; + err = parse_cs421x_input(codec); + if (err < 0) + return err; + err = parse_digital_output(codec); + if (err < 0) + return err; + return 0; +} + +#ifdef SND_HDA_NEEDS_RESUME +/* + Manage PDREF, when transitioning to D3hot + (DAC,ADC) -> D3, PDREF=1, AFG->D3 +*/ +static int cs421x_suspend(struct hda_codec *codec, pm_message_t state) +{ + unsigned int coef; + + snd_hda_shutup_pins(codec); + + snd_hda_codec_write(codec, CS4210_DAC_NID, 0, + AC_VERB_SET_POWER_STATE, AC_PWRST_D3); + snd_hda_codec_write(codec, CS4210_ADC_NID, 0, + AC_VERB_SET_POWER_STATE, AC_PWRST_D3); + + coef = cs_vendor_coef_get(codec, CS421X_IDX_DEV_CFG); + coef |= 0x0004; /* PDREF */ + cs_vendor_coef_set(codec, CS421X_IDX_DEV_CFG, coef); + + return 0; +} +#endif + +static struct hda_codec_ops cs4210_patch_ops = { + .build_controls = cs421x_build_controls, + .build_pcms = cs_build_pcms, + .init = cs421x_init, + .free = cs_free, + .unsol_event = cs421x_unsol_event, +#ifdef SND_HDA_NEEDS_RESUME + .suspend = cs421x_suspend, +#endif +}; + +static int patch_cs421x(struct hda_codec *codec) +{ + struct cs_spec *spec; + int err; + + spec = kzalloc(sizeof(*spec), GFP_KERNEL); + if (!spec) + return -ENOMEM; + codec->spec = spec; + + spec->vendor_nid = CS421X_VENDOR_NID; + + spec->board_config = + snd_hda_check_board_config(codec, CS421X_MODELS, + cs421x_models, cs421x_cfg_tbl); + if (spec->board_config >= 0) + fix_pincfg(codec, spec->board_config, cs421x_pincfgs); + /* + Setup GPIO/SENSE for each board (if used) + */ + switch (spec->board_config) { + case CS421X_CDB4210: + snd_printd("CS4210 board: %s\n", + cs421x_models[spec->board_config]); +/* spec->gpio_mask = 3; + spec->gpio_dir = 3; + spec->gpio_data = 3; +*/ + spec->sense_b = 1; + + break; + } + + /* + Update the GPIO/DMIC/SENSE_B pinmux before the configuration + is auto-parsed. If GPIO or SENSE_B is forced, DMIC input + is disabled. + */ + cs421x_pinmux_init(codec); + + err = cs421x_parse_auto_config(codec); + if (err < 0) + goto error; + + codec->patch_ops = cs4210_patch_ops; + + return 0; + + error: + kfree(codec->spec); + codec->spec = NULL; + return err; +} + /* * patch entries @@ -1279,11 +1981,13 @@ static int patch_cs420x(struct hda_codec *codec) static const struct hda_codec_preset snd_hda_preset_cirrus[] = { { .id = 0x10134206, .name = "CS4206", .patch = patch_cs420x }, { .id = 0x10134207, .name = "CS4207", .patch = patch_cs420x }, + { .id = 0x10134210, .name = "CS4210", .patch = patch_cs421x }, {} /* terminator */ }; MODULE_ALIAS("snd-hda-codec-id:10134206"); MODULE_ALIAS("snd-hda-codec-id:10134207"); +MODULE_ALIAS("snd-hda-codec-id:10134210"); MODULE_LICENSE("GPL"); MODULE_DESCRIPTION("Cirrus Logic HD-audio codec"); -- 1.7.0.4

2 2

[alsa-devel] snd_mask_min
by Tim Blechmann 26 Jul '11

26 Jul '11

hi all, i've got a problem using params_format(): it uses an inline function snd_mask_min, which is declared in sound/pcm_params.h, but this file is not included by sound/pcm.h, where it is defined. including it manually, causes a huge number of errors. it would be very helpful, if pcm.h would be self-contained. cheers, tim

2 3

[alsa-devel] [PATCH] alsa: lx6464es - initialization fixes
by Tim Blechmann 26 Jul '11

26 Jul '11

this using this fix, the driver is usable with non-rt kernels. the card initialization is moved from the prepare to the hw_params callback. also the error handling is improved. Signed-off-by: Tim Blechmann <tim(a)klingt.org> --- sound/pci/lx6464es/lx6464es.c | 72 +++++++++++++++++++++-------------------- sound/pci/lx6464es/lx6464es.h | 1 + sound/pci/lx6464es/lx_core.c | 43 +++++++++++++++---------- sound/pci/lx6464es/lx_core.h | 3 +- 4 files changed, 66 insertions(+), 53 deletions(-) diff --git a/sound/pci/lx6464es/lx6464es.c b/sound/pci/lx6464es/lx6464es.c index 1bd7a54..722212e 100644 --- a/sound/pci/lx6464es/lx6464es.c +++ b/sound/pci/lx6464es/lx6464es.c @@ -103,16 +103,14 @@ static int lx_set_granularity(struct lx6464es *chip, u32 gran); static int lx_hardware_open(struct lx6464es *chip, - struct snd_pcm_substream *substream) + struct snd_pcm_substream *substream, + int channels, snd_pcm_uframes_t period_size) { int err = 0; - struct snd_pcm_runtime *runtime = substream->runtime; - int channels = runtime->channels; int is_capture = (substream->stream == SNDRV_PCM_STREAM_CAPTURE); - snd_pcm_uframes_t period_size = runtime->period_size; - - snd_printd(LXP "allocating pipe for %d channels\n", channels); + snd_printk(LXP "allocating pipe for %d channels, %lu period size\n", + channels, period_size); err = lx_pipe_allocate(chip, 0, is_capture, channels); if (err < 0) { snd_printk(KERN_ERR LXP "allocating pipe failed\n"); @@ -130,14 +128,14 @@ static int lx_hardware_open(struct lx6464es *chip, } static int lx_hardware_start(struct lx6464es *chip, - struct snd_pcm_substream *substream) + struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params) { int err = 0; - struct snd_pcm_runtime *runtime = substream->runtime; int is_capture = (substream->stream == SNDRV_PCM_STREAM_CAPTURE); snd_printd(LXP "setting stream format\n"); - err = lx_stream_set_format(chip, runtime, 0, is_capture); + err = lx_stream_set_format(chip, hw_params, 0, is_capture); if (err < 0) { snd_printk(KERN_ERR LXP "setting stream format failed\n"); return err; @@ -295,13 +293,33 @@ static snd_pcm_uframes_t lx_pcm_stream_pointer(struct snd_pcm_substream return pos; } + static int lx_pcm_prepare(struct snd_pcm_substream *substream) { struct lx6464es *chip = snd_pcm_substream_chip(substream); int err = 0; - const int is_capture = (substream->stream == SNDRV_PCM_STREAM_CAPTURE); - snd_printdd("->lx_pcm_prepare\n"); + snd_printd("->lx_pcm_prepare %d\n", is_capture); + + mutex_lock(&chip->setup_mutex); + + if (chip->board_sample_rate != substream->runtime->rate) { + if (!err) + chip->board_sample_rate = substream->runtime->rate; + } + + mutex_unlock(&chip->setup_mutex); + snd_printd("<-lx_pcm_prepare\n"); + return err; +} + +static int lx_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params, int is_capture) +{ + struct lx6464es *chip = snd_pcm_substream_chip(substream); + int err = 0; + + snd_printd("->lx_pcm_hw_params\n"); mutex_lock(&chip->setup_mutex); @@ -322,14 +340,16 @@ static int lx_pcm_prepare(struct snd_pcm_substream *substream) } snd_printd(LXP "opening hardware\n"); - err = lx_hardware_open(chip, substream); + err = lx_hardware_open(chip, substream, + params_channels(hw_params), + params_period_size(hw_params)); if (err < 0) { snd_printk(KERN_ERR LXP "failed to open hardware. " "Error code %d\n", err); goto exit; } - err = lx_hardware_start(chip, substream); + err = lx_hardware_start(chip, substream, hw_params); if (err < 0) { snd_printk(KERN_ERR LXP "failed to start hardware. " "Error code %d\n", err); @@ -338,26 +358,6 @@ static int lx_pcm_prepare(struct snd_pcm_substream *substream) chip->hardware_running[is_capture] = 1; - if (chip->board_sample_rate != substream->runtime->rate) { - if (!err) - chip->board_sample_rate = substream->runtime->rate; - } - -exit: - mutex_unlock(&chip->setup_mutex); - return err; -} - -static int lx_pcm_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *hw_params, int is_capture) -{ - struct lx6464es *chip = snd_pcm_substream_chip(substream); - int err = 0; - - snd_printdd("->lx_pcm_hw_params\n"); - - mutex_lock(&chip->setup_mutex); - /* set dma buffer */ err = snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(hw_params)); @@ -367,6 +367,7 @@ static int lx_pcm_hw_params(struct snd_pcm_substream *substream, else chip->playback_stream.stream = substream; +exit: mutex_unlock(&chip->setup_mutex); return err; } @@ -389,7 +390,7 @@ static int lx_pcm_hw_free(struct snd_pcm_substream *substream) int err = 0; int is_capture = (substream->stream == SNDRV_PCM_STREAM_CAPTURE); - snd_printdd("->lx_pcm_hw_free\n"); + snd_printd("->lx_pcm_hw_free\n"); mutex_lock(&chip->setup_mutex); if (chip->hardware_running[is_capture]) { @@ -477,7 +478,7 @@ static void lx_trigger_stop(struct lx6464es *chip, struct lx_stream *lx_stream) int err; snd_printd(LXP "stopping: stopping stream\n"); - err = lx_stream_stop(chip, 0, is_capture); + err = lx_stream_pause(chip, 0, is_capture); if (err < 0) snd_printk(KERN_ERR LXP "couldn't stop stream\n"); else @@ -1005,6 +1006,7 @@ static int __devinit snd_lx6464es_create(struct snd_card *card, chip->card = card; chip->pci = pci; chip->irq = -1; + chip->hardware_running[0] = chip->hardware_running[1] = 0; /* initialize synchronization structs */ spin_lock_init(&chip->lock); diff --git a/sound/pci/lx6464es/lx6464es.h b/sound/pci/lx6464es/lx6464es.h index aea621e..2a97a21 100644 --- a/sound/pci/lx6464es/lx6464es.h +++ b/sound/pci/lx6464es/lx6464es.h @@ -30,6 +30,7 @@ #include <sound/core.h> #include <sound/pcm.h> +#include <sound/pcm_params.h> #include "lx_core.h" diff --git a/sound/pci/lx6464es/lx_core.c b/sound/pci/lx6464es/lx_core.c index 617f98b..3a9e1dd 100644 --- a/sound/pci/lx6464es/lx_core.c +++ b/sound/pci/lx6464es/lx_core.c @@ -353,8 +353,11 @@ polling_successful: lx_dsp_reg_readbuf(chip, eReg_CRM2, rmh->stat, rmh->stat_len); } - } else + lx_message_dump(rmh); + } else { snd_printk(LXP "rmh error: %08x\n", reg); + dump_stack(); + } /* clear Reg_CSM_MR */ lx_dsp_reg_write(chip, eReg_CSM, 0); @@ -362,15 +365,11 @@ polling_successful: switch (reg) { case ED_DSP_TIMED_OUT: snd_printk(KERN_WARNING LXP "lx_message_send: dsp timeout\n"); - return -ETIMEDOUT; case ED_DSP_CRASHED: snd_printk(KERN_WARNING LXP "lx_message_send: dsp crashed\n"); - return -EAGAIN; } - lx_message_dump(rmh); - return reg; } @@ -532,10 +531,22 @@ int lx_pipe_allocate(struct lx6464es *chip, u32 pipe, int is_capture, err = lx_message_send_atomic(chip, &chip->rmh); spin_unlock_irqrestore(&chip->msg_lock, flags); - if (err != 0) - snd_printk(KERN_ERR "lx6464es: could not allocate pipe\n"); + if (err == 0) + return 0; - return err; + switch (err) { + case EB_ALLOCATE_AUDIO_IMPOSSIBLE: + snd_printk(KERN_ERR "lx6464es: could not allocate pipe " + "(EB_ALLOCATE_AUDIO_IMPOSSIBLE)\n"); + break; + + case EB_INVALID_AUDIO: + snd_printk(KERN_ERR "lx6464es: could not allocate pipe " + "(EB_INVALID_AUDIO)\n"); + break; + } + + return -1; } int lx_pipe_release(struct lx6464es *chip, u32 pipe, int is_capture) @@ -781,7 +792,8 @@ int lx_stream_set_state(struct lx6464es *chip, u32 pipe, return err; } -int lx_stream_set_format(struct lx6464es *chip, struct snd_pcm_runtime *runtime, +int lx_stream_set_format(struct lx6464es *chip, + struct snd_pcm_hw_params *params, u32 pipe, int is_capture) { int err; @@ -789,22 +801,19 @@ int lx_stream_set_format(struct lx6464es *chip, struct snd_pcm_runtime *runtime, u32 pipe_cmd = PIPE_INFO_TO_CMD(is_capture, pipe); - u32 channels = runtime->channels; - - if (runtime->channels != channels) - snd_printk(KERN_ERR LXP "channel count mismatch: %d vs %d", - runtime->channels, channels); + const u32 channels = params_channels(params); + const snd_pcm_format_t format = params_format(params); spin_lock_irqsave(&chip->msg_lock, flags); lx_message_init(&chip->rmh, CMD_0C_DEF_STREAM); chip->rmh.cmd[0] |= pipe_cmd; - if (runtime->sample_bits == 16) + if (snd_pcm_format_width(format) == 16) /* 16 bit format */ chip->rmh.cmd[0] |= (STREAM_FMT_16b << STREAM_FMT_OFFSET); - if (snd_pcm_format_little_endian(runtime->format)) + if (snd_pcm_format_little_endian(format)) /* little endian/intel format */ chip->rmh.cmd[0] |= (STREAM_FMT_intel << STREAM_FMT_OFFSET); @@ -974,7 +983,7 @@ int lx_level_unmute(struct lx6464es *chip, int is_capture, int unmute) chip->rmh.cmd[1] = (u32)(mute_mask >> (u64)32); /* hi part */ chip->rmh.cmd[2] = (u32)(mute_mask & (u64)0xFFFFFFFF); /* lo part */ - snd_printk("mute %x %x %x\n", chip->rmh.cmd[0], chip->rmh.cmd[1], + snd_printd("mute %x %x %x\n", chip->rmh.cmd[0], chip->rmh.cmd[1], chip->rmh.cmd[2]); err = lx_message_send_atomic(chip, &chip->rmh); diff --git a/sound/pci/lx6464es/lx_core.h b/sound/pci/lx6464es/lx_core.h index 6bd9cbb..378721a 100644 --- a/sound/pci/lx6464es/lx_core.h +++ b/sound/pci/lx6464es/lx_core.h @@ -134,7 +134,8 @@ int lx_pipe_wait_for_start(struct lx6464es *chip, u32 pipe, int is_capture); int lx_pipe_wait_for_idle(struct lx6464es *chip, u32 pipe, int is_capture); /* low-level stream handling */ -int lx_stream_set_format(struct lx6464es *chip, struct snd_pcm_runtime *runtime, +int lx_stream_set_format(struct lx6464es *chip, + struct snd_pcm_hw_params *runtime, u32 pipe, int is_capture); int lx_stream_state(struct lx6464es *chip, u32 pipe, int is_capture, int *rstate); -- 1.7.4.1

2 2

[alsa-devel] HDAudio period/buffer sizes
by pl bossart 26 Jul '11

26 Jul '11

When I look at the HDAudio spec, I see that the BDL entries need to be aligned on a 128-byte boundary, but the buffer length can be any value. When I look at the HDAudio driver (azx_pcm_open in hda_codec.c), I see that we enforce a 128-byte granularity on the period/buffer sizes, but this was added in a patch that fixed alignment issues (5f1545bc) Any technical reason why the driver is more demanding than the spec? Or is this a confusion between alignment and buffer size? I can understand that rounding to multiple of 128 bytes would be more efficient in terms of data access, but it also prevents applications from getting an interrupt when they want (eg it's impossible to use 20ms periods with a 44.1kHz sampling rate, you'll get 19.59 or 20.31ms). Thanks, -Pierre

2 1

[alsa-devel] [PATCH] ASoC: dapm - change stream event dbg to vdgb
by Liam Girdwood 25 Jul '11

25 Jul '11

Stream event debug can be noisy on larger audio devices so improve the debug SNR by changing it to the verbose level. Signed-off-by: Liam Girdwood <lrg(a)ti.com> --- sound/soc/soc-dapm.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 120df5b..3956670 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -2710,7 +2710,7 @@ static void soc_dapm_stream_event(struct snd_soc_dapm_context *dapm, { if (!w->sname || w->dapm != dapm) continue; - dev_dbg(w->dapm->dev, "widget %s\n %s stream %s event %d\n", + dev_vdbg(w->dapm->dev, "widget %s\n %s stream %s event %d\n", w->name, w->sname, stream, event); if (strstr(w->sname, stream)) { switch(event) { -- 1.7.4.1

2 1