From 06f409d76f1d382167eb1cadde2e23a73272865d Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 7 Apr 2009 18:10:13 +0100 Subject: ASoC: Provide core support for symmetric sample rates Many devices require symmetric configurations of capture and playback data formats, often due to shared clocking but sometimes also due to other shared playback and record configuration in the device. Start providing core support for this by allowing the DAIs or the machine to specify that the sample rates used should be kept symmetric. A flag symmetric_rates is provided in the snd_soc_dai and snd_soc_dai_link structures. If this is set in either of the DAIs or in the machine then a constraint will be applied when a stream is already open preventing any changes in sample rate. Signed-off-by: Mark Brown --- include/sound/soc-dai.h | 1 + include/sound/soc.h | 6 ++++++ 2 files changed, 7 insertions(+) (limited to 'include') diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index 13676472ddf..22b729fbbf8 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -208,6 +208,7 @@ struct snd_soc_dai { /* DAI capabilities */ struct snd_soc_pcm_stream capture; struct snd_soc_pcm_stream playback; + unsigned int symmetric_rates:1; /* DAI runtime info */ struct snd_pcm_runtime *runtime; diff --git a/include/sound/soc.h b/include/sound/soc.h index a40bc6f316f..b1f2f8819fe 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -417,6 +417,12 @@ struct snd_soc_dai_link { /* codec/machine specific init - e.g. add machine controls */ int (*init)(struct snd_soc_codec *codec); + /* Symmetry requirements */ + unsigned int symmetric_rates:1; + + /* Symmetry data - only valid if symmetry is being enforced */ + unsigned int rate; + /* DAI pcm */ struct snd_pcm *pcm; }; -- cgit v1.2.3 From f6d655a6e6974e474a11b25052c29d10b80814b3 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 13 Apr 2009 11:27:03 +0100 Subject: ASoC: Support DAPM events for DACs and ADCs Signed-off-by: Mark Brown --- include/sound/soc-dapm.h | 10 ++++++++++ 1 file changed, 10 insertions(+) (limited to 'include') diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index a7def6a9a03..fcc929da033 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -140,9 +140,19 @@ #define SND_SOC_DAPM_DAC(wname, stname, wreg, wshift, winvert) \ { .id = snd_soc_dapm_dac, .name = wname, .sname = stname, .reg = wreg, \ .shift = wshift, .invert = winvert} +#define SND_SOC_DAPM_DAC_E(wname, stname, wreg, wshift, winvert, \ + wevent, wflags) \ +{ .id = snd_soc_dapm_dac, .name = wname, .sname = stname, .reg = wreg, \ + .shift = wshift, .invert = winvert, \ + .event = wevent, .event_flags = wflags} #define SND_SOC_DAPM_ADC(wname, stname, wreg, wshift, winvert) \ { .id = snd_soc_dapm_adc, .name = wname, .sname = stname, .reg = wreg, \ .shift = wshift, .invert = winvert} +#define SND_SOC_DAPM_ADC_E(wname, stname, wreg, wshift, winvert, \ + wevent, wflags) \ +{ .id = snd_soc_dapm_adc, .name = wname, .sname = stname, .reg = wreg, \ + .shift = wshift, .invert = winvert, \ + .event = wevent, .event_flags = wflags} /* generic register modifier widget */ #define SND_SOC_DAPM_REG(wid, wname, wreg, wshift, wmask, won_val, woff_val) \ -- cgit v1.2.3 From 02bec490450836ebbd628e97ec03f10b57def8ce Mon Sep 17 00:00:00 2001 From: Tim Blechmann Date: Tue, 24 Mar 2009 12:24:35 +0100 Subject: ALSA: lx6464es - driver for the digigram lx6464es interface prototype of a driver for the digigram lx6464es 64 channel ethersound interface. Signed-off-by: Tim Blechmann Signed-off-by: Takashi Iwai --- include/linux/pci_ids.h | 5 +++++ 1 file changed, 5 insertions(+) (limited to 'include') diff --git a/include/linux/pci_ids.h b/include/linux/pci_ids.h index ee98cd57088..2b1a69598e7 100644 --- a/include/linux/pci_ids.h +++ b/include/linux/pci_ids.h @@ -1005,6 +1005,7 @@ #define PCI_DEVICE_ID_PLX_PCI200SYN 0x3196 #define PCI_DEVICE_ID_PLX_9030 0x9030 #define PCI_DEVICE_ID_PLX_9050 0x9050 +#define PCI_DEVICE_ID_PLX_9056 0x9056 #define PCI_DEVICE_ID_PLX_9080 0x9080 #define PCI_DEVICE_ID_PLX_GTEK_SERIAL2 0xa001 @@ -1847,6 +1848,10 @@ #define PCI_SUBDEVICE_ID_HYPERCOPE_METRO 0x0107 #define PCI_SUBDEVICE_ID_HYPERCOPE_CHAMP2 0x0108 +#define PCI_VENDOR_ID_DIGIGRAM 0x1369 +#define PCI_SUBDEVICE_ID_DIGIGRAM_LX6464ES_SERIAL_SUBSYSTEM 0xc001 +#define PCI_SUBDEVICE_ID_DIGIGRAM_LX6464ES_CAE_SERIAL_SUBSYSTEM 0xc002 + #define PCI_VENDOR_ID_KAWASAKI 0x136b #define PCI_DEVICE_ID_MCHIP_KL5A72002 0xff01 -- cgit v1.2.3 From b75576d76d4be50196773f36709cb7a4f5ac2ab7 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 20 Apr 2009 17:56:13 +0100 Subject: ASoC: Make the DAPM power check an operation on the widget Rather than having switch statements at point of use make the DAPM power check a member of the widget structure and set it when we instantiate the widget. Signed-off-by: Mark Brown --- include/sound/soc-dapm.h | 2 ++ 1 file changed, 2 insertions(+) (limited to 'include') diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index fcc929da033..839a97b6326 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -367,6 +367,8 @@ struct snd_soc_dapm_widget { unsigned char suspend:1; /* was active before suspend */ unsigned char pmdown:1; /* waiting for timeout */ + int (*power_check)(struct snd_soc_dapm_widget *w); + /* external events */ unsigned short event_flags; /* flags to specify event types */ int (*event)(struct snd_soc_dapm_widget*, struct snd_kcontrol *, int); -- cgit v1.2.3 From cd474f2d548af3c0eb932d9d47ec11483861aa6f Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 21 Apr 2009 08:53:08 +0200 Subject: ALSA: Remove deprecated snd_card_new() Signed-off-by: Takashi Iwai --- include/sound/core.h | 10 ---------- 1 file changed, 10 deletions(-) (limited to 'include') diff --git a/include/sound/core.h b/include/sound/core.h index 3dea79829ac..a26bbdcc676 100644 --- a/include/sound/core.h +++ b/include/sound/core.h @@ -300,16 +300,6 @@ int snd_card_create(int idx, const char *id, struct module *module, int extra_size, struct snd_card **card_ret); -static inline __deprecated -struct snd_card *snd_card_new(int idx, const char *id, - struct module *module, int extra_size) -{ - struct snd_card *card; - if (snd_card_create(idx, id, module, extra_size, &card) < 0) - return NULL; - return card; -} - int snd_card_disconnect(struct snd_card *card); int snd_card_free(struct snd_card *card); int snd_card_free_when_closed(struct snd_card *card); -- cgit v1.2.3 From ef9dfa4b1052af23a603de382d4665b2d1fccc61 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 21 Apr 2009 08:53:41 +0200 Subject: ALSA: Remove deprecated include/sound/driver.h Signed-off-by: Takashi Iwai --- include/sound/driver.h | 1 - 1 file changed, 1 deletion(-) delete mode 100644 include/sound/driver.h (limited to 'include') diff --git a/include/sound/driver.h b/include/sound/driver.h deleted file mode 100644 index f0359437d01..00000000000 --- a/include/sound/driver.h +++ /dev/null @@ -1 +0,0 @@ -#warning "This file is deprecated" -- cgit v1.2.3 From 246d0a17f5e09af0794960164269fc8988a8811c Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 22 Apr 2009 18:24:55 +0100 Subject: ASoC: Add power supply widget to DAPM Many modern CODECs have shared resources on chip which must be enabled for portions of the chip to work but which can be disabled at other times in order to achieve power savings. Examples of such resources include power supplies and some internal clocks. Since these widgets are dependencies for the audio path but do not carry audio signals they require slightly different handling to most widgets - they do not contribute to the audio path and so should not be counted as either inputs or outputs during path walks. Cases where one supply provides a supply for another will require additional work. There is also room for more optimisation of the graph walking to avoid repeated checks for the same thing. Signed-off-by: Mark Brown --- include/sound/soc-dapm.h | 7 ++++++- 1 file changed, 6 insertions(+), 1 deletion(-) (limited to 'include') diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index 839a97b6326..533f9f25649 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -154,12 +154,16 @@ .shift = wshift, .invert = winvert, \ .event = wevent, .event_flags = wflags} -/* generic register modifier widget */ +/* generic widgets */ #define SND_SOC_DAPM_REG(wid, wname, wreg, wshift, wmask, won_val, woff_val) \ { .id = wid, .name = wname, .kcontrols = NULL, .num_kcontrols = 0, \ .reg = -((wreg) + 1), .shift = wshift, .mask = wmask, \ .on_val = won_val, .off_val = woff_val, .event = dapm_reg_event, \ .event_flags = SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD} +#define SND_SOC_DAPM_SUPPLY(wname, wreg, wshift, winvert, wevent, wflags) \ +{ .id = snd_soc_dapm_supply, .name = wname, .reg = wreg, \ + .shift = wshift, .invert = winvert, .event = wevent, \ + .event_flags = wflags} /* dapm kcontrol types */ #define SOC_DAPM_SINGLE(xname, reg, shift, max, invert) \ @@ -308,6 +312,7 @@ enum snd_soc_dapm_type { snd_soc_dapm_vmid, /* codec bias/vmid - to minimise pops */ snd_soc_dapm_pre, /* machine specific pre widget - exec first */ snd_soc_dapm_post, /* machine specific post widget - exec last */ + snd_soc_dapm_supply, /* power/clock supply */ }; /* -- cgit v1.2.3 From 7629ad24f2b3df95c8b4cd8869e3c04e1df6c442 Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Fri, 24 Apr 2009 16:37:44 +0200 Subject: ASoC: add SOC_DOUBLE_EXT macro Add a macro for double controls with special callback functions. Signed-off-by: Daniel Mack Signed-off-by: Mark Brown --- include/sound/soc.h | 8 ++++++++ 1 file changed, 8 insertions(+) (limited to 'include') diff --git a/include/sound/soc.h b/include/sound/soc.h index b1f2f8819fe..6ab80bf7abd 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -118,6 +118,14 @@ .info = snd_soc_info_volsw, \ .get = xhandler_get, .put = xhandler_put, \ .private_value = SOC_SINGLE_VALUE(xreg, xshift, xmax, xinvert) } +#define SOC_DOUBLE_EXT(xname, xreg, shift_left, shift_right, xmax, xinvert,\ + xhandler_get, xhandler_put) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname),\ + .info = snd_soc_info_volsw, \ + .get = xhandler_get, .put = xhandler_put, \ + .private_value = (unsigned long)&(struct soc_mixer_control) \ + {.reg = xreg, .shift = shift_left, .rshift = shift_right, \ + .max = xmax, .invert = xinvert} } #define SOC_SINGLE_EXT_TLV(xname, xreg, xshift, xmax, xinvert,\ xhandler_get, xhandler_put, tlv_array) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ -- cgit v1.2.3 From 33f503c96c976fd585dedb76514ca6cb286e60d9 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sat, 2 May 2009 12:24:55 +0100 Subject: ASoC: Use a shared define for AC97 CODEC data formats The AC97 wire format is completely fixed so CODECs don't have any choice about the formats they accept but controllers accept a variety of data formats and render them down onto the bus. Have a shared define so all the CODEC drivers will interoperate with any of our controller drivers. Signed-off-by: Mark Brown --- include/sound/soc-dai.h | 3 +++ 1 file changed, 3 insertions(+) (limited to 'include') diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index 22b729fbbf8..ea07b4bd516 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -96,6 +96,9 @@ struct snd_pcm_substream; #define SND_SOC_CLOCK_IN 0 #define SND_SOC_CLOCK_OUT 1 +#define SND_SOC_STD_AC97_FMTS (SNDRV_PCM_FMTBIT_S16_LE |\ + SNDRV_PCM_FMTBIT_S32_LE) + struct snd_soc_dai_ops; struct snd_soc_dai; struct snd_ac97_bus_ops; -- cgit v1.2.3 From 4072604b9dd18f25a98cc0f4d3d4553ed1ad4152 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sat, 2 May 2009 12:28:25 +0100 Subject: ASoC: Remove unused DAI format defines The defines for TDM and synchronous clocks are not used - they are mostly a legacy of the automatic clocking configuration. TDM will require configuration of the number of timeslots and which ones to use so can't be fit into the DAI format and synchronous mode is handled by symmetric_rates (and needs to be done by constraints rather than when the DAI format is being configured). Signed-off-by: Mark Brown --- include/sound/soc-dai.h | 18 ------------------ 1 file changed, 18 deletions(-) (limited to 'include') diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index ea07b4bd516..a997c2cac63 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -44,24 +44,6 @@ struct snd_pcm_substream; #define SND_SOC_DAIFMT_CONT (0 << 4) /* continuous clock */ #define SND_SOC_DAIFMT_GATED (1 << 4) /* clock is gated */ -/* - * DAI Left/Right Clocks. - * - * Specifies whether the DAI can support different samples for similtanious - * playback and capture. This usually requires a seperate physical frame - * clock for playback and capture. - */ -#define SND_SOC_DAIFMT_SYNC (0 << 5) /* Tx FRM = Rx FRM */ -#define SND_SOC_DAIFMT_ASYNC (1 << 5) /* Tx FRM ~ Rx FRM */ - -/* - * TDM - * - * Time Division Multiplexing. Allows PCM data to be multplexed with other - * data on the DAI. - */ -#define SND_SOC_DAIFMT_TDM (1 << 6) - /* * DAI hardware signal inversions. * -- cgit v1.2.3 From bbd993077d788589a86a718ba7a7895ba5e71a17 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 5 May 2009 10:27:38 +0100 Subject: ASoC: Remove redundant codec pointer from DAIs The DAI structure has two pointers to the codec, one in the body of the DAI and one in a union for a parent pointer. Drop the parent pointer version. Signed-off-by: Mark Brown --- include/sound/soc-dai.h | 7 ++----- 1 file changed, 2 insertions(+), 5 deletions(-) (limited to 'include') diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index a997c2cac63..496dc30457b 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -205,11 +205,8 @@ struct snd_soc_dai { /* DAI private data */ void *private_data; - /* parent codec/platform */ - union { - struct snd_soc_codec *codec; - struct snd_soc_platform *platform; - }; + /* parent platform */ + struct snd_soc_platform *platform; struct list_head list; }; -- cgit v1.2.3 From 4bbe1ddf89a5ba3ec30fe5980912d8bda3a3cbb2 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 13 Oct 2008 03:07:14 +0200 Subject: ALSA: Add extra delay count in PCM Added runtime->delay field to adjust the delayed samples for snd_pcm_delay(). Typically a hardware FIFO length is stored in this field, so that the extra delay between hwptr and applptr can be computed. Signed-off-by: Takashi Iwai --- include/sound/pcm.h | 1 + 1 file changed, 1 insertion(+) (limited to 'include') diff --git a/include/sound/pcm.h b/include/sound/pcm.h index c1729689161..267effddb07 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -270,6 +270,7 @@ struct snd_pcm_runtime { snd_pcm_uframes_t hw_ptr_base; /* Position at buffer restart */ snd_pcm_uframes_t hw_ptr_interrupt; /* Position at interrupt time */ unsigned long hw_ptr_jiffies; /* Time when hw_ptr is updated */ + snd_pcm_sframes_t delay; /* extra delay; typically FIFO size */ /* -- HW params -- */ snd_pcm_access_t access; /* access mode */ -- cgit v1.2.3 From d34c43078236b41146877c49af341ab85b5fb8db Mon Sep 17 00:00:00 2001 From: Jon Smirl Date: Wed, 13 May 2009 21:59:14 -0400 Subject: ASoC: Add SNDRV_PCM_FMTBIT_S32_BE as a valid AC97 format Signed-off-by: Jon Smirl Signed-off-by: Mark Brown --- include/sound/soc-dai.h | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) (limited to 'include') diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index 496dc30457b..352d7eee9b6 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -79,7 +79,8 @@ struct snd_pcm_substream; #define SND_SOC_CLOCK_OUT 1 #define SND_SOC_STD_AC97_FMTS (SNDRV_PCM_FMTBIT_S16_LE |\ - SNDRV_PCM_FMTBIT_S32_LE) + SNDRV_PCM_FMTBIT_S32_LE |\ + SNDRV_PCM_FMTBIT_S32_BE) struct snd_soc_dai_ops; struct snd_soc_dai; -- cgit v1.2.3 From 9fc20f030ba457d20eb994e1def7e2ce7d5ae1a8 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 14 May 2009 15:14:18 +0200 Subject: ALSA: ctxfi - Move PCI ID definitions to linux/pci_ids.h Signed-off-by: Takashi Iwai --- include/linux/pci_ids.h | 7 +++++++ 1 file changed, 7 insertions(+) (limited to 'include') diff --git a/include/linux/pci_ids.h b/include/linux/pci_ids.h index 06ba90c211a..61915313898 100644 --- a/include/linux/pci_ids.h +++ b/include/linux/pci_ids.h @@ -1314,6 +1314,13 @@ #define PCI_VENDOR_ID_CREATIVE 0x1102 /* duplicate: ECTIVA */ #define PCI_DEVICE_ID_CREATIVE_EMU10K1 0x0002 +#define PCI_DEVICE_ID_CREATIVE_20K1 0x0005 +#define PCI_DEVICE_ID_CREATIVE_20K2 0x000b +#define PCI_SUBDEVICE_ID_CREATIVE_SB0760 0x0024 +#define PCI_SUBDEVICE_ID_CREATIVE_SB08801 0x0041 +#define PCI_SUBDEVICE_ID_CREATIVE_SB08802 0x0042 +#define PCI_SUBDEVICE_ID_CREATIVE_SB08803 0x0043 +#define PCI_SUBDEVICE_ID_CREATIVE_HENDRIX 0x6000 #define PCI_VENDOR_ID_ECTIVA 0x1102 /* duplicate: CREATIVE */ #define PCI_DEVICE_ID_ECTIVA_EV1938 0x8938 -- cgit v1.2.3 From 6d3ddc81f5762d54ce7d1db70eb757c6c12fabbc Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sat, 16 May 2009 17:47:29 +0100 Subject: ASoC: Split DAPM power checks from sequencing of power changes DAPM has always applied any changes to the power state of widgets as soon as it has determined that they are required. Instead of doing this store all the changes that are required on lists of widgets to power up and down, then iterate over those lists and apply the changes. This changes the sequence in which changes are implemented, doing all power downs before power ups and always using the up/down sequences (previously they were only used when changes were due to DAC/ADC power events). The error handling is also changed so that we continue attempting to power widgets if some changes fail. The main benefit of this is to allow future changes to do optimisations over the whole power sequence and to reduce the number of walks of the widget graph required to check the power status of widgets. Signed-off-by: Mark Brown --- include/sound/soc-dapm.h | 3 +++ include/sound/soc.h | 2 ++ 2 files changed, 5 insertions(+) (limited to 'include') diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index 533f9f25649..b3f789d0cee 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -385,6 +385,9 @@ struct snd_soc_dapm_widget { /* widget input and outputs */ struct list_head sources; struct list_head sinks; + + /* used during DAPM updates */ + struct list_head power_list; }; #endif diff --git a/include/sound/soc.h b/include/sound/soc.h index 6ab80bf7abd..8309ce81cf3 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -372,6 +372,8 @@ struct snd_soc_codec { enum snd_soc_bias_level bias_level; enum snd_soc_bias_level suspend_bias_level; struct delayed_work delayed_work; + struct list_head up_list; + struct list_head down_list; /* codec DAI's */ struct snd_soc_dai *dai; -- cgit v1.2.3 From 452c5eaa0d5162e02ffee742ea17540887bc2904 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sun, 17 May 2009 21:41:23 +0100 Subject: ASoC: Integrate bias management with DAPM power management Rather than managing the bias level of the system based on if there is an active audio stream manage it based on there being an active DAPM widget. This simplifies the code a little, moving the power handling into one place, and improves audio performance for bypass paths when no playbacks or captures are active. Signed-off-by: Mark Brown --- include/sound/soc-dapm.h | 2 -- include/sound/soc.h | 1 + 2 files changed, 1 insertion(+), 2 deletions(-) (limited to 'include') diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index b3f789d0cee..ec8a45f9a06 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -279,8 +279,6 @@ int snd_soc_dapm_add_routes(struct snd_soc_codec *codec, /* dapm events */ int snd_soc_dapm_stream_event(struct snd_soc_codec *codec, char *stream, int event); -int snd_soc_dapm_set_bias_level(struct snd_soc_device *socdev, - enum snd_soc_bias_level level); /* dapm sys fs - used by the core */ int snd_soc_dapm_sys_add(struct device *dev); diff --git a/include/sound/soc.h b/include/sound/soc.h index 8309ce81cf3..2af3213df90 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -339,6 +339,7 @@ struct snd_soc_codec { struct module *owner; struct mutex mutex; struct device *dev; + struct snd_soc_device *socdev; struct list_head list; -- cgit v1.2.3 From 5c82f56736e4c3a9eaf53c94366b056c8622d79e Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 22 May 2009 09:41:30 +0100 Subject: AsoC: Make snd_soc_read() and snd_soc_write() functions Should be no impact on the generated code but it helps the compiler print clearer messages. Signed-off-by: Mark Brown --- include/sound/soc.h | 17 +++++++++++++---- 1 file changed, 13 insertions(+), 4 deletions(-) (limited to 'include') diff --git a/include/sound/soc.h b/include/sound/soc.h index 2af3213df90..cf6111d72b1 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -214,10 +214,6 @@ void snd_soc_jack_free_gpios(struct snd_soc_jack *jack, int count, struct snd_soc_jack_gpio *gpios); #endif -/* codec IO */ -#define snd_soc_read(codec, reg) codec->read(codec, reg) -#define snd_soc_write(codec, reg, value) codec->write(codec, reg, value) - /* codec register bit access */ int snd_soc_update_bits(struct snd_soc_codec *codec, unsigned short reg, unsigned short mask, unsigned short value); @@ -507,6 +503,19 @@ struct soc_enum { void *dapm; }; +/* codec IO */ +static inline unsigned int snd_soc_read(struct snd_soc_codec *codec, + unsigned int reg) +{ + return codec->read(codec, reg); +} + +static inline unsigned int snd_soc_write(struct snd_soc_codec *codec, + unsigned int reg, unsigned int val) +{ + return codec->write(codec, reg, val); +} + #include #endif -- cgit v1.2.3 From 86ed3669f068b514ab85ffd548456a342b3fb8d3 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 22 May 2009 15:01:19 +0100 Subject: ASoC: WM9081 mono DAC with integrated 2.6W class AB/D amplifier driver The WM9081 is designed to provide high power output at low distortion levels in space-constrained portable applications. Signed-off-by: Mark Brown --- include/sound/wm9081.h | 25 +++++++++++++++++++++++++ 1 file changed, 25 insertions(+) create mode 100644 include/sound/wm9081.h (limited to 'include') diff --git a/include/sound/wm9081.h b/include/sound/wm9081.h new file mode 100644 index 00000000000..e173ddbf6bd --- /dev/null +++ b/include/sound/wm9081.h @@ -0,0 +1,25 @@ +/* + * linux/sound/wm9081.h -- Platform data for WM9081 + * + * Copyright 2009 Wolfson Microelectronics. PLC. + * + * 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. + */ + +#ifndef __LINUX_SND_WM_9081_H +#define __LINUX_SND_WM_9081_H + +struct wm9081_retune_mobile_setting { + const char *name; + unsigned int rate; + u16 config[20]; +}; + +struct wm9081_retune_mobile_config { + struct wm9081_retune_mobile_setting *configs; + int num_configs; +}; + +#endif -- cgit v1.2.3 From 8bea869c5e56234990e6bad92a543437115bfc18 Mon Sep 17 00:00:00 2001 From: Jaroslav Kysela Date: Mon, 27 Apr 2009 09:44:40 +0200 Subject: ALSA: PCM midlevel: improve fifo_size handling Move the fifo_size assignment to hw->ioctl callback to allow lowlevel drivers overwrite the default behaviour. fifo_size is in frames not bytes as specified in asound.h and alsa-lib's documentation, but most hardware have fixed byte based FIFOs. Introduce internal SNDRV_PCM_INFO_FIFO_IN_FRAMES. Signed-off-by: Jaroslav Kysela Signed-off-by: Takashi Iwai --- include/sound/asound.h | 1 + include/sound/pcm.h | 1 + 2 files changed, 2 insertions(+) (limited to 'include') diff --git a/include/sound/asound.h b/include/sound/asound.h index 6add80fc251..82aed3f4753 100644 --- a/include/sound/asound.h +++ b/include/sound/asound.h @@ -255,6 +255,7 @@ typedef int __bitwise snd_pcm_subformat_t; #define SNDRV_PCM_INFO_HALF_DUPLEX 0x00100000 /* only half duplex */ #define SNDRV_PCM_INFO_JOINT_DUPLEX 0x00200000 /* playback and capture stream are somewhat correlated */ #define SNDRV_PCM_INFO_SYNC_START 0x00400000 /* pcm support some kind of sync go */ +#define SNDRV_PCM_INFO_FIFO_IN_FRAMES 0x80000000 /* internal kernel flag - FIFO size is in frames */ typedef int __bitwise snd_pcm_state_t; #define SNDRV_PCM_STATE_OPEN ((__force snd_pcm_state_t) 0) /* stream is open */ diff --git a/include/sound/pcm.h b/include/sound/pcm.h index 267effddb07..8a69b5c1e1c 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -98,6 +98,7 @@ struct snd_pcm_ops { #define SNDRV_PCM_IOCTL1_INFO 1 #define SNDRV_PCM_IOCTL1_CHANNEL_INFO 2 #define SNDRV_PCM_IOCTL1_GSTATE 3 +#define SNDRV_PCM_IOCTL1_FIFO_SIZE 4 #define SNDRV_PCM_TRIGGER_STOP 0 #define SNDRV_PCM_TRIGGER_START 1 -- cgit v1.2.3 From 10a8ebbb08c4b08292598947bbe534e04d6ee705 Mon Sep 17 00:00:00 2001 From: Jaroslav Kysela Date: Tue, 2 Jun 2009 12:02:38 +0200 Subject: ALSA: Core - add snd_card_set_id() function Introduce snd_card_set_id() function to allow lowlevel drivers to set default identification name for card slot. The function checks also for identification name collisions and tries to create unique name. Also, the snd_card_create() function is simplified, because this new function is used. As bonus, proper name collision checks are evaluated at the card create time. Signed-off-by: Jaroslav Kysela Signed-off-by: Takashi Iwai --- include/sound/core.h | 1 + 1 file changed, 1 insertion(+) (limited to 'include') diff --git a/include/sound/core.h b/include/sound/core.h index 3dea79829ac..0e8ae10155a 100644 --- a/include/sound/core.h +++ b/include/sound/core.h @@ -313,6 +313,7 @@ struct snd_card *snd_card_new(int idx, const char *id, int snd_card_disconnect(struct snd_card *card); int snd_card_free(struct snd_card *card); int snd_card_free_when_closed(struct snd_card *card); +void snd_card_set_id(struct snd_card *card, const char *id); int snd_card_register(struct snd_card *card); int snd_card_info_init(void); int snd_card_info_done(void); -- cgit v1.2.3 From 3f7440a6b771169e1f11fa582e53a4259b682809 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 5 Jun 2009 17:40:04 +0200 Subject: ALSA: Clean up 64bit division functions Replace the house-made div64_32() with the standard div_u64*() functions. Signed-off-by: Takashi Iwai --- include/sound/pcm.h | 74 ----------------------------------------------------- 1 file changed, 74 deletions(-) (limited to 'include') diff --git a/include/sound/pcm.h b/include/sound/pcm.h index c1729689161..0caf71e1694 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -486,80 +486,6 @@ void snd_pcm_detach_substream(struct snd_pcm_substream *substream); void snd_pcm_vma_notify_data(void *client, void *data); int snd_pcm_mmap_data(struct snd_pcm_substream *substream, struct file *file, struct vm_area_struct *area); -#if BITS_PER_LONG >= 64 - -static inline void div64_32(u_int64_t *n, u_int32_t div, u_int32_t *rem) -{ - *rem = *n % div; - *n /= div; -} - -#elif defined(i386) - -static inline void div64_32(u_int64_t *n, u_int32_t div, u_int32_t *rem) -{ - u_int32_t low, high; - low = *n & 0xffffffff; - high = *n >> 32; - if (high) { - u_int32_t high1 = high % div; - high /= div; - asm("divl %2":"=a" (low), "=d" (*rem):"rm" (div), "a" (low), "d" (high1)); - *n = (u_int64_t)high << 32 | low; - } else { - *n = low / div; - *rem = low % div; - } -} -#else - -static inline void divl(u_int32_t high, u_int32_t low, - u_int32_t div, - u_int32_t *q, u_int32_t *r) -{ - u_int64_t n = (u_int64_t)high << 32 | low; - u_int64_t d = (u_int64_t)div << 31; - u_int32_t q1 = 0; - int c = 32; - while (n > 0xffffffffU) { - q1 <<= 1; - if (n >= d) { - n -= d; - q1 |= 1; - } - d >>= 1; - c--; - } - q1 <<= c; - if (n) { - low = n; - *q = q1 | (low / div); - *r = low % div; - } else { - *r = 0; - *q = q1; - } - return; -} - -static inline void div64_32(u_int64_t *n, u_int32_t div, u_int32_t *rem) -{ - u_int32_t low, high; - low = *n & 0xffffffff; - high = *n >> 32; - if (high) { - u_int32_t high1 = high % div; - u_int32_t low1 = low; - high /= div; - divl(high1, low1, div, &low, rem); - *n = (u_int64_t)high << 32 | low; - } else { - *n = low / div; - *rem = low % div; - } -} -#endif - /* * PCM library */ -- cgit v1.2.3