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author | Takashi Iwai <tiwai@suse.de> | 2015-03-23 13:14:02 +0100 |
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committer | Takashi Iwai <tiwai@suse.de> | 2015-03-23 13:14:02 +0100 |
commit | 3372dbdd8ca11f51be8c6a30b2bc79eb04c4a902 (patch) | |
tree | d4499bf5a5665b4820ffaf96bce55bf6b895195e | |
parent | bc465aa9d045feb0e13b4a8f32cc33c1943f62d6 (diff) | |
parent | 967b1307b69b8ada8b331e01046ad1ef83742e99 (diff) | |
download | op-kernel-dev-3372dbdd8ca11f51be8c6a30b2bc79eb04c4a902.zip op-kernel-dev-3372dbdd8ca11f51be8c6a30b2bc79eb04c4a902.tar.gz |
Merge branch 'for-next' into topic/hda-core
166 files changed, 4283 insertions, 5504 deletions
diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-max98090.txt b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-max98090.txt index c949abc..c3495be 100644 --- a/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-max98090.txt +++ b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-max98090.txt @@ -18,6 +18,7 @@ Required properties: * Headphones * Speakers * Mic Jack + * Int Mic - nvidia,i2s-controller : The phandle of the Tegra I2S controller that's connected to the CODEC. diff --git a/Documentation/sound/alsa/HD-Audio.txt b/Documentation/sound/alsa/HD-Audio.txt index 42a0a39..e7193aa 100644 --- a/Documentation/sound/alsa/HD-Audio.txt +++ b/Documentation/sound/alsa/HD-Audio.txt @@ -466,7 +466,11 @@ The generic parser supports the following hints: - add_jack_modes (bool): add "xxx Jack Mode" enum controls to each I/O jack for allowing to change the headphone amp and mic bias VREF capabilities -- power_down_unused (bool): power down the unused widgets +- power_save_node (bool): advanced power management for each widget, + controlling the power sate (D0/D3) of each widget node depending on + the actual pin and stream states +- power_down_unused (bool): power down the unused widgets, a subset of + power_save_node, and will be dropped in future - add_hp_mic (bool): add the headphone to capture source if possible - hp_mic_detect (bool): enable/disable the hp/mic shared input for a single built-in mic case; default true diff --git a/Documentation/sound/alsa/timestamping.txt b/Documentation/sound/alsa/timestamping.txt new file mode 100644 index 0000000..0b191a2 --- /dev/null +++ b/Documentation/sound/alsa/timestamping.txt @@ -0,0 +1,200 @@ +The ALSA API can provide two different system timestamps: + +- Trigger_tstamp is the system time snapshot taken when the .trigger +callback is invoked. This snapshot is taken by the ALSA core in the +general case, but specific hardware may have synchronization +capabilities or conversely may only be able to provide a correct +estimate with a delay. In the latter two cases, the low-level driver +is responsible for updating the trigger_tstamp at the most appropriate +and precise moment. Applications should not rely solely on the first +trigger_tstamp but update their internal calculations if the driver +provides a refined estimate with a delay. + +- tstamp is the current system timestamp updated during the last +event or application query. +The difference (tstamp - trigger_tstamp) defines the elapsed time. + +The ALSA API provides reports two basic pieces of information, avail +and delay, which combined with the trigger and current system +timestamps allow for applications to keep track of the 'fullness' of +the ring buffer and the amount of queued samples. + +The use of these different pointers and time information depends on +the application needs: + +- 'avail' reports how much can be written in the ring buffer +- 'delay' reports the time it will take to hear a new sample after all +queued samples have been played out. + +When timestamps are enabled, the avail/delay information is reported +along with a snapshot of system time. Applications can select from +CLOCK_REALTIME (NTP corrections including going backwards), +CLOCK_MONOTONIC (NTP corrections but never going backwards), +CLOCK_MONOTIC_RAW (without NTP corrections) and change the mode +dynamically with sw_params + + +The ALSA API also provide an audio_tstamp which reflects the passage +of time as measured by different components of audio hardware. In +ascii-art, this could be represented as follows (for the playback +case): + + +--------------------------------------------------------------> time + ^ ^ ^ ^ ^ + | | | | | + analog link dma app FullBuffer + time time time time time + | | | | | + |< codec delay >|<--hw delay-->|<queued samples>|<---avail->| + |<----------------- delay---------------------->| | + |<----ring buffer length---->| + +The analog time is taken at the last stage of the playback, as close +as possible to the actual transducer + +The link time is taken at the output of the SOC/chipset as the samples +are pushed on a link. The link time can be directly measured if +supported in hardware by sample counters or wallclocks (e.g. with +HDAudio 24MHz or PTP clock for networked solutions) or indirectly +estimated (e.g. with the frame counter in USB). + +The DMA time is measured using counters - typically the least reliable +of all measurements due to the bursty natured of DMA transfers. + +The app time corresponds to the time tracked by an application after +writing in the ring buffer. + +The application can query what the hardware supports, define which +audio time it wants reported by selecting the relevant settings in +audio_tstamp_config fields, get an estimate of the timestamp +accuracy. It can also request the delay-to-analog be included in the +measurement. Direct access to the link time is very interesting on +platforms that provide an embedded DSP; measuring directly the link +time with dedicated hardware, possibly synchronized with system time, +removes the need to keep track of internal DSP processing times and +latency. + +In case the application requests an audio tstamp that is not supported +in hardware/low-level driver, the type is overridden as DEFAULT and the +timestamp will report the DMA time based on the hw_pointer value. + +For backwards compatibility with previous implementations that did not +provide timestamp selection, with a zero-valued COMPAT timestamp type +the results will default to the HDAudio wall clock for playback +streams and to the DMA time (hw_ptr) in all other cases. + +The audio timestamp accuracy can be returned to user-space, so that +appropriate decisions are made: + +- for dma time (default), the granularity of the transfers can be + inferred from the steps between updates and in turn provide + information on how much the application pointer can be rewound + safely. + +- the link time can be used to track long-term drifts between audio + and system time using the (tstamp-trigger_tstamp)/audio_tstamp + ratio, the precision helps define how much smoothing/low-pass + filtering is required. The link time can be either reset on startup + or reported as is (the latter being useful to compare progress of + different streams - but may require the wallclock to be always + running and not wrap-around during idle periods). If supported in + hardware, the absolute link time could also be used to define a + precise start time (patches WIP) + +- including the delay in the audio timestamp may + counter-intuitively not increase the precision of timestamps, e.g. if a + codec includes variable-latency DSP processing or a chain of + hardware components the delay is typically not known with precision. + +The accuracy is reported in nanosecond units (using an unsigned 32-bit +word), which gives a max precision of 4.29s, more than enough for +audio applications... + +Due to the varied nature of timestamping needs, even for a single +application, the audio_tstamp_config can be changed dynamically. In +the STATUS ioctl, the parameters are read-only and do not allow for +any application selection. To work around this limitation without +impacting legacy applications, a new STATUS_EXT ioctl is introduced +with read/write parameters. ALSA-lib will be modified to make use of +STATUS_EXT and effectively deprecate STATUS. + +The ALSA API only allows for a single audio timestamp to be reported +at a time. This is a conscious design decision, reading the audio +timestamps from hardware registers or from IPC takes time, the more +timestamps are read the more imprecise the combined measurements +are. To avoid any interpretation issues, a single (system, audio) +timestamp is reported. Applications that need different timestamps +will be required to issue multiple queries and perform an +interpolation of the results + +In some hardware-specific configuration, the system timestamp is +latched by a low-level audio subsytem, and the information provided +back to the driver. Due to potential delays in the communication with +the hardware, there is a risk of misalignment with the avail and delay +information. To make sure applications are not confused, a +driver_timestamp field is added in the snd_pcm_status structure; this +timestamp shows when the information is put together by the driver +before returning from the STATUS and STATUS_EXT ioctl. in most cases +this driver_timestamp will be identical to the regular system tstamp. + +Examples of typestamping with HDaudio: + +1. DMA timestamp, no compensation for DMA+analog delay +$ ./audio_time -p --ts_type=1 +playback: systime: 341121338 nsec, audio time 342000000 nsec, systime delta -878662 +playback: systime: 426236663 nsec, audio time 427187500 nsec, systime delta -950837 +playback: systime: 597080580 nsec, audio time 598000000 nsec, systime delta -919420 +playback: systime: 682059782 nsec, audio time 683020833 nsec, systime delta -961051 +playback: systime: 852896415 nsec, audio time 853854166 nsec, systime delta -957751 +playback: systime: 937903344 nsec, audio time 938854166 nsec, systime delta -950822 + +2. DMA timestamp, compensation for DMA+analog delay +$ ./audio_time -p --ts_type=1 -d +playback: systime: 341053347 nsec, audio time 341062500 nsec, systime delta -9153 +playback: systime: 426072447 nsec, audio time 426062500 nsec, systime delta 9947 +playback: systime: 596899518 nsec, audio time 596895833 nsec, systime delta 3685 +playback: systime: 681915317 nsec, audio time 681916666 nsec, systime delta -1349 +playback: systime: 852741306 nsec, audio time 852750000 nsec, systime delta -8694 + +3. link timestamp, compensation for DMA+analog delay +$ ./audio_time -p --ts_type=2 -d +playback: systime: 341060004 nsec, audio time 341062791 nsec, systime delta -2787 +playback: systime: 426242074 nsec, audio time 426244875 nsec, systime delta -2801 +playback: systime: 597080992 nsec, audio time 597084583 nsec, systime delta -3591 +playback: systime: 682084512 nsec, audio time 682088291 nsec, systime delta -3779 +playback: systime: 852936229 nsec, audio time 852940916 nsec, systime delta -4687 +playback: systime: 938107562 nsec, audio time 938112708 nsec, systime delta -5146 + +Example 1 shows that the timestamp at the DMA level is close to 1ms +ahead of the actual playback time (as a side time this sort of +measurement can help define rewind safeguards). Compensating for the +DMA-link delay in example 2 helps remove the hardware buffering abut +the information is still very jittery, with up to one sample of +error. In example 3 where the timestamps are measured with the link +wallclock, the timestamps show a monotonic behavior and a lower +dispersion. + +Example 3 and 4 are with USB audio class. Example 3 shows a high +offset between audio time and system time due to buffering. Example 4 +shows how compensating for the delay exposes a 1ms accuracy (due to +the use of the frame counter by the driver) + +Example 3: DMA timestamp, no compensation for delay, delta of ~5ms +$ ./audio_time -p -Dhw:1 -t1 +playback: systime: 120174019 nsec, audio time 125000000 nsec, systime delta -4825981 +playback: systime: 245041136 nsec, audio time 250000000 nsec, systime delta -4958864 +playback: systime: 370106088 nsec, audio time 375000000 nsec, systime delta -4893912 +playback: systime: 495040065 nsec, audio time 500000000 nsec, systime delta -4959935 +playback: systime: 620038179 nsec, audio time 625000000 nsec, systime delta -4961821 +playback: systime: 745087741 nsec, audio time 750000000 nsec, systime delta -4912259 +playback: systime: 870037336 nsec, audio time 875000000 nsec, systime delta -4962664 + +Example 4: DMA timestamp, compensation for delay, delay of ~1ms +$ ./audio_time -p -Dhw:1 -t1 -d +playback: systime: 120190520 nsec, audio time 120000000 nsec, systime delta 190520 +playback: systime: 245036740 nsec, audio time 244000000 nsec, systime delta 1036740 +playback: systime: 370034081 nsec, audio time 369000000 nsec, systime delta 1034081 +playback: systime: 495159907 nsec, audio time 494000000 nsec, systime delta 1159907 +playback: systime: 620098824 nsec, audio time 619000000 nsec, systime delta 1098824 +playback: systime: 745031847 nsec, audio time 744000000 nsec, systime delta 1031847 diff --git a/include/sound/compress_driver.h b/include/sound/compress_driver.h index f48089d..fa1d055 100644 --- a/include/sound/compress_driver.h +++ b/include/sound/compress_driver.h @@ -70,7 +70,7 @@ struct snd_compr_runtime { * @device: device pointer * @direction: stream direction, playback/recording * @metadata_set: metadata set flag, true when set - * @next_track: has userspace signall next track transistion, true when set + * @next_track: has userspace signal next track transition, true when set * @private_data: pointer to DSP private data */ struct snd_compr_stream { @@ -95,7 +95,7 @@ struct snd_compr_stream { * and the stream properties * @get_params: retrieve the codec parameters, mandatory * @set_metadata: Set the metadata values for a stream - * @get_metadata: retreives the requested metadata values from stream + * @get_metadata: retrieves the requested metadata values from stream * @trigger: Trigger operations like start, pause, resume, drain, stop. * This callback is mandatory * @pointer: Retrieve current h/w pointer information. Mandatory diff --git a/include/sound/control.h b/include/sound/control.h index 75f3054..95aad6d 100644 --- a/include/sound/control.h +++ b/include/sound/control.h @@ -227,7 +227,7 @@ snd_ctl_add_slave(struct snd_kcontrol *master, struct snd_kcontrol *slave) * Add a virtual slave control to the given master. * Unlike snd_ctl_add_slave(), the element added via this function * is supposed to have volatile values, and get callback is called - * at each time quried from the master. + * at each time queried from the master. * * When the control peeks the hardware values directly and the value * can be changed by other means than the put callback of the element, diff --git a/include/sound/core.h b/include/sound/core.h index da57482..b12931f 100644 --- a/include/sound/core.h +++ b/include/sound/core.h @@ -278,7 +278,8 @@ int snd_device_new(struct snd_card *card, enum snd_device_type type, void *device_data, struct snd_device_ops *ops); int snd_device_register(struct snd_card *card, void *device_data); int snd_device_register_all(struct snd_card *card); -int snd_device_disconnect_all(struct snd_card *card); +void snd_device_disconnect(struct snd_card *card, void *device_data); +void snd_device_disconnect_all(struct snd_card *card); void snd_device_free(struct snd_card *card, void *device_data); void snd_device_free_all(struct snd_card *card); diff --git a/include/sound/pcm.h b/include/sound/pcm.h index c0ddb7e..0cb7f3f 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -60,6 +60,9 @@ struct snd_pcm_hardware { struct snd_pcm_substream; +struct snd_pcm_audio_tstamp_config; /* definitions further down */ +struct snd_pcm_audio_tstamp_report; + struct snd_pcm_ops { int (*open)(struct snd_pcm_substream *substream); int (*close)(struct snd_pcm_substream *substream); @@ -71,8 +74,10 @@ struct snd_pcm_ops { int (*prepare)(struct snd_pcm_substream *substream); int (*trigger)(struct snd_pcm_substream *substream, int cmd); snd_pcm_uframes_t (*pointer)(struct snd_pcm_substream *substream); - int (*wall_clock)(struct snd_pcm_substream *substream, - struct timespec *audio_ts); + int (*get_time_info)(struct snd_pcm_substream *substream, + struct timespec *system_ts, struct timespec *audio_ts, + struct snd_pcm_audio_tstamp_config *audio_tstamp_config, + struct snd_pcm_audio_tstamp_report *audio_tstamp_report); int (*copy)(struct snd_pcm_substream *substream, int channel, snd_pcm_uframes_t pos, void __user *buf, snd_pcm_uframes_t count); @@ -281,6 +286,58 @@ struct snd_pcm_hw_constraint_ranges { struct snd_pcm_hwptr_log; +/* + * userspace-provided audio timestamp config to kernel, + * structure is for internal use only and filled with dedicated unpack routine + */ +struct snd_pcm_audio_tstamp_config { + /* 5 of max 16 bits used */ + u32 type_requested:4; + u32 report_delay:1; /* add total delay to A/D or D/A */ +}; + +static inline void snd_pcm_unpack_audio_tstamp_config(__u32 data, + struct snd_pcm_audio_tstamp_config *config) +{ + config->type_requested = data & 0xF; + config->report_delay = (data >> 4) & 1; +} + +/* + * kernel-provided audio timestamp report to user-space + * structure is for internal use only and read by dedicated pack routine + */ +struct snd_pcm_audio_tstamp_report { + /* 6 of max 16 bits used for bit-fields */ + + /* for backwards compatibility */ + u32 valid:1; + + /* actual type if hardware could not support requested timestamp */ + u32 actual_type:4; + + /* accuracy represented in ns units */ + u32 accuracy_report:1; /* 0 if accuracy unknown, 1 if accuracy field is valid */ + u32 accuracy; /* up to 4.29s, will be packed in separate field */ +}; + +static inline void snd_pcm_pack_audio_tstamp_report(__u32 *data, __u32 *accuracy, + const struct snd_pcm_audio_tstamp_report *report) +{ + u32 tmp; + + tmp = report->accuracy_report; + tmp <<= 4; + tmp |= report->actual_type; + tmp <<= 1; + tmp |= report->valid; + + *data &= 0xffff; /* zero-clear MSBs */ + *data |= (tmp << 16); + *accuracy = report->accuracy; +} + + struct snd_pcm_runtime { /* -- Status -- */ struct snd_pcm_substream *trigger_master; @@ -361,6 +418,11 @@ struct snd_pcm_runtime { struct snd_dma_buffer *dma_buffer_p; /* allocated buffer */ + /* -- audio timestamp config -- */ + struct snd_pcm_audio_tstamp_config audio_tstamp_config; + struct snd_pcm_audio_tstamp_report audio_tstamp_report; + struct timespec driver_tstamp; + #if defined(CONFIG_SND_PCM_OSS) || defined(CONFIG_SND_PCM_OSS_MODULE) /* -- OSS things -- */ struct snd_pcm_oss_runtime oss; diff --git a/include/sound/pcm_params.h b/include/sound/pcm_params.h index 3c45f39..c704357 100644 --- a/include/sound/pcm_params.h +++ b/include/sound/pcm_params.h @@ -366,4 +366,11 @@ static inline int params_physical_width(const struct snd_pcm_hw_params *p) return snd_pcm_format_physical_width(params_format(p)); } +static inline void +params_set_format(struct snd_pcm_hw_params *p, snd_pcm_format_t fmt) +{ + snd_mask_set(hw_param_mask(p, SNDRV_PCM_HW_PARAM_FORMAT), + (__force int)fmt); +} + #endif /* __SOUND_PCM_PARAMS_H */ diff --git a/include/sound/seq_device.h b/include/sound/seq_device.h index 2b5f24c..ddc0d50 100644 --- a/include/sound/seq_device.h +++ b/include/sound/seq_device.h @@ -25,29 +25,26 @@ * registered device information */ -#define ID_LEN 32 - -/* status flag */ -#define SNDRV_SEQ_DEVICE_FREE 0 -#define SNDRV_SEQ_DEVICE_REGISTERED 1 - struct snd_seq_device { /* device info */ struct snd_card *card; /* sound card */ int device; /* device number */ - char id[ID_LEN]; /* driver id */ + const char *id; /* driver id */ char name[80]; /* device name */ int argsize; /* size of the argument */ void *driver_data; /* private data for driver */ - int status; /* flag - read only */ void *private_data; /* private data for the caller */ void (*private_free)(struct snd_seq_device *device); - struct list_head list; /* link to next device */ + struct device dev; }; +#define to_seq_dev(_dev) \ + container_of(_dev, struct snd_seq_device, dev) + +/* sequencer driver */ /* driver operators - * init_device: + * probe: * Initialize the device with given parameters. * Typically, * 1. call snd_hwdep_new @@ -55,25 +52,40 @@ struct snd_seq_device { * 3. call snd_hwdep_register * 4. store the instance to dev->driver_data pointer. * - * free_device: + * remove: * Release the private data. * Typically, call snd_device_free(dev->card, dev->driver_data) */ -struct snd_seq_dev_ops { - int (*init_device)(struct snd_seq_device *dev); - int (*free_device)(struct snd_seq_device *dev); +struct snd_seq_driver { + struct device_driver driver; + char *id; + int argsize; }; +#define to_seq_drv(_drv) \ + container_of(_drv, struct snd_seq_driver, driver) + /* * prototypes */ +#ifdef CONFIG_MODULES void snd_seq_device_load_drivers(void); -int snd_seq_device_new(struct snd_card *card, int device, char *id, int argsize, struct snd_seq_device **result); -int snd_seq_device_register_driver(char *id, struct snd_seq_dev_ops *entry, int argsize); -int snd_seq_device_unregister_driver(char *id); +#else +#define snd_seq_device_load_drivers() +#endif +int snd_seq_device_new(struct snd_card *card, int device, const char *id, + int argsize, struct snd_seq_device **result); #define SNDRV_SEQ_DEVICE_ARGPTR(dev) (void *)((char *)(dev) + sizeof(struct snd_seq_device)) +int __must_check __snd_seq_driver_register(struct snd_seq_driver *drv, + struct module *mod); +#define snd_seq_driver_register(drv) \ + __snd_seq_driver_register(drv, THIS_MODULE) +void snd_seq_driver_unregister(struct snd_seq_driver *drv); + +#define module_snd_seq_driver(drv) \ + module_driver(drv, snd_seq_driver_register, snd_seq_driver_unregister) /* * id strings for generic devices diff --git a/include/sound/seq_kernel.h b/include/sound/seq_kernel.h index 18a2ac5..feb58d4 100644 --- a/include/sound/seq_kernel.h +++ b/include/sound/seq_kernel.h @@ -99,13 +99,9 @@ int snd_seq_event_port_attach(int client, struct snd_seq_port_callback *pcbp, int snd_seq_event_port_detach(int client, int port); #ifdef CONFIG_MODULES -void snd_seq_autoload_lock(void); -void snd_seq_autoload_unlock(void); void snd_seq_autoload_init(void); -#define snd_seq_autoload_exit() snd_seq_autoload_lock() +void snd_seq_autoload_exit(void); #else -#define snd_seq_autoload_lock() -#define snd_seq_autoload_unlock() #define snd_seq_autoload_init() #define snd_seq_autoload_exit() #endif diff --git a/include/sound/soc.h b/include/sound/soc.h index 0d1ade1..b371aef 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -450,8 +450,10 @@ int soc_dai_hw_params(struct snd_pcm_substream *substream, struct snd_soc_dai *dai); /* Jack reporting */ -int snd_soc_jack_new(struct snd_soc_codec *codec, const char *id, int type, - struct snd_soc_jack *jack); +int snd_soc_card_jack_new(struct snd_soc_card *card, const char *id, int type, + struct snd_soc_jack *jack, struct snd_soc_jack_pin *pins, + unsigned int num_pins); + void snd_soc_jack_report(struct snd_soc_jack *jack, int status, int mask); int snd_soc_jack_add_pins(struct snd_soc_jack *jack, int count, struct snd_soc_jack_pin *pins); @@ -659,7 +661,7 @@ struct snd_soc_jack_gpio { struct snd_soc_jack { struct mutex mutex; struct snd_jack *jack; - struct snd_soc_codec *codec; + struct snd_soc_card *card; struct list_head pins; int status; struct blocking_notifier_head notifier; @@ -954,6 +956,9 @@ struct snd_soc_dai_link { unsigned int symmetric_channels:1; unsigned int symmetric_samplebits:1; + /* Mark this pcm with non atomic ops */ + bool nonatomic; + /* Do not create a PCM for this DAI link (Backend link) */ unsigned int no_pcm:1; @@ -1071,11 +1076,16 @@ struct snd_soc_card { /* * Card-specific routes and widgets. + * Note: of_dapm_xxx for Device Tree; Otherwise for driver build-in. */ const struct snd_soc_dapm_widget *dapm_widgets; int num_dapm_widgets; const struct snd_soc_dapm_route *dapm_routes; int num_dapm_routes; + const struct snd_soc_dapm_widget *of_dapm_widgets; + int num_of_dapm_widgets; + const struct snd_soc_dapm_route *of_dapm_routes; + int num_of_dapm_routes; bool fully_routed; struct work_struct deferred_resume_work; @@ -1469,7 +1479,7 @@ static inline struct snd_soc_codec *snd_soc_kcontrol_codec( } /** - * snd_soc_kcontrol_platform() - Returns the platform that registerd the control + * snd_soc_kcontrol_platform() - Returns the platform that registered the control * @kcontrol: The control for which to get the platform * * Note: This function will only work correctly if the control has been diff --git a/include/uapi/sound/asequencer.h b/include/uapi/sound/asequencer.h index 09c8a00..5a5fa49 100644 --- a/include/uapi/sound/asequencer.h +++ b/include/uapi/sound/asequencer.h @@ -22,6 +22,7 @@ #ifndef _UAPI__SOUND_ASEQUENCER_H #define _UAPI__SOUND_ASEQUENCER_H +#include <sound/asound.h> /** version of the sequencer */ #define SNDRV_SEQ_VERSION SNDRV_PROTOCOL_VERSION (1, 0, 1) diff --git a/include/uapi/sound/asound.h b/include/uapi/sound/asound.h index 0e88e7a..46145a5 100644 --- a/include/uapi/sound/asound.h +++ b/include/uapi/sound/asound.h @@ -25,6 +25,9 @@ #include <linux/types.h> +#ifndef __KERNEL__ +#include <stdlib.h> +#endif /* * protocol version @@ -140,7 +143,7 @@ struct snd_hwdep_dsp_image { * * *****************************************************************************/ -#define SNDRV_PCM_VERSION SNDRV_PROTOCOL_VERSION(2, 0, 12) +#define SNDRV_PCM_VERSION SNDRV_PROTOCOL_VERSION(2, 0, 13) typedef unsigned long snd_pcm_uframes_t; typedef signed long snd_pcm_sframes_t; @@ -267,10 +270,17 @@ typedef int __bitwise snd_pcm_subformat_t; #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_NO_PERIOD_WAKEUP 0x00800000 /* period wakeup can be disabled */ -#define SNDRV_PCM_INFO_HAS_WALL_CLOCK 0x01000000 /* has audio wall clock for audio/system time sync */ +#define SNDRV_PCM_INFO_HAS_WALL_CLOCK 0x01000000 /* (Deprecated)has audio wall clock for audio/system time sync */ +#define SNDRV_PCM_INFO_HAS_LINK_ATIME 0x01000000 /* report hardware link audio time, reset on startup */ +#define SNDRV_PCM_INFO_HAS_LINK_ABSOLUTE_ATIME 0x02000000 /* report absolute hardware link audio time, not reset on startup */ +#define SNDRV_PCM_INFO_HAS_LINK_ESTIMATED_ATIME 0x04000000 /* report estimated link audio time */ +#define SNDRV_PCM_INFO_HAS_LINK_SYNCHRONIZED_ATIME 0x08000000 /* report synchronized audio/system time */ + #define SNDRV_PCM_INFO_DRAIN_TRIGGER 0x40000000 /* internal kernel flag - trigger in drain */ #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 */ #define SNDRV_PCM_STATE_SETUP ((__force snd_pcm_state_t) 1) /* stream has a setup */ @@ -408,6 +418,22 @@ struct snd_pcm_channel_info { unsigned int step; /* samples distance in bits */ }; +enum { + /* + * first definition for backwards compatibility only, + * maps to wallclock/link time for HDAudio playback and DEFAULT/DMA time for everything else + */ + SNDRV_PCM_AUDIO_TSTAMP_TYPE_COMPAT = 0, + + /* timestamp definitions */ + SNDRV_PCM_AUDIO_TSTAMP_TYPE_DEFAULT = 1, /* DMA time, reported as per hw_ptr */ + SNDRV_PCM_AUDIO_TSTAMP_TYPE_LINK = 2, /* link time reported by sample or wallclock counter, reset on startup */ + SNDRV_PCM_AUDIO_TSTAMP_TYPE_LINK_ABSOLUTE = 3, /* link time reported by sample or wallclock counter, not reset on startup */ + SNDRV_PCM_AUDIO_TSTAMP_TYPE_LINK_ESTIMATED = 4, /* link time estimated indirectly */ + SNDRV_PCM_AUDIO_TSTAMP_TYPE_LINK_SYNCHRONIZED = 5, /* link time synchronized with system time */ + SNDRV_PCM_AUDIO_TSTAMP_TYPE_LAST = SNDRV_PCM_AUDIO_TSTAMP_TYPE_LINK_SYNCHRONIZED +}; + struct snd_pcm_status { snd_pcm_state_t state; /* stream state */ struct timespec trigger_tstamp; /* time when stream was started/stopped/paused */ @@ -419,9 +445,11 @@ struct snd_pcm_status { snd_pcm_uframes_t avail_max; /* max frames available on hw since last status */ snd_pcm_uframes_t overrange; /* count of ADC (capture) overrange detections from last status */ snd_pcm_state_t suspended_state; /* suspended stream state */ - __u32 reserved_alignment; /* must be filled with zero */ - struct timespec audio_tstamp; /* from sample counter or wall clock */ - unsigned char reserved[56-sizeof(struct timespec)]; /* must be filled with zero */ + __u32 audio_tstamp_data; /* needed for 64-bit alignment, used for configs/report to/from userspace */ + struct timespec audio_tstamp; /* sample counter, wall clock, PHC or on-demand sync'ed */ + struct timespec driver_tstamp; /* useful in case reference system tstamp is reported with delay */ + __u32 audio_tstamp_accuracy; /* in ns units, only valid if indicated in audio_tstamp_data */ + unsigned char reserved[52-2*sizeof(struct timespec)]; /* must be filled with zero */ }; struct snd_pcm_mmap_status { @@ -534,6 +562,7 @@ enum { #define SNDRV_PCM_IOCTL_DELAY _IOR('A', 0x21, snd_pcm_sframes_t) #define SNDRV_PCM_IOCTL_HWSYNC _IO('A', 0x22) #define SNDRV_PCM_IOCTL_SYNC_PTR _IOWR('A', 0x23, struct snd_pcm_sync_ptr) +#define SNDRV_PCM_IOCTL_STATUS_EXT _IOWR('A', 0x24, struct snd_pcm_status) #define SNDRV_PCM_IOCTL_CHANNEL_INFO _IOR('A', 0x32, struct snd_pcm_channel_info) #define SNDRV_PCM_IOCTL_PREPARE _IO('A', 0x40) #define SNDRV_PCM_IOCTL_RESET _IO('A', 0x41) diff --git a/include/uapi/sound/compress_offload.h b/include/uapi/sound/compress_offload.h index 22ed8cb..e00d8cb 100644 --- a/include/uapi/sound/compress_offload.h +++ b/include/uapi/sound/compress_offload.h @@ -75,7 +75,7 @@ struct snd_compr_tstamp { /** * struct snd_compr_avail - avail descriptor * @avail: Number of bytes available in ring buffer for writing/reading - * @tstamp: timestamp infomation + * @tstamp: timestamp information */ struct snd_compr_avail { __u64 avail; diff --git a/include/uapi/sound/emu10k1.h b/include/uapi/sound/emu10k1.h index d1bbaf7..ec1535b 100644 --- a/include/uapi/sound/emu10k1.h +++ b/include/uapi/sound/emu10k1.h @@ -23,8 +23,7 @@ #define _UAPI__SOUND_EMU10K1_H #include <linux/types.h> - - +#include <sound/asound.h> /* * ---- FX8010 ---- diff --git a/include/uapi/sound/hdspm.h b/include/uapi/sound/hdspm.h index b357f1a5..5737332 100644 --- a/include/uapi/sound/hdspm.h +++ b/include/uapi/sound/hdspm.h @@ -20,6 +20,12 @@ * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA. */ +#ifdef __KERNEL__ +#include <linux/types.h> +#else +#include <stdint.h> +#endif + /* Maximum channels is 64 even on 56Mode you have 64playbacks to matrix */ #define HDSPM_MAX_CHANNELS 64 diff --git a/sound/aoa/soundbus/i2sbus/core.c b/sound/aoa/soundbus/i2sbus/core.c index b9737fa..1cbf210 100644 --- a/sound/aoa/soundbus/i2sbus/core.c +++ b/sound/aoa/soundbus/i2sbus/core.c @@ -31,7 +31,7 @@ module_param(force, int, 0444); MODULE_PARM_DESC(force, "Force loading i2sbus even when" " no layout-id property is present"); -static struct of_device_id i2sbus_match[] = { +static const struct of_device_id i2sbus_match[] = { { .name = "i2s" }, { } }; diff --git a/sound/core/control.c b/sound/core/control.c index eeb691d..d677c27 100644 --- a/sound/core/control.c +++ b/sound/core/control.c @@ -192,36 +192,41 @@ void snd_ctl_notify(struct snd_card *card, unsigned int mask, EXPORT_SYMBOL(snd_ctl_notify); /** - * snd_ctl_new - create a control instance from the template - * @control: the control template - * @access: the default control access + * snd_ctl_new - create a new control instance with some elements + * @kctl: the pointer to store new control instance + * @count: the number of elements in this control + * @access: the default access flags for elements in this control + * @file: given when locking these elements * - * Allocates a new struct snd_kcontrol instance and copies the given template - * to the new instance. It does not copy volatile data (access). + * Allocates a memory object for a new control instance. The instance has + * elements as many as the given number (@count). Each element has given + * access permissions (@access). Each element is locked when @file is given. * - * Return: The pointer of the new instance, or %NULL on failure. + * Return: 0 on success, error code on failure */ -static struct snd_kcontrol *snd_ctl_new(struct snd_kcontrol *control, - unsigned int access) +static int snd_ctl_new(struct snd_kcontrol **kctl, unsigned int count, + unsigned int access, struct snd_ctl_file *file) { - struct snd_kcontrol *kctl; + unsigned int size; unsigned int idx; - if (snd_BUG_ON(!control || !control->count)) - return NULL; + if (count == 0 || count > MAX_CONTROL_COUNT) + return -EINVAL; - if (control->count > MAX_CONTROL_COUNT) - return NULL; + size = sizeof(struct snd_kcontrol); + size += sizeof(struct snd_kcontrol_volatile) * count; - kctl = kzalloc(sizeof(*kctl) + sizeof(struct snd_kcontrol_volatile) * control->count, GFP_KERNEL); - if (kctl == NULL) { - pr_err("ALSA: Cannot allocate control instance\n"); - return NULL; + *kctl = kzalloc(size, GFP_KERNEL); + if (!*kctl) + return -ENOMEM; + + for (idx = 0; idx < count; idx++) { + (*kctl)->vd[idx].access = access; + (*kctl)->vd[idx].owner = file; } - *kctl = *control; - for (idx = 0; idx < kctl->count; idx++) - kctl->vd[idx].access = access; - return kctl; + (*kctl)->count = count; + + return 0; } /** @@ -238,37 +243,53 @@ static struct snd_kcontrol *snd_ctl_new(struct snd_kcontrol *control, struct snd_kcontrol *snd_ctl_new1(const struct snd_kcontrol_new *ncontrol, void *private_data) { - struct snd_kcontrol kctl; + struct snd_kcontrol *kctl; + unsigned int count; unsigned int access; + int err; if (snd_BUG_ON(!ncontrol || !ncontrol->info)) return NULL; - memset(&kctl, 0, sizeof(kctl)); - kctl.id.iface = ncontrol->iface; - kctl.id.device = ncontrol->device; - kctl.id.subdevice = ncontrol->subdevice; + + count = ncontrol->count; + if (count == 0) + count = 1; + + access = ncontrol->access; + if (access == 0) + access = SNDRV_CTL_ELEM_ACCESS_READWRITE; + access &= (SNDRV_CTL_ELEM_ACCESS_READWRITE | + SNDRV_CTL_ELEM_ACCESS_VOLATILE | + SNDRV_CTL_ELEM_ACCESS_INACTIVE | + SNDRV_CTL_ELEM_ACCESS_TLV_READWRITE | + SNDRV_CTL_ELEM_ACCESS_TLV_COMMAND | + SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK); + + err = snd_ctl_new(&kctl, count, access, NULL); + if (err < 0) + return NULL; + + /* The 'numid' member is decided when calling snd_ctl_add(). */ + kctl->id.iface = ncontrol->iface; + kctl->id.device = ncontrol->device; + kctl->id.subdevice = ncontrol->subdevice; if (ncontrol->name) { - strlcpy(kctl.id.name, ncontrol->name, sizeof(kctl.id.name)); - if (strcmp(ncontrol->name, kctl.id.name) != 0) + strlcpy(kctl->id.name, ncontrol->name, sizeof(kctl->id.name)); + if (strcmp(ncontrol->name, kctl->id.name) != 0) pr_warn("ALSA: Control name '%s' truncated to '%s'\n", - ncontrol->name, kctl.id.name); + ncontrol->name, kctl->id.name); } - kctl.id.index = ncontrol->index; - kctl.count = ncontrol->count ? ncontrol->count : 1; - access = ncontrol->access == 0 ? SNDRV_CTL_ELEM_ACCESS_READWRITE : - (ncontrol->access & (SNDRV_CTL_ELEM_ACCESS_READWRITE| - SNDRV_CTL_ELEM_ACCESS_VOLATILE| - SNDRV_CTL_ELEM_ACCESS_INACTIVE| - SNDRV_CTL_ELEM_ACCESS_TLV_READWRITE| - SNDRV_CTL_ELEM_ACCESS_TLV_COMMAND| - SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK)); - kctl.info = ncontrol->info; - kctl.get = ncontrol->get; - kctl.put = ncontrol->put; - kctl.tlv.p = ncontrol->tlv.p; - kctl.private_value = ncontrol->private_value; - kctl.private_data = private_data; - return snd_ctl_new(&kctl, access); + kctl->id.index = ncontrol->index; + + kctl->info = ncontrol->info; + kctl->get = ncontrol->get; + kctl->put = ncontrol->put; + kctl->tlv.p = ncontrol->tlv.p; + + kctl->private_value = ncontrol->private_value; + kctl->private_data = private_data; + + return kctl; } EXPORT_SYMBOL(snd_ctl_new1); @@ -1161,84 +1182,102 @@ static void snd_ctl_elem_user_free(struct snd_kcontrol *kcontrol) static int snd_ctl_elem_add(struct snd_ctl_file *file, struct snd_ctl_elem_info *info, int replace) { + /* The capacity of struct snd_ctl_elem_value.value.*/ + static const unsigned int value_sizes[] = { + [SNDRV_CTL_ELEM_TYPE_BOOLEAN] = sizeof(long), + [SNDRV_CTL_ELEM_TYPE_INTEGER] = sizeof(long), + [SNDRV_CTL_ELEM_TYPE_ENUMERATED] = sizeof(unsigned int), + [SNDRV_CTL_ELEM_TYPE_BYTES] = sizeof(unsigned char), + [SNDRV_CTL_ELEM_TYPE_IEC958] = sizeof(struct snd_aes_iec958), + [SNDRV_CTL_ELEM_TYPE_INTEGER64] = sizeof(long long), + }; + static const unsigned int max_value_counts[] = { + [SNDRV_CTL_ELEM_TYPE_BOOLEAN] = 128, + [SNDRV_CTL_ELEM_TYPE_INTEGER] = 128, + [SNDRV_CTL_ELEM_TYPE_ENUMERATED] = 128, + [SNDRV_CTL_ELEM_TYPE_BYTES] = 512, + [SNDRV_CTL_ELEM_TYPE_IEC958] = 1, + [SNDRV_CTL_ELEM_TYPE_INTEGER64] = 64, + }; struct snd_card *card = file->card; - struct snd_kcontrol kctl, *_kctl; + struct snd_kcontrol *kctl; + unsigned int count; unsigned int access; long private_size; struct user_element *ue; - int idx, err; + int err; - if (info->count < 1) - return -EINVAL; if (!*info->id.name) return -EINVAL; if (strnlen(info->id.name, sizeof(info->id.name)) >= sizeof(info->id.name)) return -EINVAL; - access = info->access == 0 ? SNDRV_CTL_ELEM_ACCESS_READWRITE : - (info->access & (SNDRV_CTL_ELEM_ACCESS_READWRITE| - SNDRV_CTL_ELEM_ACCESS_INACTIVE| - SNDRV_CTL_ELEM_ACCESS_TLV_READWRITE)); - info->id.numid = 0; - memset(&kctl, 0, sizeof(kctl)); + /* Delete a control to replace them if needed. */ if (replace) { + info->id.numid = 0; err = snd_ctl_remove_user_ctl(file, &info->id); if (err) return err; } - if (card->user_ctl_count >= MAX_USER_CONTROLS) + /* + * The number of userspace controls are counted control by control, + * not element by element. + */ + if (card->user_ctl_count + 1 > MAX_USER_CONTROLS) return -ENOMEM; - memcpy(&kctl.id, &info->id, sizeof(info->id)); - kctl.count = info->owner ? info->owner : 1; - access |= SNDRV_CTL_ELEM_ACCESS_USER; - if (info->type == SNDRV_CTL_ELEM_TYPE_ENUMERATED) - kctl.info = snd_ctl_elem_user_enum_info; - else - kctl.info = snd_ctl_elem_user_info; - if (access & SNDRV_CTL_ELEM_ACCESS_READ) - kctl.get = snd_ctl_elem_user_get; - if (access & SNDRV_CTL_ELEM_ACCESS_WRITE) - kctl.put = snd_ctl_elem_user_put; - if (access & SNDRV_CTL_ELEM_ACCESS_TLV_READWRITE) { - kctl.tlv.c = snd_ctl_elem_user_tlv; + /* Check the number of elements for this userspace control. */ + count = info->owner; + if (count == 0) + count = 1; + + /* Arrange access permissions if needed. */ + access = info->access; + if (access == 0) + access = SNDRV_CTL_ELEM_ACCESS_READWRITE; + access &= (SNDRV_CTL_ELEM_ACCESS_READWRITE | + SNDRV_CTL_ELEM_ACCESS_INACTIVE | + SNDRV_CTL_ELEM_ACCESS_TLV_READWRITE); + if (access & SNDRV_CTL_ELEM_ACCESS_TLV_READWRITE) access |= SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK; - } - switch (info->type) { - case SNDRV_CTL_ELEM_TYPE_BOOLEAN: - case SNDRV_CTL_ELEM_TYPE_INTEGER: - private_size = sizeof(long); - if (info->count > 128) - return -EINVAL; - break; - case SNDRV_CTL_ELEM_TYPE_INTEGER64: - private_size = sizeof(long long); - if (info->count > 64) - return -EINVAL; - break; - case SNDRV_CTL_ELEM_TYPE_ENUMERATED: - private_size = sizeof(unsigned int); - if (info->count > 128 || info->value.enumerated.items == 0) - return -EINVAL; - break; - case SNDRV_CTL_ELEM_TYPE_BYTES: - private_size = sizeof(unsigned char); - if (info->count > 512) - return -EINVAL; - break; - case SNDRV_CTL_ELEM_TYPE_IEC958: - private_size = sizeof(struct snd_aes_iec958); - if (info->count != 1) - return -EINVAL; - break; - default: + access |= SNDRV_CTL_ELEM_ACCESS_USER; + + /* + * Check information and calculate the size of data specific to + * this userspace control. + */ + if (info->type < SNDRV_CTL_ELEM_TYPE_BOOLEAN || + info->type > SNDRV_CTL_ELEM_TYPE_INTEGER64) return -EINVAL; - } - private_size *= info->count; - ue = kzalloc(sizeof(struct user_element) + private_size, GFP_KERNEL); - if (ue == NULL) + if (info->type == SNDRV_CTL_ELEM_TYPE_ENUMERATED && + info->value.enumerated.items == 0) + return -EINVAL; + if (info->count < 1 || + info->count > max_value_counts[info->type]) + return -EINVAL; + private_size = value_sizes[info->type] * info->count; + + /* + * Keep memory object for this userspace control. After passing this + * code block, the instance should be freed by snd_ctl_free_one(). + * + * Note that these elements in this control are locked. + */ + err = snd_ctl_new(&kctl, count, access, file); + if (err < 0) + return err; + memcpy(&kctl->id, &info->id, sizeof(kctl->id)); + kctl->private_data = kzalloc(sizeof(struct user_element) + private_size, + GFP_KERNEL); + if (kctl->private_data == NULL) { + kfree(kctl); return -ENOMEM; + } + kctl->private_free = snd_ctl_elem_user_free; + + /* Set private data for this userspace control. */ + ue = (struct user_element *)kctl->private_data; ue->card = card; ue->info = *info; ue->info.access = 0; @@ -1247,21 +1286,25 @@ static int snd_ctl_elem_add(struct snd_ctl_file *file, if (ue->info.type == SNDRV_CTL_ELEM_TYPE_ENUMERATED) { err = snd_ctl_elem_init_enum_names(ue); if (err < 0) { - kfree(ue); + snd_ctl_free_one(kctl); return err; } } - kctl.private_free = snd_ctl_elem_user_free; - _kctl = snd_ctl_new(&kctl, access); - if (_kctl == NULL) { - kfree(ue->priv_data); - kfree(ue); - return -ENOMEM; - } - _kctl->private_data = ue; - for (idx = 0; idx < _kctl->count; idx++) - _kctl->vd[idx].owner = file; - err = snd_ctl_add(card, _kctl); + + /* Set callback functions. */ + if (info->type == SNDRV_CTL_ELEM_TYPE_ENUMERATED) + kctl->info = snd_ctl_elem_user_enum_info; + else + kctl->info = snd_ctl_elem_user_info; + if (access & SNDRV_CTL_ELEM_ACCESS_READ) + kctl->get = snd_ctl_elem_user_get; + if (access & SNDRV_CTL_ELEM_ACCESS_WRITE) + kctl->put = snd_ctl_elem_user_put; + if (access & SNDRV_CTL_ELEM_ACCESS_TLV_READWRITE) + kctl->tlv.c = snd_ctl_elem_user_tlv; + + /* This function manage to free the instance on failure. */ + err = snd_ctl_add(card, kctl); if (err < 0) return err; diff --git a/sound/core/device.c b/sound/core/device.c index 41bec30..8918838 100644 --- a/sound/core/device.c +++ b/sound/core/device.c @@ -50,10 +50,8 @@ int snd_device_new(struct snd_card *card, enum snd_device_type type, if (snd_BUG_ON(!card || !device_data || !ops)) return -ENXIO; dev = kzalloc(sizeof(*dev), GFP_KERNEL); - if (dev == NULL) { - dev_err(card->dev, "Cannot allocate device, type=%d\n", type); + if (!dev) return -ENOMEM; - } INIT_LIST_HEAD(&dev->list); dev->card = card; dev->type = type; @@ -73,7 +71,7 @@ int snd_device_new(struct snd_card *card, enum snd_device_type type, } EXPORT_SYMBOL(snd_device_new); -static int __snd_device_disconnect(struct snd_device *dev) +static void __snd_device_disconnect(struct snd_device *dev) { if (dev->state == SNDRV_DEV_REGISTERED) { if (dev->ops->dev_disconnect && @@ -81,7 +79,6 @@ static int __snd_device_disconnect(struct snd_device *dev) dev_err(dev->card->dev, "device disconnect failure\n"); dev->state = SNDRV_DEV_DISCONNECTED; } - return 0; } static void __snd_device_free(struct snd_device *dev) @@ -109,6 +106,34 @@ static struct snd_device *look_for_dev(struct snd_card *card, void *device_data) } /** + * snd_device_disconnect - disconnect the device + * @card: the card instance + * @device_data: the data pointer to disconnect + * + * Turns the device into the disconnection state, invoking + * dev_disconnect callback, if the device was already registered. + * + * Usually called from snd_card_disconnect(). + * + * Return: Zero if successful, or a negative error code on failure or if the + * device not found. + */ +void snd_device_disconnect(struct snd_card *card, void *device_data) +{ + struct snd_device *dev; + + if (snd_BUG_ON(!card || !device_data)) + return; + dev = look_for_dev(card, device_data); + if (dev) + __snd_device_disconnect(dev); + else + dev_dbg(card->dev, "device disconnect %p (from %pF), not found\n", + device_data, __builtin_return_address(0)); +} +EXPORT_SYMBOL_GPL(snd_device_disconnect); + +/** * snd_device_free - release the device from the card * @card: the card instance * @device_data: the data pointer to release @@ -195,18 +220,14 @@ int snd_device_register_all(struct snd_card *card) * disconnect all the devices on the card. * called from init.c */ -int snd_device_disconnect_all(struct snd_card *card) +void snd_device_disconnect_all(struct snd_card *card) { struct snd_device *dev; - int err = 0; if (snd_BUG_ON(!card)) - return -ENXIO; - list_for_each_entry_reverse(dev, &card->devices, list) { - if (__snd_device_disconnect(dev) < 0) - err = -ENXIO; - } - return err; + return; + list_for_each_entry_reverse(dev, &card->devices, list) + __snd_device_disconnect(dev); } /* diff --git a/sound/core/hwdep.c b/sound/core/hwdep.c index 84244a5..51692c8 100644 --- a/sound/core/hwdep.c +++ b/sound/core/hwdep.c @@ -378,10 +378,8 @@ int snd_hwdep_new(struct snd_card *card, char *id, int device, if (rhwdep) *rhwdep = NULL; hwdep = kzalloc(sizeof(*hwdep), GFP_KERNEL); - if (hwdep == NULL) { - dev_err(card->dev, "hwdep: cannot allocate\n"); + if (!hwdep) return -ENOMEM; - } init_waitqueue_head(&hwdep->open_wait); mutex_init(&hwdep->open_mutex); diff --git a/sound/core/init.c b/sound/core/init.c index 3541905..04734e0 100644 --- a/sound/core/init.c +++ b/sound/core/init.c @@ -400,7 +400,6 @@ static const struct file_operations snd_shutdown_f_ops = int snd_card_disconnect(struct snd_card *card) { struct snd_monitor_file *mfile; - int err; if (!card) return -EINVAL; @@ -445,9 +444,7 @@ int snd_card_disconnect(struct snd_card *card) #endif /* notify all devices that we are disconnected */ - err = snd_device_disconnect_all(card); - if (err < 0) - dev_err(card->dev, "not all devices for card %i can be disconnected\n", card->number); + snd_device_disconnect_all(card); snd_info_card_disconnect(card); if (card->registered) { diff --git a/sound/core/oss/mixer_oss.c b/sound/core/oss/mixer_oss.c index 5e6349f..056f8e2 100644 --- a/sound/core/oss/mixer_oss.c +++ b/sound/core/oss/mixer_oss.c @@ -1212,10 +1212,8 @@ static void snd_mixer_oss_proc_write(struct snd_info_entry *entry, /* not changed */ goto __unlock; tbl = kmalloc(sizeof(*tbl), GFP_KERNEL); - if (! tbl) { - pr_err("ALSA: mixer_oss: no memory\n"); + if (!tbl) goto __unlock; - } tbl->oss_id = ch; tbl->name = kstrdup(str, GFP_KERNEL); if (! tbl->name) { diff --git a/sound/core/oss/pcm_oss.c b/sound/core/oss/pcm_oss.c index 80423a4c..58550cc 100644 --- a/sound/core/oss/pcm_oss.c +++ b/sound/core/oss/pcm_oss.c @@ -854,7 +854,6 @@ static int snd_pcm_oss_change_params(struct snd_pcm_substream *substream) params = kmalloc(sizeof(*params), GFP_KERNEL); sparams = kmalloc(sizeof(*sparams), GFP_KERNEL); if (!sw_params || !params || !sparams) { - pcm_dbg(substream->pcm, "No memory\n"); err = -ENOMEM; goto failure; } diff --git a/sound/core/pcm.c b/sound/core/pcm.c index 0345e53..b25bcf5 100644 --- a/sound/core/pcm.c +++ b/sound/core/pcm.c @@ -49,8 +49,6 @@ static struct snd_pcm *snd_pcm_get(struct snd_card *card, int device) struct snd_pcm *pcm; list_for_each_entry(pcm, &snd_pcm_devices, list) { - if (pcm->internal) - continue; if (pcm->card == card && pcm->device == device) return pcm; } @@ -62,8 +60,6 @@ static int snd_pcm_next(struct snd_card *card, int device) struct snd_pcm *pcm; list_for_each_entry(pcm, &snd_pcm_devices, list) { - if (pcm->internal) - continue; if (pcm->card == card && pcm->device > device) return pcm->device; else if (pcm->card->number > card->number) @@ -76,6 +72,9 @@ static int snd_pcm_add(struct snd_pcm *newpcm) { struct snd_pcm *pcm; + if (newpcm->internal) + return 0; + list_for_each_entry(pcm, &snd_pcm_devices, list) { if (pcm->card == newpcm->card && pcm->device == newpcm->device) return -EBUSY; @@ -344,11 +343,8 @@ static void snd_pcm_proc_info_read(struct snd_pcm_substream *substream, return; info = kmalloc(sizeof(*info), GFP_KERNEL); - if (! info) { - pcm_dbg(substream->pcm, - "snd_pcm_proc_info_read: cannot malloc\n"); + if (!info) return; - } err = snd_pcm_info(substream, info); if (err < 0) { @@ -718,10 +714,8 @@ int snd_pcm_new_stream(struct snd_pcm *pcm, int stream, int substream_count) prev = NULL; for (idx = 0, prev = NULL; idx < substream_count; idx++) { substream = kzalloc(sizeof(*substream), GFP_KERNEL); - if (substream == NULL) { - pcm_err(pcm, "Cannot allocate PCM substream\n"); + if (!substream) return -ENOMEM; - } substream->pcm = pcm; substream->pstr = pstr; substream->number = idx; @@ -775,13 +769,14 @@ static int _snd_pcm_new(struct snd_card *card, const char *id, int device, if (rpcm) *rpcm = NULL; pcm = kzalloc(sizeof(*pcm), GFP_KERNEL); - if (pcm == NULL) { - dev_err(card->dev, "Cannot allocate PCM\n"); + if (!pcm) return -ENOMEM; - } pcm->card = card; pcm->device = device; pcm->internal = internal; + mutex_init(&pcm->open_mutex); + init_waitqueue_head(&pcm->open_wait); + INIT_LIST_HEAD(&pcm->list); if (id) strlcpy(pcm->id, id, sizeof(pcm->id)); if ((err = snd_pcm_new_stream(pcm, SNDRV_PCM_STREAM_PLAYBACK, playback_count)) < 0) { @@ -792,8 +787,6 @@ static int _snd_pcm_new(struct snd_card *card, const char *id, int device, snd_pcm_free(pcm); return err; } - mutex_init(&pcm->open_mutex); - init_waitqueue_head(&pcm->open_wait); if ((err = snd_device_new(card, SNDRV_DEV_PCM, pcm, &ops)) < 0) { snd_pcm_free(pcm); return err; @@ -888,8 +881,9 @@ static int snd_pcm_free(struct snd_pcm *pcm) if (!pcm) return 0; - list_for_each_entry(notify, &snd_pcm_notify_list, list) { - notify->n_unregister(pcm); + if (!pcm->internal) { + list_for_each_entry(notify, &snd_pcm_notify_list, list) + notify->n_unregister(pcm); } if (pcm->private_free) pcm->private_free(pcm); @@ -919,6 +913,9 @@ int snd_pcm_attach_substream(struct snd_pcm *pcm, int stream, if (snd_BUG_ON(!pcm || !rsubstream)) return -ENXIO; + if (snd_BUG_ON(stream != SNDRV_PCM_STREAM_PLAYBACK && + stream != SNDRV_PCM_STREAM_CAPTURE)) + return -EINVAL; *rsubstream = NULL; pstr = &pcm->streams[stream]; if (pstr->substream == NULL || pstr->substream_count == 0) @@ -927,25 +924,14 @@ int snd_pcm_attach_substream(struct snd_pcm *pcm, int stream, card = pcm->card; prefer_subdevice = snd_ctl_get_preferred_subdevice(card, SND_CTL_SUBDEV_PCM); - switch (stream) { - case SNDRV_PCM_STREAM_PLAYBACK: - if (pcm->info_flags & SNDRV_PCM_INFO_HALF_DUPLEX) { - for (substream = pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream; substream; substream = substream->next) { - if (SUBSTREAM_BUSY(substream)) - return -EAGAIN; - } - } - break; - case SNDRV_PCM_STREAM_CAPTURE: - if (pcm->info_flags & SNDRV_PCM_INFO_HALF_DUPLEX) { - for (substream = pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream; substream; substream = substream->next) { - if (SUBSTREAM_BUSY(substream)) - return -EAGAIN; - } + if (pcm->info_flags & SNDRV_PCM_INFO_HALF_DUPLEX) { + int opposite = !stream; + + for (substream = pcm->streams[opposite].substream; substream; + substream = substream->next) { + if (SUBSTREAM_BUSY(substream)) + return -EAGAIN; } - break; - default: - return -EINVAL; } if (file->f_flags & O_APPEND) { @@ -968,15 +954,12 @@ int snd_pcm_attach_substream(struct snd_pcm *pcm, int stream, return 0; } - if (prefer_subdevice >= 0) { - for (substream = pstr->substream; substream; substream = substream->next) - if (!SUBSTREAM_BUSY(substream) && substream->number == prefer_subdevice) - goto __ok; - } - for (substream = pstr->substream; substream; substream = substream->next) - if (!SUBSTREAM_BUSY(substream)) + for (substream = pstr->substream; substream; substream = substream->next) { + if (!SUBSTREAM_BUSY(substream) && + (prefer_subdevice == -1 || + substream->number == prefer_subdevice)) break; - __ok: + } if (substream == NULL) return -EAGAIN; @@ -1086,15 +1069,16 @@ static int snd_pcm_dev_register(struct snd_device *device) if (snd_BUG_ON(!device || !device->device_data)) return -ENXIO; pcm = device->device_data; + if (pcm->internal) + return 0; + mutex_lock(®ister_mutex); err = snd_pcm_add(pcm); - if (err) { - mutex_unlock(®ister_mutex); - return err; - } + if (err) + goto unlock; for (cidx = 0; cidx < 2; cidx++) { int devtype = -1; - if (pcm->streams[cidx].substream == NULL || pcm->internal) + if (pcm->streams[cidx].substream == NULL) continue; switch (cidx) { case SNDRV_PCM_STREAM_PLAYBACK: @@ -1109,9 +1093,8 @@ static int snd_pcm_dev_register(struct snd_device *device) &snd_pcm_f_ops[cidx], pcm, &pcm->streams[cidx].dev); if (err < 0) { - list_del(&pcm->list); - mutex_unlock(®ister_mutex); - return err; + list_del_init(&pcm->list); + goto unlock; } for (substream = pcm->streams[cidx].substream; substream; substream = substream->next) @@ -1121,8 +1104,9 @@ static int snd_pcm_dev_register(struct snd_device *device) list_for_each_entry(notify, &snd_pcm_notify_list, list) notify->n_register(pcm); + unlock: mutex_unlock(®ister_mutex); - return 0; + return err; } static int snd_pcm_dev_disconnect(struct snd_device *device) @@ -1133,13 +1117,10 @@ static int snd_pcm_dev_disconnect(struct snd_device *device) int cidx; mutex_lock(®ister_mutex); - if (list_empty(&pcm->list)) - goto unlock; - mutex_lock(&pcm->open_mutex); wake_up(&pcm->open_wait); list_del_init(&pcm->list); - for (cidx = 0; cidx < 2; cidx++) + for (cidx = 0; cidx < 2; cidx++) { for (substream = pcm->streams[cidx].substream; substream; substream = substream->next) { snd_pcm_stream_lock_irq(substream); if (substream->runtime) { @@ -1149,18 +1130,20 @@ static int snd_pcm_dev_disconnect(struct snd_device *device) } snd_pcm_stream_unlock_irq(substream); } - list_for_each_entry(notify, &snd_pcm_notify_list, list) { - notify->n_disconnect(pcm); + } + if (!pcm->internal) { + list_for_each_entry(notify, &snd_pcm_notify_list, list) + notify->n_disconnect(pcm); } for (cidx = 0; cidx < 2; cidx++) { - snd_unregister_device(&pcm->streams[cidx].dev); + if (!pcm->internal) + snd_unregister_device(&pcm->streams[cidx].dev); if (pcm->streams[cidx].chmap_kctl) { snd_ctl_remove(pcm->card, pcm->streams[cidx].chmap_kctl); pcm->streams[cidx].chmap_kctl = NULL; } } mutex_unlock(&pcm->open_mutex); - unlock: mutex_unlock(®ister_mutex); return 0; } diff --git a/sound/core/pcm_compat.c b/sound/core/pcm_compat.c index 2d957ba..b48b434 100644 --- a/sound/core/pcm_compat.c +++ b/sound/core/pcm_compat.c @@ -194,18 +194,30 @@ struct snd_pcm_status32 { u32 avail_max; u32 overrange; s32 suspended_state; - u32 reserved_alignment; + u32 audio_tstamp_data; struct compat_timespec audio_tstamp; - unsigned char reserved[56-sizeof(struct compat_timespec)]; + struct compat_timespec driver_tstamp; + u32 audio_tstamp_accuracy; + unsigned char reserved[52-2*sizeof(struct compat_timespec)]; } __attribute__((packed)); static int snd_pcm_status_user_compat(struct snd_pcm_substream *substream, - struct snd_pcm_status32 __user *src) + struct snd_pcm_status32 __user *src, + bool ext) { struct snd_pcm_status status; int err; + memset(&status, 0, sizeof(status)); + /* + * with extension, parameters are read/write, + * get audio_tstamp_data from user, + * ignore rest of status structure + */ + if (ext && get_user(status.audio_tstamp_data, + (u32 __user *)(&src->audio_tstamp_data))) + return -EFAULT; err = snd_pcm_status(substream, &status); if (err < 0) return err; @@ -222,7 +234,10 @@ static int snd_pcm_status_user_compat(struct snd_pcm_substream *substream, put_user(status.avail_max, &src->avail_max) || put_user(status.overrange, &src->overrange) || put_user(status.suspended_state, &src->suspended_state) || - compat_put_timespec(&status.audio_tstamp, &src->audio_tstamp)) + put_user(status.audio_tstamp_data, &src->audio_tstamp_data) || + compat_put_timespec(&status.audio_tstamp, &src->audio_tstamp) || + compat_put_timespec(&status.driver_tstamp, &src->driver_tstamp) || + put_user(status.audio_tstamp_accuracy, &src->audio_tstamp_accuracy)) return -EFAULT; return err; @@ -457,6 +472,7 @@ enum { SNDRV_PCM_IOCTL_HW_PARAMS32 = _IOWR('A', 0x11, struct snd_pcm_hw_params32), SNDRV_PCM_IOCTL_SW_PARAMS32 = _IOWR('A', 0x13, struct snd_pcm_sw_params32), SNDRV_PCM_IOCTL_STATUS32 = _IOR('A', 0x20, struct snd_pcm_status32), + SNDRV_PCM_IOCTL_STATUS_EXT32 = _IOWR('A', 0x24, struct snd_pcm_status32), SNDRV_PCM_IOCTL_DELAY32 = _IOR('A', 0x21, s32), SNDRV_PCM_IOCTL_CHANNEL_INFO32 = _IOR('A', 0x32, struct snd_pcm_channel_info32), SNDRV_PCM_IOCTL_REWIND32 = _IOW('A', 0x46, u32), @@ -517,7 +533,9 @@ static long snd_pcm_ioctl_compat(struct file *file, unsigned int cmd, unsigned l case SNDRV_PCM_IOCTL_SW_PARAMS32: return snd_pcm_ioctl_sw_params_compat(substream, argp); case SNDRV_PCM_IOCTL_STATUS32: - return snd_pcm_status_user_compat(substream, argp); + return snd_pcm_status_user_compat(substream, argp, false); + case SNDRV_PCM_IOCTL_STATUS_EXT32: + return snd_pcm_status_user_compat(substream, argp, true); case SNDRV_PCM_IOCTL_SYNC_PTR32: return snd_pcm_ioctl_sync_ptr_compat(substream, argp); case SNDRV_PCM_IOCTL_CHANNEL_INFO32: diff --git a/sound/core/pcm_dmaengine.c b/sound/core/pcm_dmaengine.c index 6542c40..fba365a 100644 --- a/sound/core/pcm_dmaengine.c +++ b/sound/core/pcm_dmaengine.c @@ -289,7 +289,7 @@ EXPORT_SYMBOL_GPL(snd_dmaengine_pcm_request_channel); * * The function should usually be called from the pcm open callback. Note that * this function will use private_data field of the substream's runtime. So it - * is not availabe to your pcm driver implementation. + * is not available to your pcm driver implementation. */ int snd_dmaengine_pcm_open(struct snd_pcm_substream *substream, struct dma_chan *chan) @@ -328,7 +328,7 @@ EXPORT_SYMBOL_GPL(snd_dmaengine_pcm_open); * This function will request a DMA channel using the passed filter function and * data. The function should usually be called from the pcm open callback. Note * that this function will use private_data field of the substream's runtime. So - * it is not availabe to your pcm driver implementation. + * it is not available to your pcm driver implementation. */ int snd_dmaengine_pcm_open_request_chan(struct snd_pcm_substream *substream, dma_filter_fn filter_fn, void *filter_data) diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index ffd6560..ac6b33f 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -232,6 +232,49 @@ int snd_pcm_update_state(struct snd_pcm_substream *substream, return 0; } +static void update_audio_tstamp(struct snd_pcm_substream *substream, + struct timespec *curr_tstamp, + struct timespec *audio_tstamp) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + u64 audio_frames, audio_nsecs; + struct timespec driver_tstamp; + + if (runtime->tstamp_mode != SNDRV_PCM_TSTAMP_ENABLE) + return; + + if (!(substream->ops->get_time_info) || + (runtime->audio_tstamp_report.actual_type == + SNDRV_PCM_AUDIO_TSTAMP_TYPE_DEFAULT)) { + + /* + * provide audio timestamp derived from pointer position + * add delay only if requested + */ + + audio_frames = runtime->hw_ptr_wrap + runtime->status->hw_ptr; + + if (runtime->audio_tstamp_config.report_delay) { + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + audio_frames -= runtime->delay; + else + audio_frames += runtime->delay; + } + audio_nsecs = div_u64(audio_frames * 1000000000LL, + runtime->rate); + *audio_tstamp = ns_to_timespec(audio_nsecs); + } + runtime->status->audio_tstamp = *audio_tstamp; + runtime->status->tstamp = *curr_tstamp; + + /* + * re-take a driver timestamp to let apps detect if the reference tstamp + * read by low-level hardware was provided with a delay + */ + snd_pcm_gettime(substream->runtime, (struct timespec *)&driver_tstamp); + runtime->driver_tstamp = driver_tstamp; +} + static int snd_pcm_update_hw_ptr0(struct snd_pcm_substream *substream, unsigned int in_interrupt) { @@ -256,11 +299,18 @@ static int snd_pcm_update_hw_ptr0(struct snd_pcm_substream *substream, pos = substream->ops->pointer(substream); curr_jiffies = jiffies; if (runtime->tstamp_mode == SNDRV_PCM_TSTAMP_ENABLE) { - snd_pcm_gettime(runtime, (struct timespec *)&curr_tstamp); - - if ((runtime->hw.info & SNDRV_PCM_INFO_HAS_WALL_CLOCK) && - (substream->ops->wall_clock)) - substream->ops->wall_clock(substream, &audio_tstamp); + if ((substream->ops->get_time_info) && + (runtime->audio_tstamp_config.type_requested != SNDRV_PCM_AUDIO_TSTAMP_TYPE_DEFAULT)) { + substream->ops->get_time_info(substream, &curr_tstamp, + &audio_tstamp, + &runtime->audio_tstamp_config, + &runtime->audio_tstamp_report); + + /* re-test in case tstamp type is not supported in hardware and was demoted to DEFAULT */ + if (runtime->audio_tstamp_report.actual_type == SNDRV_PCM_AUDIO_TSTAMP_TYPE_DEFAULT) + snd_pcm_gettime(runtime, (struct timespec *)&curr_tstamp); + } else + snd_pcm_gettime(runtime, (struct timespec *)&curr_tstamp); } if (pos == SNDRV_PCM_POS_XRUN) { @@ -403,8 +453,10 @@ static int snd_pcm_update_hw_ptr0(struct snd_pcm_substream *substream, } no_delta_check: - if (runtime->status->hw_ptr == new_hw_ptr) + if (runtime->status->hw_ptr == new_hw_ptr) { + update_audio_tstamp(substream, &curr_tstamp, &audio_tstamp); return 0; + } if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK && runtime->silence_size > 0) @@ -426,30 +478,8 @@ static int snd_pcm_update_hw_ptr0(struct snd_pcm_substream *substream, snd_BUG_ON(crossed_boundary != 1); runtime->hw_ptr_wrap += runtime->boundary; } - if (runtime->tstamp_mode == SNDRV_PCM_TSTAMP_ENABLE) { - runtime->status->tstamp = curr_tstamp; - if (!(runtime->hw.info & SNDRV_PCM_INFO_HAS_WALL_CLOCK)) { - /* - * no wall clock available, provide audio timestamp - * derived from pointer position+delay - */ - u64 audio_frames, audio_nsecs; - - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - audio_frames = runtime->hw_ptr_wrap - + runtime->status->hw_ptr - - runtime->delay; - else - audio_frames = runtime->hw_ptr_wrap - + runtime->status->hw_ptr - + runtime->delay; - audio_nsecs = div_u64(audio_frames * 1000000000LL, - runtime->rate); - audio_tstamp = ns_to_timespec(audio_nsecs); - } - runtime->status->audio_tstamp = audio_tstamp; - } + update_audio_tstamp(substream, &curr_tstamp, &audio_tstamp); return snd_pcm_update_state(substream, runtime); } diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 279e24f..abe1e81 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -707,6 +707,23 @@ int snd_pcm_status(struct snd_pcm_substream *substream, struct snd_pcm_runtime *runtime = substream->runtime; snd_pcm_stream_lock_irq(substream); + + snd_pcm_unpack_audio_tstamp_config(status->audio_tstamp_data, + &runtime->audio_tstamp_config); + + /* backwards compatible behavior */ + if (runtime->audio_tstamp_config.type_requested == + SNDRV_PCM_AUDIO_TSTAMP_TYPE_COMPAT) { + if (runtime->hw.info & SNDRV_PCM_INFO_HAS_WALL_CLOCK) + runtime->audio_tstamp_config.type_requested = + SNDRV_PCM_AUDIO_TSTAMP_TYPE_LINK; + else + runtime->audio_tstamp_config.type_requested = + SNDRV_PCM_AUDIO_TSTAMP_TYPE_DEFAULT; + runtime->audio_tstamp_report.valid = 0; + } else + runtime->audio_tstamp_report.valid = 1; + status->state = runtime->status->state; status->suspended_state = runtime->status->suspended_state; if (status->state == SNDRV_PCM_STATE_OPEN) @@ -716,8 +733,15 @@ int snd_pcm_status(struct snd_pcm_substream *substream, snd_pcm_update_hw_ptr(substream); if (runtime->tstamp_mode == SNDRV_PCM_TSTAMP_ENABLE) { status->tstamp = runtime->status->tstamp; + status->driver_tstamp = runtime->driver_tstamp; status->audio_tstamp = runtime->status->audio_tstamp; + if (runtime->audio_tstamp_report.valid == 1) + /* backwards compatibility, no report provided in COMPAT mode */ + snd_pcm_pack_audio_tstamp_report(&status->audio_tstamp_data, + &status->audio_tstamp_accuracy, + &runtime->audio_tstamp_report); + goto _tstamp_end; } } else { @@ -753,12 +777,21 @@ int snd_pcm_status(struct snd_pcm_substream *substream, } static int snd_pcm_status_user(struct snd_pcm_substream *substream, - struct snd_pcm_status __user * _status) + struct snd_pcm_status __user * _status, + bool ext) { struct snd_pcm_status status; int res; - + memset(&status, 0, sizeof(status)); + /* + * with extension, parameters are read/write, + * get audio_tstamp_data from user, + * ignore rest of status structure + */ + if (ext && get_user(status.audio_tstamp_data, + (u32 __user *)(&_status->audio_tstamp_data))) + return -EFAULT; res = snd_pcm_status(substream, &status); if (res < 0) return res; @@ -2725,7 +2758,9 @@ static int snd_pcm_common_ioctl1(struct file *file, case SNDRV_PCM_IOCTL_SW_PARAMS: return snd_pcm_sw_params_user(substream, arg); case SNDRV_PCM_IOCTL_STATUS: - return snd_pcm_status_user(substream, arg); + return snd_pcm_status_user(substream, arg, false); + case SNDRV_PCM_IOCTL_STATUS_EXT: + return snd_pcm_status_user(substream, arg, true); case SNDRV_PCM_IOCTL_CHANNEL_INFO: return snd_pcm_channel_info_user(substream, arg); case SNDRV_PCM_IOCTL_PREPARE: diff --git a/sound/core/rawmidi.c b/sound/core/rawmidi.c index b5a7485..a775984 100644 --- a/sound/core/rawmidi.c +++ b/sound/core/rawmidi.c @@ -1429,10 +1429,8 @@ static int snd_rawmidi_alloc_substreams(struct snd_rawmidi *rmidi, for (idx = 0; idx < count; idx++) { substream = kzalloc(sizeof(*substream), GFP_KERNEL); - if (substream == NULL) { - rmidi_err(rmidi, "rawmidi: cannot allocate substream\n"); + if (!substream) return -ENOMEM; - } substream->stream = direction; substream->number = idx; substream->rmidi = rmidi; @@ -1479,10 +1477,8 @@ int snd_rawmidi_new(struct snd_card *card, char *id, int device, if (rrawmidi) *rrawmidi = NULL; rmidi = kzalloc(sizeof(*rmidi), GFP_KERNEL); - if (rmidi == NULL) { - dev_err(card->dev, "rawmidi: cannot allocate\n"); + if (!rmidi) return -ENOMEM; - } rmidi->card = card; rmidi->device = device; mutex_init(&rmidi->open_mutex); diff --git a/sound/core/seq/oss/seq_oss.c b/sound/core/seq/oss/seq_oss.c index 16d4267..72873a4 100644 --- a/sound/core/seq/oss/seq_oss.c +++ b/sound/core/seq/oss/seq_oss.c @@ -65,15 +65,20 @@ static unsigned int odev_poll(struct file *file, poll_table * wait); * module interface */ +static struct snd_seq_driver seq_oss_synth_driver = { + .driver = { + .name = KBUILD_MODNAME, + .probe = snd_seq_oss_synth_probe, + .remove = snd_seq_oss_synth_remove, + }, + .id = SNDRV_SEQ_DEV_ID_OSS, + .argsize = sizeof(struct snd_seq_oss_reg), +}; + static int __init alsa_seq_oss_init(void) { int rc; - static struct snd_seq_dev_ops ops = { - snd_seq_oss_synth_register, - snd_seq_oss_synth_unregister, - }; - snd_seq_autoload_lock(); if ((rc = register_device()) < 0) goto error; if ((rc = register_proc()) < 0) { @@ -86,8 +91,8 @@ static int __init alsa_seq_oss_init(void) goto error; } - if ((rc = snd_seq_device_register_driver(SNDRV_SEQ_DEV_ID_OSS, &ops, - sizeof(struct snd_seq_oss_reg))) < 0) { + rc = snd_seq_driver_register(&seq_oss_synth_driver); + if (rc < 0) { snd_seq_oss_delete_client(); unregister_proc(); unregister_device(); @@ -98,13 +103,12 @@ static int __init alsa_seq_oss_init(void) snd_seq_oss_synth_init(); error: - snd_seq_autoload_unlock(); return rc; } static void __exit alsa_seq_oss_exit(void) { - snd_seq_device_unregister_driver(SNDRV_SEQ_DEV_ID_OSS); + snd_seq_driver_unregister(&seq_oss_synth_driver); snd_seq_oss_delete_client(); unregister_proc(); unregister_device(); diff --git a/sound/core/seq/oss/seq_oss_init.c b/sound/core/seq/oss/seq_oss_init.c index b0e32e1..2de3fef 100644 --- a/sound/core/seq/oss/seq_oss_init.c +++ b/sound/core/seq/oss/seq_oss_init.c @@ -188,10 +188,8 @@ snd_seq_oss_open(struct file *file, int level) struct seq_oss_devinfo *dp; dp = kzalloc(sizeof(*dp), GFP_KERNEL); - if (!dp) { - pr_err("ALSA: seq_oss: can't malloc device info\n"); + if (!dp) return -ENOMEM; - } dp->cseq = system_client; dp->port = -1; diff --git a/sound/core/seq/oss/seq_oss_midi.c b/sound/core/seq/oss/seq_oss_midi.c index e79cc44..96e8395 100644 --- a/sound/core/seq/oss/seq_oss_midi.c +++ b/sound/core/seq/oss/seq_oss_midi.c @@ -173,10 +173,9 @@ snd_seq_oss_midi_check_new_port(struct snd_seq_port_info *pinfo) /* * allocate midi info record */ - if ((mdev = kzalloc(sizeof(*mdev), GFP_KERNEL)) == NULL) { - pr_err("ALSA: seq_oss: can't malloc midi info\n"); + mdev = kzalloc(sizeof(*mdev), GFP_KERNEL); + if (!mdev) return -ENOMEM; - } /* copy the port information */ mdev->client = pinfo->addr.client; diff --git a/sound/core/seq/oss/seq_oss_readq.c b/sound/core/seq/oss/seq_oss_readq.c index 654d17a..c080c73 100644 --- a/sound/core/seq/oss/seq_oss_readq.c +++ b/sound/core/seq/oss/seq_oss_readq.c @@ -47,13 +47,12 @@ snd_seq_oss_readq_new(struct seq_oss_devinfo *dp, int maxlen) { struct seq_oss_readq *q; - if ((q = kzalloc(sizeof(*q), GFP_KERNEL)) == NULL) { - pr_err("ALSA: seq_oss: can't malloc read queue\n"); + q = kzalloc(sizeof(*q), GFP_KERNEL); + if (!q) return NULL; - } - if ((q->q = kcalloc(maxlen, sizeof(union evrec), GFP_KERNEL)) == NULL) { - pr_err("ALSA: seq_oss: can't malloc read queue buffer\n"); + q->q = kcalloc(maxlen, sizeof(union evrec), GFP_KERNEL); + if (!q->q) { kfree(q); return NULL; } diff --git a/sound/core/seq/oss/seq_oss_synth.c b/sound/core/seq/oss/seq_oss_synth.c index 701feb7..48e4fe1 100644 --- a/sound/core/seq/oss/seq_oss_synth.c +++ b/sound/core/seq/oss/seq_oss_synth.c @@ -98,17 +98,17 @@ snd_seq_oss_synth_init(void) * registration of the synth device */ int -snd_seq_oss_synth_register(struct snd_seq_device *dev) +snd_seq_oss_synth_probe(struct device *_dev) { + struct snd_seq_device *dev = to_seq_dev(_dev); int i; struct seq_oss_synth *rec; struct snd_seq_oss_reg *reg = SNDRV_SEQ_DEVICE_ARGPTR(dev); unsigned long flags; - if ((rec = kzalloc(sizeof(*rec), GFP_KERNEL)) == NULL) { - pr_err("ALSA: seq_oss: can't malloc synth info\n"); + rec = kzalloc(sizeof(*rec), GFP_KERNEL); + if (!rec) return -ENOMEM; - } rec->seq_device = -1; rec->synth_type = reg->type; rec->synth_subtype = reg->subtype; @@ -149,8 +149,9 @@ snd_seq_oss_synth_register(struct snd_seq_device *dev) int -snd_seq_oss_synth_unregister(struct snd_seq_device *dev) +snd_seq_oss_synth_remove(struct device *_dev) { + struct snd_seq_device *dev = to_seq_dev(_dev); int index; struct seq_oss_synth *rec = dev->driver_data; unsigned long flags; @@ -247,7 +248,6 @@ snd_seq_oss_synth_setup(struct seq_oss_devinfo *dp) if (info->nr_voices > 0) { info->ch = kcalloc(info->nr_voices, sizeof(struct seq_oss_chinfo), GFP_KERNEL); if (!info->ch) { - pr_err("ALSA: seq_oss: Cannot malloc voices\n"); rec->oper.close(&info->arg); module_put(rec->oper.owner); snd_use_lock_free(&rec->use_lock); diff --git a/sound/core/seq/oss/seq_oss_synth.h b/sound/core/seq/oss/seq_oss_synth.h index dbdfcbb..74ac55f 100644 --- a/sound/core/seq/oss/seq_oss_synth.h +++ b/sound/core/seq/oss/seq_oss_synth.h @@ -28,8 +28,8 @@ #include <sound/seq_device.h> void snd_seq_oss_synth_init(void); -int snd_seq_oss_synth_register(struct snd_seq_device *dev); -int snd_seq_oss_synth_unregister(struct snd_seq_device *dev); +int snd_seq_oss_synth_probe(struct device *dev); +int snd_seq_oss_synth_remove(struct device *dev); void snd_seq_oss_synth_setup(struct seq_oss_devinfo *dp); void snd_seq_oss_synth_setup_midi(struct seq_oss_devinfo *dp); void snd_seq_oss_synth_cleanup(struct seq_oss_devinfo *dp); diff --git a/sound/core/seq/seq_device.c b/sound/core/seq/seq_device.c index 0631bda..d99f99d 100644 --- a/sound/core/seq/seq_device.c +++ b/sound/core/seq/seq_device.c @@ -36,6 +36,7 @@ * */ +#include <linux/device.h> #include <linux/init.h> #include <linux/module.h> #include <sound/core.h> @@ -51,140 +52,78 @@ MODULE_AUTHOR("Takashi Iwai <tiwai@suse.de>"); MODULE_DESCRIPTION("ALSA sequencer device management"); MODULE_LICENSE("GPL"); -/* driver state */ -#define DRIVER_EMPTY 0 -#define DRIVER_LOADED (1<<0) -#define DRIVER_REQUESTED (1<<1) -#define DRIVER_LOCKED (1<<2) -#define DRIVER_REQUESTING (1<<3) - -struct ops_list { - char id[ID_LEN]; /* driver id */ - int driver; /* driver state */ - int used; /* reference counter */ - int argsize; /* argument size */ - - /* operators */ - struct snd_seq_dev_ops ops; - - /* registered devices */ - struct list_head dev_list; /* list of devices */ - int num_devices; /* number of associated devices */ - int num_init_devices; /* number of initialized devices */ - struct mutex reg_mutex; - - struct list_head list; /* next driver */ -}; +/* + * bus definition + */ +static int snd_seq_bus_match(struct device *dev, struct device_driver *drv) +{ + struct snd_seq_device *sdev = to_seq_dev(dev); + struct snd_seq_driver *sdrv = to_seq_drv(drv); + return strcmp(sdrv->id, sdev->id) == 0 && + sdrv->argsize == sdev->argsize; +} -static LIST_HEAD(opslist); -static int num_ops; -static DEFINE_MUTEX(ops_mutex); -#ifdef CONFIG_PROC_FS -static struct snd_info_entry *info_entry; -#endif +static struct bus_type snd_seq_bus_type = { + .name = "snd_seq", + .match = snd_seq_bus_match, +}; /* - * prototypes + * proc interface -- just for compatibility */ -static int snd_seq_device_free(struct snd_seq_device *dev); -static int snd_seq_device_dev_free(struct snd_device *device); -static int snd_seq_device_dev_register(struct snd_device *device); -static int snd_seq_device_dev_disconnect(struct snd_device *device); - -static int init_device(struct snd_seq_device *dev, struct ops_list *ops); -static int free_device(struct snd_seq_device *dev, struct ops_list *ops); -static struct ops_list *find_driver(char *id, int create_if_empty); -static struct ops_list *create_driver(char *id); -static void unlock_driver(struct ops_list *ops); -static void remove_drivers(void); +#ifdef CONFIG_PROC_FS +static struct snd_info_entry *info_entry; -/* - * show all drivers and their status - */ +static int print_dev_info(struct device *dev, void *data) +{ + struct snd_seq_device *sdev = to_seq_dev(dev); + struct snd_info_buffer *buffer = data; + + snd_iprintf(buffer, "snd-%s,%s,%d\n", sdev->id, + dev->driver ? "loaded" : "empty", + dev->driver ? 1 : 0); + return 0; +} -#ifdef CONFIG_PROC_FS static void snd_seq_device_info(struct snd_info_entry *entry, struct snd_info_buffer *buffer) { - struct ops_list *ops; - - mutex_lock(&ops_mutex); - list_for_each_entry(ops, &opslist, list) { - snd_iprintf(buffer, "snd-%s%s%s%s,%d\n", - ops->id, - ops->driver & DRIVER_LOADED ? ",loaded" : (ops->driver == DRIVER_EMPTY ? ",empty" : ""), - ops->driver & DRIVER_REQUESTED ? ",requested" : "", - ops->driver & DRIVER_LOCKED ? ",locked" : "", - ops->num_devices); - } - mutex_unlock(&ops_mutex); + bus_for_each_dev(&snd_seq_bus_type, NULL, buffer, print_dev_info); } #endif - + /* * load all registered drivers (called from seq_clientmgr.c) */ #ifdef CONFIG_MODULES -/* avoid auto-loading during module_init() */ +/* flag to block auto-loading */ static atomic_t snd_seq_in_init = ATOMIC_INIT(1); /* blocked as default */ -void snd_seq_autoload_lock(void) -{ - atomic_inc(&snd_seq_in_init); -} -void snd_seq_autoload_unlock(void) +static int request_seq_drv(struct device *dev, void *data) { - atomic_dec(&snd_seq_in_init); + struct snd_seq_device *sdev = to_seq_dev(dev); + + if (!dev->driver) + request_module("snd-%s", sdev->id); + return 0; } -static void autoload_drivers(void) +static void autoload_drivers(struct work_struct *work) { /* avoid reentrance */ - if (atomic_inc_return(&snd_seq_in_init) == 1) { - struct ops_list *ops; - - mutex_lock(&ops_mutex); - list_for_each_entry(ops, &opslist, list) { - if ((ops->driver & DRIVER_REQUESTING) && - !(ops->driver & DRIVER_REQUESTED)) { - ops->used++; - mutex_unlock(&ops_mutex); - ops->driver |= DRIVER_REQUESTED; - request_module("snd-%s", ops->id); - mutex_lock(&ops_mutex); - ops->used--; - } - } - mutex_unlock(&ops_mutex); - } + if (atomic_inc_return(&snd_seq_in_init) == 1) + bus_for_each_dev(&snd_seq_bus_type, NULL, NULL, + request_seq_drv); atomic_dec(&snd_seq_in_init); } -static void call_autoload(struct work_struct *work) -{ - autoload_drivers(); -} - -static DECLARE_WORK(autoload_work, call_autoload); - -static void try_autoload(struct ops_list *ops) -{ - if (!ops->driver) { - ops->driver |= DRIVER_REQUESTING; - schedule_work(&autoload_work); - } -} +static DECLARE_WORK(autoload_work, autoload_drivers); static void queue_autoload_drivers(void) { - struct ops_list *ops; - - mutex_lock(&ops_mutex); - list_for_each_entry(ops, &opslist, list) - try_autoload(ops); - mutex_unlock(&ops_mutex); + schedule_work(&autoload_work); } void snd_seq_autoload_init(void) @@ -195,384 +134,143 @@ void snd_seq_autoload_init(void) queue_autoload_drivers(); #endif } -#else -#define try_autoload(ops) /* NOP */ -#endif +EXPORT_SYMBOL(snd_seq_autoload_init); -void snd_seq_device_load_drivers(void) +void snd_seq_autoload_exit(void) { -#ifdef CONFIG_MODULES - queue_autoload_drivers(); - flush_work(&autoload_work); -#endif + atomic_inc(&snd_seq_in_init); } +EXPORT_SYMBOL(snd_seq_autoload_exit); -/* - * register a sequencer device - * card = card info - * device = device number (if any) - * id = id of driver - * result = return pointer (NULL allowed if unnecessary) - */ -int snd_seq_device_new(struct snd_card *card, int device, char *id, int argsize, - struct snd_seq_device **result) +void snd_seq_device_load_drivers(void) { - struct snd_seq_device *dev; - struct ops_list *ops; - int err; - static struct snd_device_ops dops = { - .dev_free = snd_seq_device_dev_free, - .dev_register = snd_seq_device_dev_register, - .dev_disconnect = snd_seq_device_dev_disconnect, - }; - - if (result) - *result = NULL; - - if (snd_BUG_ON(!id)) - return -EINVAL; - - ops = find_driver(id, 1); - if (ops == NULL) - return -ENOMEM; - - dev = kzalloc(sizeof(*dev)*2 + argsize, GFP_KERNEL); - if (dev == NULL) { - unlock_driver(ops); - return -ENOMEM; - } - - /* set up device info */ - dev->card = card; - dev->device = device; - strlcpy(dev->id, id, sizeof(dev->id)); - dev->argsize = argsize; - dev->status = SNDRV_SEQ_DEVICE_FREE; - - /* add this device to the list */ - mutex_lock(&ops->reg_mutex); - list_add_tail(&dev->list, &ops->dev_list); - ops->num_devices++; - mutex_unlock(&ops->reg_mutex); - - if ((err = snd_device_new(card, SNDRV_DEV_SEQUENCER, dev, &dops)) < 0) { - snd_seq_device_free(dev); - return err; - } - - try_autoload(ops); - unlock_driver(ops); - - if (result) - *result = dev; - - return 0; + queue_autoload_drivers(); + flush_work(&autoload_work); } +EXPORT_SYMBOL(snd_seq_device_load_drivers); +#else +#define queue_autoload_drivers() /* NOP */ +#endif /* - * free the existing device + * device management */ -static int snd_seq_device_free(struct snd_seq_device *dev) -{ - struct ops_list *ops; - - if (snd_BUG_ON(!dev)) - return -EINVAL; - - ops = find_driver(dev->id, 0); - if (ops == NULL) - return -ENXIO; - - /* remove the device from the list */ - mutex_lock(&ops->reg_mutex); - list_del(&dev->list); - ops->num_devices--; - mutex_unlock(&ops->reg_mutex); - - free_device(dev, ops); - if (dev->private_free) - dev->private_free(dev); - kfree(dev); - - unlock_driver(ops); - - return 0; -} - static int snd_seq_device_dev_free(struct snd_device *device) { struct snd_seq_device *dev = device->device_data; - return snd_seq_device_free(dev); + + put_device(&dev->dev); + return 0; } -/* - * register the device - */ static int snd_seq_device_dev_register(struct snd_device *device) { struct snd_seq_device *dev = device->device_data; - struct ops_list *ops; - - ops = find_driver(dev->id, 0); - if (ops == NULL) - return -ENOENT; - - /* initialize this device if the corresponding driver was - * already loaded - */ - if (ops->driver & DRIVER_LOADED) - init_device(dev, ops); + int err; - unlock_driver(ops); + err = device_add(&dev->dev); + if (err < 0) + return err; + if (!dev->dev.driver) + queue_autoload_drivers(); return 0; } -/* - * disconnect the device - */ static int snd_seq_device_dev_disconnect(struct snd_device *device) { struct snd_seq_device *dev = device->device_data; - struct ops_list *ops; - - ops = find_driver(dev->id, 0); - if (ops == NULL) - return -ENOENT; - - free_device(dev, ops); - unlock_driver(ops); + device_del(&dev->dev); return 0; } -/* - * register device driver - * id = driver id - * entry = driver operators - duplicated to each instance - */ -int snd_seq_device_register_driver(char *id, struct snd_seq_dev_ops *entry, - int argsize) +static void snd_seq_dev_release(struct device *dev) { - struct ops_list *ops; - struct snd_seq_device *dev; + struct snd_seq_device *sdev = to_seq_dev(dev); - if (id == NULL || entry == NULL || - entry->init_device == NULL || entry->free_device == NULL) - return -EINVAL; - - ops = find_driver(id, 1); - if (ops == NULL) - return -ENOMEM; - if (ops->driver & DRIVER_LOADED) { - pr_warn("ALSA: seq: driver_register: driver '%s' already exists\n", id); - unlock_driver(ops); - return -EBUSY; - } - - mutex_lock(&ops->reg_mutex); - /* copy driver operators */ - ops->ops = *entry; - ops->driver |= DRIVER_LOADED; - ops->argsize = argsize; - - /* initialize existing devices if necessary */ - list_for_each_entry(dev, &ops->dev_list, list) { - init_device(dev, ops); - } - mutex_unlock(&ops->reg_mutex); - - unlock_driver(ops); - - return 0; + if (sdev->private_free) + sdev->private_free(sdev); + kfree(sdev); } - -/* - * create driver record - */ -static struct ops_list * create_driver(char *id) -{ - struct ops_list *ops; - - ops = kzalloc(sizeof(*ops), GFP_KERNEL); - if (ops == NULL) - return ops; - - /* set up driver entry */ - strlcpy(ops->id, id, sizeof(ops->id)); - mutex_init(&ops->reg_mutex); - /* - * The ->reg_mutex locking rules are per-driver, so we create - * separate per-driver lock classes: - */ - lockdep_set_class(&ops->reg_mutex, (struct lock_class_key *)id); - - ops->driver = DRIVER_EMPTY; - INIT_LIST_HEAD(&ops->dev_list); - /* lock this instance */ - ops->used = 1; - - /* register driver entry */ - mutex_lock(&ops_mutex); - list_add_tail(&ops->list, &opslist); - num_ops++; - mutex_unlock(&ops_mutex); - - return ops; -} - - /* - * unregister the specified driver + * register a sequencer device + * card = card info + * device = device number (if any) + * id = id of driver + * result = return pointer (NULL allowed if unnecessary) */ -int snd_seq_device_unregister_driver(char *id) +int snd_seq_device_new(struct snd_card *card, int device, const char *id, + int argsize, struct snd_seq_device **result) { - struct ops_list *ops; struct snd_seq_device *dev; + int err; + static struct snd_device_ops dops = { + .dev_free = snd_seq_device_dev_free, + .dev_register = snd_seq_device_dev_register, + .dev_disconnect = snd_seq_device_dev_disconnect, + }; - ops = find_driver(id, 0); - if (ops == NULL) - return -ENXIO; - if (! (ops->driver & DRIVER_LOADED) || - (ops->driver & DRIVER_LOCKED)) { - pr_err("ALSA: seq: driver_unregister: cannot unload driver '%s': status=%x\n", - id, ops->driver); - unlock_driver(ops); - return -EBUSY; - } - - /* close and release all devices associated with this driver */ - mutex_lock(&ops->reg_mutex); - ops->driver |= DRIVER_LOCKED; /* do not remove this driver recursively */ - list_for_each_entry(dev, &ops->dev_list, list) { - free_device(dev, ops); - } - - ops->driver = 0; - if (ops->num_init_devices > 0) - pr_err("ALSA: seq: free_driver: init_devices > 0!! (%d)\n", - ops->num_init_devices); - mutex_unlock(&ops->reg_mutex); - - unlock_driver(ops); + if (result) + *result = NULL; - /* remove empty driver entries */ - remove_drivers(); + if (snd_BUG_ON(!id)) + return -EINVAL; - return 0; -} + dev = kzalloc(sizeof(*dev) + argsize, GFP_KERNEL); + if (!dev) + return -ENOMEM; + /* set up device info */ + dev->card = card; + dev->device = device; + dev->id = id; + dev->argsize = argsize; -/* - * remove empty driver entries - */ -static void remove_drivers(void) -{ - struct list_head *head; - - mutex_lock(&ops_mutex); - head = opslist.next; - while (head != &opslist) { - struct ops_list *ops = list_entry(head, struct ops_list, list); - if (! (ops->driver & DRIVER_LOADED) && - ops->used == 0 && ops->num_devices == 0) { - head = head->next; - list_del(&ops->list); - kfree(ops); - num_ops--; - } else - head = head->next; - } - mutex_unlock(&ops_mutex); -} + device_initialize(&dev->dev); + dev->dev.parent = &card->card_dev; + dev->dev.bus = &snd_seq_bus_type; + dev->dev.release = snd_seq_dev_release; + dev_set_name(&dev->dev, "%s-%d-%d", dev->id, card->number, device); -/* - * initialize the device - call init_device operator - */ -static int init_device(struct snd_seq_device *dev, struct ops_list *ops) -{ - if (! (ops->driver & DRIVER_LOADED)) - return 0; /* driver is not loaded yet */ - if (dev->status != SNDRV_SEQ_DEVICE_FREE) - return 0; /* already initialized */ - if (ops->argsize != dev->argsize) { - pr_err("ALSA: seq: incompatible device '%s' for plug-in '%s' (%d %d)\n", - dev->name, ops->id, ops->argsize, dev->argsize); - return -EINVAL; - } - if (ops->ops.init_device(dev) >= 0) { - dev->status = SNDRV_SEQ_DEVICE_REGISTERED; - ops->num_init_devices++; - } else { - pr_err("ALSA: seq: init_device failed: %s: %s\n", - dev->name, dev->id); + /* add this device to the list */ + err = snd_device_new(card, SNDRV_DEV_SEQUENCER, dev, &dops); + if (err < 0) { + put_device(&dev->dev); + return err; } + + if (result) + *result = dev; return 0; } +EXPORT_SYMBOL(snd_seq_device_new); /* - * release the device - call free_device operator + * driver registration */ -static int free_device(struct snd_seq_device *dev, struct ops_list *ops) +int __snd_seq_driver_register(struct snd_seq_driver *drv, struct module *mod) { - int result; - - if (! (ops->driver & DRIVER_LOADED)) - return 0; /* driver is not loaded yet */ - if (dev->status != SNDRV_SEQ_DEVICE_REGISTERED) - return 0; /* not registered */ - if (ops->argsize != dev->argsize) { - pr_err("ALSA: seq: incompatible device '%s' for plug-in '%s' (%d %d)\n", - dev->name, ops->id, ops->argsize, dev->argsize); + if (WARN_ON(!drv->driver.name || !drv->id)) return -EINVAL; - } - if ((result = ops->ops.free_device(dev)) >= 0 || result == -ENXIO) { - dev->status = SNDRV_SEQ_DEVICE_FREE; - dev->driver_data = NULL; - ops->num_init_devices--; - } else { - pr_err("ALSA: seq: free_device failed: %s: %s\n", - dev->name, dev->id); - } - - return 0; + drv->driver.bus = &snd_seq_bus_type; + drv->driver.owner = mod; + return driver_register(&drv->driver); } +EXPORT_SYMBOL_GPL(__snd_seq_driver_register); -/* - * find the matching driver with given id - */ -static struct ops_list * find_driver(char *id, int create_if_empty) +void snd_seq_driver_unregister(struct snd_seq_driver *drv) { - struct ops_list *ops; - - mutex_lock(&ops_mutex); - list_for_each_entry(ops, &opslist, list) { - if (strcmp(ops->id, id) == 0) { - ops->used++; - mutex_unlock(&ops_mutex); - return ops; - } - } - mutex_unlock(&ops_mutex); - if (create_if_empty) - return create_driver(id); - return NULL; + driver_unregister(&drv->driver); } - -static void unlock_driver(struct ops_list *ops) -{ - mutex_lock(&ops_mutex); - ops->used--; - mutex_unlock(&ops_mutex); -} - +EXPORT_SYMBOL_GPL(snd_seq_driver_unregister); /* * module part */ -static int __init alsa_seq_device_init(void) +static int __init seq_dev_proc_init(void) { #ifdef CONFIG_PROC_FS info_entry = snd_info_create_module_entry(THIS_MODULE, "drivers", @@ -589,28 +287,29 @@ static int __init alsa_seq_device_init(void) return 0; } +static int __init alsa_seq_device_init(void) +{ + int err; + + err = bus_register(&snd_seq_bus_type); + if (err < 0) + return err; + err = seq_dev_proc_init(); + if (err < 0) + bus_unregister(&snd_seq_bus_type); + return err; +} + static void __exit alsa_seq_device_exit(void) { #ifdef CONFIG_MODULES cancel_work_sync(&autoload_work); #endif - remove_drivers(); #ifdef CONFIG_PROC_FS snd_info_free_entry(info_entry); #endif - if (num_ops) - pr_err("ALSA: seq: drivers not released (%d)\n", num_ops); + bus_unregister(&snd_seq_bus_type); } -module_init(alsa_seq_device_init) +subsys_initcall(alsa_seq_device_init) module_exit(alsa_seq_device_exit) - -EXPORT_SYMBOL(snd_seq_device_load_drivers); -EXPORT_SYMBOL(snd_seq_device_new); -EXPORT_SYMBOL(snd_seq_device_register_driver); -EXPORT_SYMBOL(snd_seq_device_unregister_driver); -#ifdef CONFIG_MODULES -EXPORT_SYMBOL(snd_seq_autoload_init); -EXPORT_SYMBOL(snd_seq_autoload_lock); -EXPORT_SYMBOL(snd_seq_autoload_unlock); -#endif diff --git a/sound/core/seq/seq_dummy.c b/sound/core/seq/seq_dummy.c index 5d905d9..d3a2ec4 100644 --- a/sound/core/seq/seq_dummy.c +++ b/sound/core/seq/seq_dummy.c @@ -214,11 +214,7 @@ delete_client(void) static int __init alsa_seq_dummy_init(void) { - int err; - snd_seq_autoload_lock(); - err = register_client(); - snd_seq_autoload_unlock(); - return err; + return register_client(); } static void __exit alsa_seq_dummy_exit(void) diff --git a/sound/core/seq/seq_fifo.c b/sound/core/seq/seq_fifo.c index 53a403e..1d5acbe 100644 --- a/sound/core/seq/seq_fifo.c +++ b/sound/core/seq/seq_fifo.c @@ -33,10 +33,8 @@ struct snd_seq_fifo *snd_seq_fifo_new(int poolsize) struct snd_seq_fifo *f; f = kzalloc(sizeof(*f), GFP_KERNEL); - if (f == NULL) { - pr_debug("ALSA: seq: malloc failed for snd_seq_fifo_new() \n"); + if (!f) return NULL; - } f->pool = snd_seq_pool_new(poolsize); if (f->pool == NULL) { diff --git a/sound/core/seq/seq_memory.c b/sound/core/seq/seq_memory.c index ba8e4a6..8010766 100644 --- a/sound/core/seq/seq_memory.c +++ b/sound/core/seq/seq_memory.c @@ -387,10 +387,8 @@ int snd_seq_pool_init(struct snd_seq_pool *pool) return 0; pool->ptr = vmalloc(sizeof(struct snd_seq_event_cell) * pool->size); - if (pool->ptr == NULL) { - pr_debug("ALSA: seq: malloc for sequencer events failed\n"); + if (!pool->ptr) return -ENOMEM; - } /* add new cells to the free cell list */ spin_lock_irqsave(&pool->lock, flags); @@ -463,10 +461,8 @@ struct snd_seq_pool *snd_seq_pool_new(int poolsize) /* create pool block */ pool = kzalloc(sizeof(*pool), GFP_KERNEL); - if (pool == NULL) { - pr_debug("ALSA: seq: malloc failed for pool\n"); + if (!pool) return NULL; - } spin_lock_init(&pool->lock); pool->ptr = NULL; pool->free = NULL; diff --git a/sound/core/seq/seq_midi.c b/sound/core/seq/seq_midi.c index 68fec77..5dd0ee2 100644 --- a/sound/core/seq/seq_midi.c +++ b/sound/core/seq/seq_midi.c @@ -273,8 +273,9 @@ static void snd_seq_midisynth_delete(struct seq_midisynth *msynth) /* register new midi synth port */ static int -snd_seq_midisynth_register_port(struct snd_seq_device *dev) +snd_seq_midisynth_probe(struct device *_dev) { + struct snd_seq_device *dev = to_seq_dev(_dev); struct seq_midisynth_client *client; struct seq_midisynth *msynth, *ms; struct snd_seq_port_info *port; @@ -427,8 +428,9 @@ snd_seq_midisynth_register_port(struct snd_seq_device *dev) /* release midi synth port */ static int -snd_seq_midisynth_unregister_port(struct snd_seq_device *dev) +snd_seq_midisynth_remove(struct device *_dev) { + struct snd_seq_device *dev = to_seq_dev(_dev); struct seq_midisynth_client *client; struct seq_midisynth *msynth; struct snd_card *card = dev->card; @@ -457,24 +459,14 @@ snd_seq_midisynth_unregister_port(struct snd_seq_device *dev) return 0; } +static struct snd_seq_driver seq_midisynth_driver = { + .driver = { + .name = KBUILD_MODNAME, + .probe = snd_seq_midisynth_probe, + .remove = snd_seq_midisynth_remove, + }, + .id = SNDRV_SEQ_DEV_ID_MIDISYNTH, + .argsize = 0, +}; -static int __init alsa_seq_midi_init(void) -{ - static struct snd_seq_dev_ops ops = { - snd_seq_midisynth_register_port, - snd_seq_midisynth_unregister_port, - }; - memset(&synths, 0, sizeof(synths)); - snd_seq_autoload_lock(); - snd_seq_device_register_driver(SNDRV_SEQ_DEV_ID_MIDISYNTH, &ops, 0); - snd_seq_autoload_unlock(); - return 0; -} - -static void __exit alsa_seq_midi_exit(void) -{ - snd_seq_device_unregister_driver(SNDRV_SEQ_DEV_ID_MIDISYNTH); -} - -module_init(alsa_seq_midi_init) -module_exit(alsa_seq_midi_exit) +module_snd_seq_driver(seq_midisynth_driver); diff --git a/sound/core/seq/seq_ports.c b/sound/core/seq/seq_ports.c index 46ff593..55170a2 100644 --- a/sound/core/seq/seq_ports.c +++ b/sound/core/seq/seq_ports.c @@ -141,10 +141,8 @@ struct snd_seq_client_port *snd_seq_create_port(struct snd_seq_client *client, /* create a new port */ new_port = kzalloc(sizeof(*new_port), GFP_KERNEL); - if (! new_port) { - pr_debug("ALSA: seq: malloc failed for registering client port\n"); + if (!new_port) return NULL; /* failure, out of memory */ - } /* init port data */ new_port->addr.client = client->number; new_port->addr.port = -1; diff --git a/sound/core/seq/seq_prioq.c b/sound/core/seq/seq_prioq.c index 021b02b..bc1c848 100644 --- a/sound/core/seq/seq_prioq.c +++ b/sound/core/seq/seq_prioq.c @@ -59,10 +59,8 @@ struct snd_seq_prioq *snd_seq_prioq_new(void) struct snd_seq_prioq *f; f = kzalloc(sizeof(*f), GFP_KERNEL); - if (f == NULL) { - pr_debug("ALSA: seq: malloc failed for snd_seq_prioq_new()\n"); + if (!f) return NULL; - } spin_lock_init(&f->lock); f->head = NULL; diff --git a/sound/core/seq/seq_queue.c b/sound/core/seq/seq_queue.c index aad4878..a0cda38 100644 --- a/sound/core/seq/seq_queue.c +++ b/sound/core/seq/seq_queue.c @@ -111,10 +111,8 @@ static struct snd_seq_queue *queue_new(int owner, int locked) struct snd_seq_queue *q; q = kzalloc(sizeof(*q), GFP_KERNEL); - if (q == NULL) { - pr_debug("ALSA: seq: malloc failed for snd_seq_queue_new()\n"); + if (!q) return NULL; - } spin_lock_init(&q->owner_lock); spin_lock_init(&q->check_lock); diff --git a/sound/core/seq/seq_timer.c b/sound/core/seq/seq_timer.c index e736053..186f161 100644 --- a/sound/core/seq/seq_timer.c +++ b/sound/core/seq/seq_timer.c @@ -56,10 +56,8 @@ struct snd_seq_timer *snd_seq_timer_new(void) struct snd_seq_timer *tmr; tmr = kzalloc(sizeof(*tmr), GFP_KERNEL); - if (tmr == NULL) { - pr_debug("ALSA: seq: malloc failed for snd_seq_timer_new() \n"); + if (!tmr) return NULL; - } spin_lock_init(&tmr->lock); /* reset setup to defaults */ diff --git a/sound/core/sound.c b/sound/core/sound.c index 185cec0..5fc93d0 100644 --- a/sound/core/sound.c +++ b/sound/core/sound.c @@ -186,7 +186,7 @@ static const struct file_operations snd_fops = }; #ifdef CONFIG_SND_DYNAMIC_MINORS -static int snd_find_free_minor(int type) +static int snd_find_free_minor(int type, struct snd_card *card, int dev) { int minor; @@ -209,7 +209,7 @@ static int snd_find_free_minor(int type) return -EBUSY; } #else -static int snd_kernel_minor(int type, struct snd_card *card, int dev) +static int snd_find_free_minor(int type, struct snd_card *card, int dev) { int minor; @@ -237,6 +237,8 @@ static int snd_kernel_minor(int type, struct snd_card *card, int dev) } if (snd_BUG_ON(minor < 0 || minor >= SNDRV_OS_MINORS)) return -EINVAL; + if (snd_minors[minor]) + return -EBUSY; return minor; } #endif @@ -276,13 +278,7 @@ int snd_register_device(int type, struct snd_card *card, int dev, preg->private_data = private_data; preg->card_ptr = card; mutex_lock(&sound_mutex); -#ifdef CONFIG_SND_DYNAMIC_MINORS - minor = snd_find_free_minor(type); -#else - minor = snd_kernel_minor(type, card, dev); - if (minor >= 0 && snd_minors[minor]) - minor = -EBUSY; -#endif + minor = snd_find_free_minor(type, card, dev); if (minor < 0) { err = minor; goto error; diff --git a/sound/core/timer.c b/sound/core/timer.c index 490b489..a9a1a04 100644 --- a/sound/core/timer.c +++ b/sound/core/timer.c @@ -774,10 +774,8 @@ int snd_timer_new(struct snd_card *card, char *id, struct snd_timer_id *tid, if (rtimer) *rtimer = NULL; timer = kzalloc(sizeof(*timer), GFP_KERNEL); - if (timer == NULL) { - pr_err("ALSA: timer: cannot allocate\n"); + if (!timer) return -ENOMEM; - } timer->tmr_class = tid->dev_class; timer->card = card; timer->tmr_device = tid->device; diff --git a/sound/drivers/opl3/opl3_seq.c b/sound/drivers/opl3/opl3_seq.c index a9f618e..fdae5d7 100644 --- a/sound/drivers/opl3/opl3_seq.c +++ b/sound/drivers/opl3/opl3_seq.c @@ -216,8 +216,9 @@ static int snd_opl3_synth_create_port(struct snd_opl3 * opl3) /* ------------------------------ */ -static int snd_opl3_seq_new_device(struct snd_seq_device *dev) +static int snd_opl3_seq_probe(struct device *_dev) { + struct snd_seq_device *dev = to_seq_dev(_dev); struct snd_opl3 *opl3; int client, err; char name[32]; @@ -257,8 +258,9 @@ static int snd_opl3_seq_new_device(struct snd_seq_device *dev) return 0; } -static int snd_opl3_seq_delete_device(struct snd_seq_device *dev) +static int snd_opl3_seq_remove(struct device *_dev) { + struct snd_seq_device *dev = to_seq_dev(_dev); struct snd_opl3 *opl3; opl3 = *(struct snd_opl3 **)SNDRV_SEQ_DEVICE_ARGPTR(dev); @@ -275,22 +277,14 @@ static int snd_opl3_seq_delete_device(struct snd_seq_device *dev) return 0; } -static int __init alsa_opl3_seq_init(void) -{ - static struct snd_seq_dev_ops ops = - { - snd_opl3_seq_new_device, - snd_opl3_seq_delete_device - }; - - return snd_seq_device_register_driver(SNDRV_SEQ_DEV_ID_OPL3, &ops, - sizeof(struct snd_opl3 *)); -} - -static void __exit alsa_opl3_seq_exit(void) -{ - snd_seq_device_unregister_driver(SNDRV_SEQ_DEV_ID_OPL3); -} +static struct snd_seq_driver opl3_seq_driver = { + .driver = { + .name = KBUILD_MODNAME, + .probe = snd_opl3_seq_probe, + .remove = snd_opl3_seq_remove, + }, + .id = SNDRV_SEQ_DEV_ID_OPL3, + .argsize = sizeof(struct snd_opl3 *), +}; -module_init(alsa_opl3_seq_init) -module_exit(alsa_opl3_seq_exit) +module_snd_seq_driver(opl3_seq_driver); diff --git a/sound/drivers/opl4/opl4_seq.c b/sound/drivers/opl4/opl4_seq.c index 9919769..03d6202 100644 --- a/sound/drivers/opl4/opl4_seq.c +++ b/sound/drivers/opl4/opl4_seq.c @@ -124,8 +124,9 @@ static void snd_opl4_seq_free_port(void *private_data) snd_midi_channel_free_set(opl4->chset); } -static int snd_opl4_seq_new_device(struct snd_seq_device *dev) +static int snd_opl4_seq_probe(struct device *_dev) { + struct snd_seq_device *dev = to_seq_dev(_dev); struct snd_opl4 *opl4; int client; struct snd_seq_port_callback pcallbacks; @@ -180,8 +181,9 @@ static int snd_opl4_seq_new_device(struct snd_seq_device *dev) return 0; } -static int snd_opl4_seq_delete_device(struct snd_seq_device *dev) +static int snd_opl4_seq_remove(struct device *_dev) { + struct snd_seq_device *dev = to_seq_dev(_dev); struct snd_opl4 *opl4; opl4 = *(struct snd_opl4 **)SNDRV_SEQ_DEVICE_ARGPTR(dev); @@ -195,21 +197,14 @@ static int snd_opl4_seq_delete_device(struct snd_seq_device *dev) return 0; } -static int __init alsa_opl4_synth_init(void) -{ - static struct snd_seq_dev_ops ops = { - snd_opl4_seq_new_device, - snd_opl4_seq_delete_device - }; - - return snd_seq_device_register_driver(SNDRV_SEQ_DEV_ID_OPL4, &ops, - sizeof(struct snd_opl4 *)); -} - -static void __exit alsa_opl4_synth_exit(void) -{ - snd_seq_device_unregister_driver(SNDRV_SEQ_DEV_ID_OPL4); -} +static struct snd_seq_driver opl4_seq_driver = { + .driver = { + .name = KBUILD_MODNAME, + .probe = snd_opl4_seq_probe, + .remove = snd_opl4_seq_remove, + }, + .id = SNDRV_SEQ_DEV_ID_OPL4, + .argsize = sizeof(struct snd_opl4 *), +}; -module_init(alsa_opl4_synth_init) -module_exit(alsa_opl4_synth_exit) +module_snd_seq_driver(opl4_seq_driver); diff --git a/sound/firewire/amdtp.c b/sound/firewire/amdtp.c index 5cc356d..e061355 100644 --- a/sound/firewire/amdtp.c +++ b/sound/firewire/amdtp.c @@ -166,10 +166,10 @@ int amdtp_stream_add_pcm_hw_constraints(struct amdtp_stream *s, * One AMDTP packet can include some frames. In blocking mode, the * number equals to SYT_INTERVAL. So the number is 8, 16 or 32, * depending on its sampling rate. For accurate period interrupt, it's - * preferrable to aligh period/buffer sizes to current SYT_INTERVAL. + * preferrable to align period/buffer sizes to current SYT_INTERVAL. * - * TODO: These constraints can be improved with propper rules. - * Currently apply LCM of SYT_INTEVALs. + * TODO: These constraints can be improved with proper rules. + * Currently apply LCM of SYT_INTERVALs. */ err = snd_pcm_hw_constraint_step(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_SIZE, 32); @@ -270,7 +270,7 @@ static void amdtp_read_s32(struct amdtp_stream *s, * @s: the AMDTP stream to configure * @format: the format of the ALSA PCM device * - * The sample format must be set after the other paramters (rate/PCM channels/ + * The sample format must be set after the other parameters (rate/PCM channels/ * MIDI) and before the stream is started, and must not be changed while the * stream is running. */ diff --git a/sound/firewire/fireworks/fireworks_transaction.c b/sound/firewire/fireworks/fireworks_transaction.c index 2a85e42..f550808 100644 --- a/sound/firewire/fireworks/fireworks_transaction.c +++ b/sound/firewire/fireworks/fireworks_transaction.c @@ -13,7 +13,7 @@ * * Transaction substance: * At first, 6 data exist. Following to the data, parameters for each command - * exist. All of the parameters are 32 bit alighed to big endian. + * exist. All of the parameters are 32 bit aligned to big endian. * data[0]: Length of transaction substance * data[1]: Transaction version * data[2]: Sequence number. This is incremented by the device diff --git a/sound/isa/sb/emu8000_synth.c b/sound/isa/sb/emu8000_synth.c index 72332df..4aa719c 100644 --- a/sound/isa/sb/emu8000_synth.c +++ b/sound/isa/sb/emu8000_synth.c @@ -34,8 +34,9 @@ MODULE_LICENSE("GPL"); /* * create a new hardware dependent device for Emu8000 */ -static int snd_emu8000_new_device(struct snd_seq_device *dev) +static int snd_emu8000_probe(struct device *_dev) { + struct snd_seq_device *dev = to_seq_dev(_dev); struct snd_emu8000 *hw; struct snd_emux *emu; @@ -93,8 +94,9 @@ static int snd_emu8000_new_device(struct snd_seq_device *dev) /* * free all resources */ -static int snd_emu8000_delete_device(struct snd_seq_device *dev) +static int snd_emu8000_remove(struct device *_dev) { + struct snd_seq_device *dev = to_seq_dev(_dev); struct snd_emu8000 *hw; if (dev->driver_data == NULL) @@ -114,21 +116,14 @@ static int snd_emu8000_delete_device(struct snd_seq_device *dev) * INIT part */ -static int __init alsa_emu8000_init(void) -{ - - static struct snd_seq_dev_ops ops = { - snd_emu8000_new_device, - snd_emu8000_delete_device, - }; - return snd_seq_device_register_driver(SNDRV_SEQ_DEV_ID_EMU8000, &ops, - sizeof(struct snd_emu8000*)); -} - -static void __exit alsa_emu8000_exit(void) -{ - snd_seq_device_unregister_driver(SNDRV_SEQ_DEV_ID_EMU8000); -} - -module_init(alsa_emu8000_init) -module_exit(alsa_emu8000_exit) +static struct snd_seq_driver emu8000_driver = { + .driver = { + .name = KBUILD_MODNAME, + .probe = snd_emu8000_probe, + .remove = snd_emu8000_remove, + }, + .id = SNDRV_SEQ_DEV_ID_EMU8000, + .argsize = sizeof(struct snd_emu8000 *), +}; + +module_snd_seq_driver(emu8000_driver); diff --git a/sound/oss/opl3.c b/sound/oss/opl3.c index 607cee4..b6d19ad 100644 --- a/sound/oss/opl3.c +++ b/sound/oss/opl3.c @@ -666,7 +666,7 @@ static int opl3_start_note (int dev, int voice, int note, int volume) opl3_command(map->ioaddr, FNUM_LOW + map->voice_num, data); data = 0x20 | ((block & 0x7) << 2) | ((fnum >> 8) & 0x3); - devc->voc[voice].keyon_byte = data; + devc->voc[voice].keyon_byte = data; opl3_command(map->ioaddr, KEYON_BLOCK + map->voice_num, data); if (voice_mode == 4) opl3_command(map->ioaddr, KEYON_BLOCK + map->voice_num + 3, data); @@ -717,7 +717,7 @@ static void freq_to_fnum (int freq, int *block, int *fnum) static void opl3_command (int io_addr, unsigned int addr, unsigned int val) { - int i; + int i; /* * The original 2-OP synth requires a quite long delay after writing to a diff --git a/sound/oss/sb_ess.c b/sound/oss/sb_ess.c index b47a690..57f7d25 100644 --- a/sound/oss/sb_ess.c +++ b/sound/oss/sb_ess.c @@ -604,7 +604,7 @@ static void ess_audio_output_block_audio2 ess_chgmixer (devc, 0x78, 0x03, 0x03); /* Go */ devc->irq_mode_16 = IMODE_OUTPUT; - devc->intr_active_16 = 1; + devc->intr_active_16 = 1; } static void ess_audio_output_block @@ -1183,17 +1183,12 @@ FKS_test (devc); chip = "ES1688"; } - printk ( KERN_INFO "ESS chip %s %s%s\n" - , chip - , ( devc->sbmo.esstype == ESSTYPE_DETECT || devc->sbmo.esstype == ESSTYPE_LIKE20 - ? "detected" - : "specified" - ) - , ( devc->sbmo.esstype == ESSTYPE_LIKE20 - ? " (kernel 2.0 compatible)" - : "" - ) - ); + printk(KERN_INFO "ESS chip %s %s%s\n", chip, + (devc->sbmo.esstype == ESSTYPE_DETECT || + devc->sbmo.esstype == ESSTYPE_LIKE20) ? + "detected" : "specified", + devc->sbmo.esstype == ESSTYPE_LIKE20 ? + " (kernel 2.0 compatible)" : ""); sprintf(name,"ESS %s AudioDrive (rev %d)", chip, ess_minor & 0x0f); } else { diff --git a/sound/oss/sb_midi.c b/sound/oss/sb_midi.c index f139028..551ee75 100644 --- a/sound/oss/sb_midi.c +++ b/sound/oss/sb_midi.c @@ -179,14 +179,14 @@ void sb_dsp_midi_init(sb_devc * devc, struct module *owner) { printk(KERN_WARNING "Sound Blaster: failed to allocate MIDI memory.\n"); sound_unload_mididev(dev); - return; + return; } memcpy((char *) midi_devs[dev], (char *) &sb_midi_operations, sizeof(struct midi_operations)); if (owner) - midi_devs[dev]->owner = owner; - + midi_devs[dev]->owner = owner; + midi_devs[dev]->devc = devc; diff --git a/sound/oss/sys_timer.c b/sound/oss/sys_timer.c index 9f03983..2226dda 100644 --- a/sound/oss/sys_timer.c +++ b/sound/oss/sys_timer.c @@ -50,29 +50,24 @@ tmr2ticks(int tmr_value) static void poll_def_tmr(unsigned long dummy) { + if (!opened) + return; + def_tmr.expires = (1) + jiffies; + add_timer(&def_tmr); - if (opened) - { + if (!tmr_running) + return; - { - def_tmr.expires = (1) + jiffies; - add_timer(&def_tmr); - } + spin_lock(&lock); + tmr_ctr++; + curr_ticks = ticks_offs + tmr2ticks(tmr_ctr); - if (tmr_running) - { - spin_lock(&lock); - tmr_ctr++; - curr_ticks = ticks_offs + tmr2ticks(tmr_ctr); - - if (curr_ticks >= next_event_time) - { - next_event_time = (unsigned long) -1; - sequencer_timer(0); - } - spin_unlock(&lock); - } - } + if (curr_ticks >= next_event_time) { + next_event_time = (unsigned long) -1; + sequencer_timer(0); + } + + spin_unlock(&lock); } static void diff --git a/sound/pci/ac97/ac97_codec.c b/sound/pci/ac97/ac97_codec.c index 5ee2f17..5bca1a3 100644 --- a/sound/pci/ac97/ac97_codec.c +++ b/sound/pci/ac97/ac97_codec.c @@ -177,6 +177,7 @@ static const struct ac97_codec_id snd_ac97_codec_ids[] = { { 0x54524123, 0xffffffff, "TR28602", NULL, NULL }, // only guess --jk [TR28023 = eMicro EM28023 (new CT1297)] { 0x54584e03, 0xffffffff, "TLV320AIC27", NULL, NULL }, { 0x54584e20, 0xffffffff, "TLC320AD9xC", NULL, NULL }, +{ 0x56494120, 0xfffffff0, "VIA1613", patch_vt1613, NULL }, { 0x56494161, 0xffffffff, "VIA1612A", NULL, NULL }, // modified ICE1232 with S/PDIF { 0x56494170, 0xffffffff, "VIA1617A", patch_vt1617a, NULL }, // modified VT1616 with S/PDIF { 0x56494182, 0xffffffff, "VIA1618", patch_vt1618, NULL }, diff --git a/sound/pci/ac97/ac97_patch.c b/sound/pci/ac97/ac97_patch.c index ceaac1c..f4234ed 100644 --- a/sound/pci/ac97/ac97_patch.c +++ b/sound/pci/ac97/ac97_patch.c @@ -3352,6 +3352,33 @@ static int patch_cm9780(struct snd_ac97 *ac97) } /* + * VIA VT1613 codec + */ +static const struct snd_kcontrol_new snd_ac97_controls_vt1613[] = { +AC97_SINGLE("DC Offset removal", 0x5a, 10, 1, 0), +}; + +static int patch_vt1613_specific(struct snd_ac97 *ac97) +{ + return patch_build_controls(ac97, &snd_ac97_controls_vt1613[0], + ARRAY_SIZE(snd_ac97_controls_vt1613)); +}; + +static const struct snd_ac97_build_ops patch_vt1613_ops = { + .build_specific = patch_vt1613_specific +}; + +static int patch_vt1613(struct snd_ac97 *ac97) +{ + ac97->build_ops = &patch_vt1613_ops; + + ac97->flags |= AC97_HAS_NO_VIDEO; + ac97->caps |= AC97_BC_HEADPHONE; + + return 0; +} + +/* * VIA VT1616 codec */ static const struct snd_kcontrol_new snd_ac97_controls_vt1616[] = { diff --git a/sound/pci/azt3328.c b/sound/pci/azt3328.c index a40a2b4..33b2a0a 100644 --- a/sound/pci/azt3328.c +++ b/sound/pci/azt3328.c @@ -1385,8 +1385,8 @@ snd_azf3328_ctrl_codec_activity(struct snd_azf3328 *chip, .running) && (!chip->codecs[peer_codecs[codec_type].other2] .running)); - } - if (call_function) + } + if (call_function) snd_azf3328_ctrl_enable_codecs(chip, enable); /* ...and adjust clock, too @@ -2126,7 +2126,8 @@ static struct snd_pcm_ops snd_azf3328_i2s_out_ops = { static int snd_azf3328_pcm(struct snd_azf3328 *chip) { -enum { AZF_PCMDEV_STD, AZF_PCMDEV_I2S_OUT, NUM_AZF_PCMDEVS }; /* pcm devices */ + /* pcm devices */ + enum { AZF_PCMDEV_STD, AZF_PCMDEV_I2S_OUT, NUM_AZF_PCMDEVS }; struct snd_pcm *pcm; int err; diff --git a/sound/pci/cmipci.c b/sound/pci/cmipci.c index 1d0f2ca..6cf464d 100644 --- a/sound/pci/cmipci.c +++ b/sound/pci/cmipci.c @@ -2062,7 +2062,7 @@ static int snd_cmipci_get_volume(struct snd_kcontrol *kcontrol, val = (snd_cmipci_mixer_read(cm, reg.right_reg) >> reg.right_shift) & reg.mask; if (reg.invert) val = reg.mask - val; - ucontrol->value.integer.value[1] = val; + ucontrol->value.integer.value[1] = val; } spin_unlock_irq(&cm->reg_lock); return 0; diff --git a/sound/pci/emu10k1/emu10k1_synth.c b/sound/pci/emu10k1/emu10k1_synth.c index 4c41c90..5457d56 100644 --- a/sound/pci/emu10k1/emu10k1_synth.c +++ b/sound/pci/emu10k1/emu10k1_synth.c @@ -29,8 +29,9 @@ MODULE_LICENSE("GPL"); /* * create a new hardware dependent device for Emu10k1 */ -static int snd_emu10k1_synth_new_device(struct snd_seq_device *dev) +static int snd_emu10k1_synth_probe(struct device *_dev) { + struct snd_seq_device *dev = to_seq_dev(_dev); struct snd_emux *emux; struct snd_emu10k1 *hw; struct snd_emu10k1_synth_arg *arg; @@ -79,8 +80,9 @@ static int snd_emu10k1_synth_new_device(struct snd_seq_device *dev) return 0; } -static int snd_emu10k1_synth_delete_device(struct snd_seq_device *dev) +static int snd_emu10k1_synth_remove(struct device *_dev) { + struct snd_seq_device *dev = to_seq_dev(_dev); struct snd_emux *emux; struct snd_emu10k1 *hw; unsigned long flags; @@ -104,21 +106,14 @@ static int snd_emu10k1_synth_delete_device(struct snd_seq_device *dev) * INIT part */ -static int __init alsa_emu10k1_synth_init(void) -{ - - static struct snd_seq_dev_ops ops = { - snd_emu10k1_synth_new_device, - snd_emu10k1_synth_delete_device, - }; - return snd_seq_device_register_driver(SNDRV_SEQ_DEV_ID_EMU10K1_SYNTH, &ops, - sizeof(struct snd_emu10k1_synth_arg)); -} - -static void __exit alsa_emu10k1_synth_exit(void) -{ - snd_seq_device_unregister_driver(SNDRV_SEQ_DEV_ID_EMU10K1_SYNTH); -} - -module_init(alsa_emu10k1_synth_init) -module_exit(alsa_emu10k1_synth_exit) +static struct snd_seq_driver emu10k1_synth_driver = { + .driver = { + .name = KBUILD_MODNAME, + .probe = snd_emu10k1_synth_probe, + .remove = snd_emu10k1_synth_remove, + }, + .id = SNDRV_SEQ_DEV_ID_EMU10K1_SYNTH, + .argsize = sizeof(struct snd_emu10k1_synth_arg), +}; + +module_snd_seq_driver(emu10k1_synth_driver); diff --git a/sound/pci/hda/Makefile b/sound/pci/hda/Makefile index 194f3093..96caaeb 100644 --- a/sound/pci/hda/Makefile +++ b/sound/pci/hda/Makefile @@ -4,7 +4,7 @@ snd-hda-tegra-objs := hda_tegra.o # for haswell power well snd-hda-intel-$(CONFIG_SND_HDA_I915) += hda_i915.o -snd-hda-codec-y := hda_codec.o hda_jack.o hda_auto_parser.o hda_sysfs.o +snd-hda-codec-y := hda_bind.o hda_codec.o hda_jack.o hda_auto_parser.o hda_sysfs.o snd-hda-codec-$(CONFIG_PROC_FS) += hda_proc.o snd-hda-codec-$(CONFIG_SND_HDA_HWDEP) += hda_hwdep.o snd-hda-codec-$(CONFIG_SND_HDA_INPUT_BEEP) += hda_beep.o diff --git a/sound/pci/hda/hda_beep.c b/sound/pci/hda/hda_beep.c index 1e7de08..4cdac3a 100644 --- a/sound/pci/hda/hda_beep.c +++ b/sound/pci/hda/hda_beep.c @@ -33,30 +33,36 @@ enum { DIGBEEP_HZ_MAX = 12000000, /* 12 KHz */ }; -static void snd_hda_generate_beep(struct work_struct *work) +/* generate or stop tone */ +static void generate_tone(struct hda_beep *beep, int tone) { - struct hda_beep *beep = - container_of(work, struct hda_beep, beep_work); struct hda_codec *codec = beep->codec; - int tone; - if (!beep->enabled) - return; - - tone = beep->tone; if (tone && !beep->playing) { snd_hda_power_up(codec); + if (beep->power_hook) + beep->power_hook(beep, true); beep->playing = 1; } - /* generate tone */ snd_hda_codec_write(codec, beep->nid, 0, AC_VERB_SET_BEEP_CONTROL, tone); if (!tone && beep->playing) { beep->playing = 0; + if (beep->power_hook) + beep->power_hook(beep, false); snd_hda_power_down(codec); } } +static void snd_hda_generate_beep(struct work_struct *work) +{ + struct hda_beep *beep = + container_of(work, struct hda_beep, beep_work); + + if (beep->enabled) + generate_tone(beep, beep->tone); +} + /* (non-standard) Linear beep tone calculation for IDT/STAC codecs * * The tone frequency of beep generator on IDT/STAC codecs is @@ -130,10 +136,7 @@ static void turn_off_beep(struct hda_beep *beep) cancel_work_sync(&beep->beep_work); if (beep->playing) { /* turn off beep */ - snd_hda_codec_write(beep->codec, beep->nid, 0, - AC_VERB_SET_BEEP_CONTROL, 0); - beep->playing = 0; - snd_hda_power_down(beep->codec); + generate_tone(beep, 0); } } @@ -160,6 +163,7 @@ static int snd_hda_do_attach(struct hda_beep *beep) input_dev->name = "HDA Digital PCBeep"; input_dev->phys = beep->phys; input_dev->id.bustype = BUS_PCI; + input_dev->dev.parent = &codec->card->card_dev; input_dev->id.vendor = codec->vendor_id >> 16; input_dev->id.product = codec->vendor_id & 0xffff; @@ -168,7 +172,6 @@ static int snd_hda_do_attach(struct hda_beep *beep) input_dev->evbit[0] = BIT_MASK(EV_SND); input_dev->sndbit[0] = BIT_MASK(SND_BELL) | BIT_MASK(SND_TONE); input_dev->event = snd_hda_beep_event; - input_dev->dev.parent = &codec->dev; input_set_drvdata(input_dev, beep); beep->dev = input_dev; @@ -224,7 +227,7 @@ int snd_hda_attach_beep_device(struct hda_codec *codec, int nid) if (beep == NULL) return -ENOMEM; snprintf(beep->phys, sizeof(beep->phys), - "card%d/codec#%d/beep0", codec->bus->card->number, codec->addr); + "card%d/codec#%d/beep0", codec->card->number, codec->addr); /* enable linear scale */ snd_hda_codec_write_cache(codec, nid, 0, AC_VERB_SET_DIGI_CONVERT_2, 0x01); diff --git a/sound/pci/hda/hda_beep.h b/sound/pci/hda/hda_beep.h index a63b5e0..46524ff 100644 --- a/sound/pci/hda/hda_beep.h +++ b/sound/pci/hda/hda_beep.h @@ -40,6 +40,7 @@ struct hda_beep { unsigned int playing:1; struct work_struct beep_work; /* scheduled task for beep event */ struct mutex mutex; + void (*power_hook)(struct hda_beep *beep, bool on); }; #ifdef CONFIG_SND_HDA_INPUT_BEEP diff --git a/sound/pci/hda/hda_bind.c b/sound/pci/hda/hda_bind.c new file mode 100644 index 0000000..1f40ce3 --- /dev/null +++ b/sound/pci/hda/hda_bind.c @@ -0,0 +1,342 @@ +/* + * HD-audio codec driver binding + * Copyright (c) Takashi Iwai <tiwai@suse.de> + */ + +#include <linux/init.h> +#include <linux/slab.h> +#include <linux/mutex.h> +#include <linux/module.h> +#include <linux/export.h> +#include <linux/pm.h> +#include <linux/pm_runtime.h> +#include <sound/core.h> +#include "hda_codec.h" +#include "hda_local.h" + +/* codec vendor labels */ +struct hda_vendor_id { + unsigned int id; + const char *name; +}; + +static struct hda_vendor_id hda_vendor_ids[] = { + { 0x1002, "ATI" }, + { 0x1013, "Cirrus Logic" }, + { 0x1057, "Motorola" }, + { 0x1095, "Silicon Image" }, + { 0x10de, "Nvidia" }, + { 0x10ec, "Realtek" }, + { 0x1102, "Creative" }, + { 0x1106, "VIA" }, + { 0x111d, "IDT" }, + { 0x11c1, "LSI" }, + { 0x11d4, "Analog Devices" }, + { 0x13f6, "C-Media" }, + { 0x14f1, "Conexant" }, + { 0x17e8, "Chrontel" }, + { 0x1854, "LG" }, + { 0x1aec, "Wolfson Microelectronics" }, + { 0x1af4, "QEMU" }, + { 0x434d, "C-Media" }, + { 0x8086, "Intel" }, + { 0x8384, "SigmaTel" }, + {} /* terminator */ +}; + +/* + * find a matching codec preset + */ +static int hda_bus_match(struct device *dev, struct device_driver *drv) +{ + struct hda_codec *codec = container_of(dev, struct hda_codec, dev); + struct hda_codec_driver *driver = + container_of(drv, struct hda_codec_driver, driver); + const struct hda_codec_preset *preset; + /* check probe_id instead of vendor_id if set */ + u32 id = codec->probe_id ? codec->probe_id : codec->vendor_id; + + for (preset = driver->preset; preset->id; preset++) { + u32 mask = preset->mask; + + if (preset->afg && preset->afg != codec->afg) + continue; + if (preset->mfg && preset->mfg != codec->mfg) + continue; + if (!mask) + mask = ~0; + if (preset->id == (id & mask) && + (!preset->rev || preset->rev == codec->revision_id)) { + codec->preset = preset; + return 1; + } + } + return 0; +} + +/* reset the codec name from the preset */ +static int codec_refresh_name(struct hda_codec *codec, const char *name) +{ + char tmp[16]; + + kfree(codec->chip_name); + if (!name) { + sprintf(tmp, "ID %x", codec->vendor_id & 0xffff); + name = tmp; + } + codec->chip_name = kstrdup(name, GFP_KERNEL); + return codec->chip_name ? 0 : -ENOMEM; +} + +static int hda_codec_driver_probe(struct device *dev) +{ + struct hda_codec *codec = dev_to_hda_codec(dev); + struct module *owner = dev->driver->owner; + int err; + + if (WARN_ON(!codec->preset)) + return -EINVAL; + + err = codec_refresh_name(codec, codec->preset->name); + if (err < 0) + goto error; + + if (!try_module_get(owner)) { + err = -EINVAL; + goto error; + } + + err = codec->preset->patch(codec); + if (err < 0) + goto error_module; + + err = snd_hda_codec_build_pcms(codec); + if (err < 0) + goto error_module; + err = snd_hda_codec_build_controls(codec); + if (err < 0) + goto error_module; + if (codec->card->registered) { + err = snd_card_register(codec->card); + if (err < 0) + goto error_module; + } + + return 0; + + error_module: + module_put(owner); + + error: + snd_hda_codec_cleanup_for_unbind(codec); + return err; +} + +static int hda_codec_driver_remove(struct device *dev) +{ + struct hda_codec *codec = dev_to_hda_codec(dev); + + if (codec->patch_ops.free) + codec->patch_ops.free(codec); + snd_hda_codec_cleanup_for_unbind(codec); + module_put(dev->driver->owner); + return 0; +} + +static void hda_codec_driver_shutdown(struct device *dev) +{ + struct hda_codec *codec = dev_to_hda_codec(dev); + + if (!pm_runtime_suspended(dev) && codec->patch_ops.reboot_notify) + codec->patch_ops.reboot_notify(codec); +} + +int __hda_codec_driver_register(struct hda_codec_driver *drv, const char *name, + struct module *owner) +{ + drv->driver.name = name; + drv->driver.owner = owner; + drv->driver.bus = &snd_hda_bus_type; + drv->driver.probe = hda_codec_driver_probe; + drv->driver.remove = hda_codec_driver_remove; + drv->driver.shutdown = hda_codec_driver_shutdown; + drv->driver.pm = &hda_codec_driver_pm; + return driver_register(&drv->driver); +} +EXPORT_SYMBOL_GPL(__hda_codec_driver_register); + +void hda_codec_driver_unregister(struct hda_codec_driver *drv) +{ + driver_unregister(&drv->driver); +} +EXPORT_SYMBOL_GPL(hda_codec_driver_unregister); + +static inline bool codec_probed(struct hda_codec *codec) +{ + return device_attach(hda_codec_dev(codec)) > 0 && codec->preset; +} + +/* try to auto-load and bind the codec module */ +static void codec_bind_module(struct hda_codec *codec) +{ +#ifdef MODULE + request_module("snd-hda-codec-id:%08x", codec->vendor_id); + if (codec_probed(codec)) + return; + request_module("snd-hda-codec-id:%04x*", + (codec->vendor_id >> 16) & 0xffff); + if (codec_probed(codec)) + return; +#endif +} + +/* store the codec vendor name */ +static int get_codec_vendor_name(struct hda_codec *codec) +{ + const struct hda_vendor_id *c; + const char *vendor = NULL; + u16 vendor_id = codec->vendor_id >> 16; + char tmp[16]; + + for (c = hda_vendor_ids; c->id; c++) { + if (c->id == vendor_id) { + vendor = c->name; + break; + } + } + if (!vendor) { + sprintf(tmp, "Generic %04x", vendor_id); + vendor = tmp; + } + codec->vendor_name = kstrdup(vendor, GFP_KERNEL); + if (!codec->vendor_name) + return -ENOMEM; + return 0; +} + +#if IS_ENABLED(CONFIG_SND_HDA_CODEC_HDMI) +/* if all audio out widgets are digital, let's assume the codec as a HDMI/DP */ +static bool is_likely_hdmi_codec(struct hda_codec *codec) +{ + hda_nid_t nid = codec->start_nid; + int i; + + for (i = 0; i < codec->num_nodes; i++, nid++) { + unsigned int wcaps = get_wcaps(codec, nid); + switch (get_wcaps_type(wcaps)) { + case AC_WID_AUD_IN: + return false; /* HDMI parser supports only HDMI out */ + case AC_WID_AUD_OUT: + if (!(wcaps & AC_WCAP_DIGITAL)) + return false; + break; + } + } + return true; +} +#else +/* no HDMI codec parser support */ +#define is_likely_hdmi_codec(codec) false +#endif /* CONFIG_SND_HDA_CODEC_HDMI */ + +static int codec_bind_generic(struct hda_codec *codec) +{ + if (codec->probe_id) + return -ENODEV; + + if (is_likely_hdmi_codec(codec)) { + codec->probe_id = HDA_CODEC_ID_GENERIC_HDMI; +#if IS_MODULE(CONFIG_SND_HDA_CODEC_HDMI) + request_module("snd-hda-codec-hdmi"); +#endif + if (codec_probed(codec)) + return 0; + } + + codec->probe_id = HDA_CODEC_ID_GENERIC; +#if IS_MODULE(CONFIG_SND_HDA_GENERIC) + request_module("snd-hda-codec-generic"); +#endif + if (codec_probed(codec)) + return 0; + return -ENODEV; +} + +#if IS_ENABLED(CONFIG_SND_HDA_GENERIC) +#define is_generic_config(codec) \ + (codec->modelname && !strcmp(codec->modelname, "generic")) +#else +#define is_generic_config(codec) 0 +#endif + +/** + * snd_hda_codec_configure - (Re-)configure the HD-audio codec + * @codec: the HDA codec + * + * Start parsing of the given codec tree and (re-)initialize the whole + * patch instance. + * + * Returns 0 if successful or a negative error code. + */ +int snd_hda_codec_configure(struct hda_codec *codec) +{ + int err; + + if (!codec->vendor_name) { + err = get_codec_vendor_name(codec); + if (err < 0) + return err; + } + + if (is_generic_config(codec)) + codec->probe_id = HDA_CODEC_ID_GENERIC; + else + codec->probe_id = 0; + + err = device_add(hda_codec_dev(codec)); + if (err < 0) + return err; + + if (!codec->preset) + codec_bind_module(codec); + if (!codec->preset) { + err = codec_bind_generic(codec); + if (err < 0) { + codec_err(codec, "Unable to bind the codec\n"); + goto error; + } + } + + /* audio codec should override the mixer name */ + if (codec->afg || !*codec->card->mixername) + snprintf(codec->card->mixername, + sizeof(codec->card->mixername), + "%s %s", codec->vendor_name, codec->chip_name); + return 0; + + error: + device_del(hda_codec_dev(codec)); + return err; +} +EXPORT_SYMBOL_GPL(snd_hda_codec_configure); + +/* + * bus registration + */ +struct bus_type snd_hda_bus_type = { + .name = "hdaudio", + .match = hda_bus_match, +}; + +static int __init hda_codec_init(void) +{ + return bus_register(&snd_hda_bus_type); +} + +static void __exit hda_codec_exit(void) +{ + bus_unregister(&snd_hda_bus_type); +} + +module_init(hda_codec_init); +module_exit(hda_codec_exit); diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 2fe86d2..7e38d6f 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -26,6 +26,8 @@ #include <linux/mutex.h> #include <linux/module.h> #include <linux/async.h> +#include <linux/pm.h> +#include <linux/pm_runtime.h> #include <sound/core.h> #include "hda_codec.h" #include <sound/asoundef.h> @@ -40,92 +42,13 @@ #define CREATE_TRACE_POINTS #include "hda_trace.h" -/* - * vendor / preset table - */ - -struct hda_vendor_id { - unsigned int id; - const char *name; -}; - -/* codec vendor labels */ -static struct hda_vendor_id hda_vendor_ids[] = { - { 0x1002, "ATI" }, - { 0x1013, "Cirrus Logic" }, - { 0x1057, "Motorola" }, - { 0x1095, "Silicon Image" }, - { 0x10de, "Nvidia" }, - { 0x10ec, "Realtek" }, - { 0x1102, "Creative" }, - { 0x1106, "VIA" }, - { 0x111d, "IDT" }, - { 0x11c1, "LSI" }, - { 0x11d4, "Analog Devices" }, - { 0x13f6, "C-Media" }, - { 0x14f1, "Conexant" }, - { 0x17e8, "Chrontel" }, - { 0x1854, "LG" }, - { 0x1aec, "Wolfson Microelectronics" }, - { 0x1af4, "QEMU" }, - { 0x434d, "C-Media" }, - { 0x8086, "Intel" }, - { 0x8384, "SigmaTel" }, - {} /* terminator */ -}; - -static DEFINE_MUTEX(preset_mutex); -static LIST_HEAD(hda_preset_tables); - -/** - * snd_hda_add_codec_preset - Add a codec preset to the chain - * @preset: codec preset table to add - */ -int snd_hda_add_codec_preset(struct hda_codec_preset_list *preset) -{ - mutex_lock(&preset_mutex); - list_add_tail(&preset->list, &hda_preset_tables); - mutex_unlock(&preset_mutex); - return 0; -} -EXPORT_SYMBOL_GPL(snd_hda_add_codec_preset); - -/** - * snd_hda_delete_codec_preset - Delete a codec preset from the chain - * @preset: codec preset table to delete - */ -int snd_hda_delete_codec_preset(struct hda_codec_preset_list *preset) -{ - mutex_lock(&preset_mutex); - list_del(&preset->list); - mutex_unlock(&preset_mutex); - return 0; -} -EXPORT_SYMBOL_GPL(snd_hda_delete_codec_preset); - #ifdef CONFIG_PM -#define codec_in_pm(codec) ((codec)->in_pm) -static void hda_power_work(struct work_struct *work); -static void hda_keep_power_on(struct hda_codec *codec); -#define hda_codec_is_power_on(codec) ((codec)->power_on) - -static void hda_call_pm_notify(struct hda_codec *codec, bool power_up) -{ - struct hda_bus *bus = codec->bus; - - if ((power_up && codec->pm_up_notified) || - (!power_up && !codec->pm_up_notified)) - return; - if (bus->ops.pm_notify) - bus->ops.pm_notify(bus, power_up); - codec->pm_up_notified = power_up; -} - +#define codec_in_pm(codec) atomic_read(&(codec)->in_pm) +#define hda_codec_is_power_on(codec) \ + (!pm_runtime_suspended(hda_codec_dev(codec))) #else #define codec_in_pm(codec) 0 -static inline void hda_keep_power_on(struct hda_codec *codec) {} #define hda_codec_is_power_on(codec) 1 -#define hda_call_pm_notify(codec, state) {} #endif /** @@ -758,14 +681,11 @@ int snd_hda_queue_unsol_event(struct hda_bus *bus, u32 res, u32 res_ex) struct hda_bus_unsolicited *unsol; unsigned int wp; - if (!bus || !bus->workq) + if (!bus) return 0; trace_hda_unsol_event(bus, res, res_ex); - unsol = bus->unsol; - if (!unsol) - return 0; - + unsol = &bus->unsol; wp = (unsol->wp + 1) % HDA_UNSOL_QUEUE_SIZE; unsol->wp = wp; @@ -773,7 +693,7 @@ int snd_hda_queue_unsol_event(struct hda_bus *bus, u32 res, u32 res_ex) unsol->queue[wp] = res; unsol->queue[wp + 1] = res_ex; - queue_work(bus->workq, &unsol->work); + schedule_work(&unsol->work); return 0; } @@ -784,9 +704,8 @@ EXPORT_SYMBOL_GPL(snd_hda_queue_unsol_event); */ static void process_unsol_events(struct work_struct *work) { - struct hda_bus_unsolicited *unsol = - container_of(work, struct hda_bus_unsolicited, work); - struct hda_bus *bus = unsol->bus; + struct hda_bus *bus = container_of(work, struct hda_bus, unsol.work); + struct hda_bus_unsolicited *unsol = &bus->unsol; struct hda_codec *codec; unsigned int rp, caddr, res; @@ -805,27 +724,6 @@ static void process_unsol_events(struct work_struct *work) } /* - * initialize unsolicited queue - */ -static int init_unsol_queue(struct hda_bus *bus) -{ - struct hda_bus_unsolicited *unsol; - - if (bus->unsol) /* already initialized */ - return 0; - - unsol = kzalloc(sizeof(*unsol), GFP_KERNEL); - if (!unsol) { - dev_err(bus->card->dev, "can't allocate unsolicited queue\n"); - return -ENOMEM; - } - INIT_WORK(&unsol->work, process_unsol_events); - unsol->bus = bus; - bus->unsol = unsol; - return 0; -} - -/* * destructor */ static void snd_hda_bus_free(struct hda_bus *bus) @@ -834,14 +732,9 @@ static void snd_hda_bus_free(struct hda_bus *bus) return; WARN_ON(!list_empty(&bus->codec_list)); - if (bus->workq) - flush_workqueue(bus->workq); - kfree(bus->unsol); + cancel_work_sync(&bus->unsol.work); if (bus->ops.private_free) bus->ops.private_free(bus); - if (bus->workq) - destroy_workqueue(bus->workq); - kfree(bus); } @@ -861,14 +754,12 @@ static int snd_hda_bus_dev_disconnect(struct snd_device *device) /** * snd_hda_bus_new - create a HDA bus * @card: the card entry - * @temp: the template for hda_bus information * @busp: the pointer to store the created bus instance * * Returns 0 if successful, or a negative error code. */ int snd_hda_bus_new(struct snd_card *card, - const struct hda_bus_template *temp, - struct hda_bus **busp) + struct hda_bus **busp) { struct hda_bus *bus; int err; @@ -877,40 +768,18 @@ int snd_hda_bus_new(struct snd_card *card, .dev_free = snd_hda_bus_dev_free, }; - if (snd_BUG_ON(!temp)) - return -EINVAL; - if (snd_BUG_ON(!temp->ops.command || !temp->ops.get_response)) - return -EINVAL; - if (busp) *busp = NULL; bus = kzalloc(sizeof(*bus), GFP_KERNEL); - if (bus == NULL) { - dev_err(card->dev, "can't allocate struct hda_bus\n"); + if (!bus) return -ENOMEM; - } bus->card = card; - bus->private_data = temp->private_data; - bus->pci = temp->pci; - bus->modelname = temp->modelname; - bus->power_save = temp->power_save; - bus->ops = temp->ops; - mutex_init(&bus->cmd_mutex); mutex_init(&bus->prepare_mutex); INIT_LIST_HEAD(&bus->codec_list); - - snprintf(bus->workq_name, sizeof(bus->workq_name), - "hd-audio%d", card->number); - bus->workq = create_singlethread_workqueue(bus->workq_name); - if (!bus->workq) { - dev_err(card->dev, "cannot create workqueue %s\n", - bus->workq_name); - kfree(bus); - return -ENOMEM; - } + INIT_WORK(&bus->unsol.work, process_unsol_events); err = snd_device_new(card, SNDRV_DEV_BUS, bus, &dev_ops); if (err < 0) { @@ -923,111 +792,6 @@ int snd_hda_bus_new(struct snd_card *card, } EXPORT_SYMBOL_GPL(snd_hda_bus_new); -#if IS_ENABLED(CONFIG_SND_HDA_GENERIC) -#define is_generic_config(codec) \ - (codec->modelname && !strcmp(codec->modelname, "generic")) -#else -#define is_generic_config(codec) 0 -#endif - -#ifdef MODULE -#define HDA_MODREQ_MAX_COUNT 2 /* two request_modules()'s */ -#else -#define HDA_MODREQ_MAX_COUNT 0 /* all presets are statically linked */ -#endif - -/* - * find a matching codec preset - */ -static const struct hda_codec_preset * -find_codec_preset(struct hda_codec *codec) -{ - struct hda_codec_preset_list *tbl; - const struct hda_codec_preset *preset; - unsigned int mod_requested = 0; - - again: - mutex_lock(&preset_mutex); - list_for_each_entry(tbl, &hda_preset_tables, list) { - if (!try_module_get(tbl->owner)) { - codec_err(codec, "cannot module_get\n"); - continue; - } - for (preset = tbl->preset; preset->id; preset++) { - u32 mask = preset->mask; - if (preset->afg && preset->afg != codec->afg) - continue; - if (preset->mfg && preset->mfg != codec->mfg) - continue; - if (!mask) - mask = ~0; - if (preset->id == (codec->vendor_id & mask) && - (!preset->rev || - preset->rev == codec->revision_id)) { - mutex_unlock(&preset_mutex); - codec->owner = tbl->owner; - return preset; - } - } - module_put(tbl->owner); - } - mutex_unlock(&preset_mutex); - - if (mod_requested < HDA_MODREQ_MAX_COUNT) { - if (!mod_requested) - request_module("snd-hda-codec-id:%08x", - codec->vendor_id); - else - request_module("snd-hda-codec-id:%04x*", - (codec->vendor_id >> 16) & 0xffff); - mod_requested++; - goto again; - } - return NULL; -} - -/* - * get_codec_name - store the codec name - */ -static int get_codec_name(struct hda_codec *codec) -{ - const struct hda_vendor_id *c; - const char *vendor = NULL; - u16 vendor_id = codec->vendor_id >> 16; - char tmp[16]; - - if (codec->vendor_name) - goto get_chip_name; - - for (c = hda_vendor_ids; c->id; c++) { - if (c->id == vendor_id) { - vendor = c->name; - break; - } - } - if (!vendor) { - sprintf(tmp, "Generic %04x", vendor_id); - vendor = tmp; - } - codec->vendor_name = kstrdup(vendor, GFP_KERNEL); - if (!codec->vendor_name) - return -ENOMEM; - - get_chip_name: - if (codec->chip_name) - return 0; - - if (codec->preset && codec->preset->name) - codec->chip_name = kstrdup(codec->preset->name, GFP_KERNEL); - else { - sprintf(tmp, "ID %x", codec->vendor_id & 0xffff); - codec->chip_name = kstrdup(tmp, GFP_KERNEL); - } - if (!codec->chip_name) - return -ENOMEM; - return 0; -} - /* * look for an AFG and MFG nodes */ @@ -1290,8 +1054,8 @@ static void hda_jackpoll_work(struct work_struct *work) if (!codec->jackpoll_interval) return; - queue_delayed_work(codec->bus->workq, &codec->jackpoll_work, - codec->jackpoll_interval); + schedule_delayed_work(&codec->jackpoll_work, + codec->jackpoll_interval); } static void init_hda_cache(struct hda_cache_rec *cache, @@ -1339,54 +1103,92 @@ get_hda_cvt_setup(struct hda_codec *codec, hda_nid_t nid) } /* - * Dynamic symbol binding for the codec parsers + * PCM device */ +static void release_pcm(struct kref *kref) +{ + struct hda_pcm *pcm = container_of(kref, struct hda_pcm, kref); -#define load_parser(codec, sym) \ - ((codec)->parser = (int (*)(struct hda_codec *))symbol_request(sym)) + if (pcm->pcm) + snd_device_free(pcm->codec->card, pcm->pcm); + clear_bit(pcm->device, pcm->codec->bus->pcm_dev_bits); + kfree(pcm->name); + kfree(pcm); +} -static void unload_parser(struct hda_codec *codec) +void snd_hda_codec_pcm_put(struct hda_pcm *pcm) { - if (codec->parser) - symbol_put_addr(codec->parser); - codec->parser = NULL; + kref_put(&pcm->kref, release_pcm); +} +EXPORT_SYMBOL_GPL(snd_hda_codec_pcm_put); + +struct hda_pcm *snd_hda_codec_pcm_new(struct hda_codec *codec, + const char *fmt, ...) +{ + struct hda_pcm *pcm; + va_list args; + + va_start(args, fmt); + pcm = kzalloc(sizeof(*pcm), GFP_KERNEL); + if (!pcm) + return NULL; + + pcm->codec = codec; + kref_init(&pcm->kref); + pcm->name = kvasprintf(GFP_KERNEL, fmt, args); + if (!pcm->name) { + kfree(pcm); + return NULL; + } + + list_add_tail(&pcm->list, &codec->pcm_list_head); + return pcm; } +EXPORT_SYMBOL_GPL(snd_hda_codec_pcm_new); /* * codec destructor */ -static void snd_hda_codec_free(struct hda_codec *codec) +static void codec_release_pcms(struct hda_codec *codec) +{ + struct hda_pcm *pcm, *n; + + list_for_each_entry_safe(pcm, n, &codec->pcm_list_head, list) { + list_del_init(&pcm->list); + if (pcm->pcm) + snd_device_disconnect(codec->card, pcm->pcm); + snd_hda_codec_pcm_put(pcm); + } +} + +void snd_hda_codec_cleanup_for_unbind(struct hda_codec *codec) { - if (!codec) - return; cancel_delayed_work_sync(&codec->jackpoll_work); + if (!codec->in_freeing) + snd_hda_ctls_clear(codec); + codec_release_pcms(codec); + snd_hda_detach_beep_device(codec); + memset(&codec->patch_ops, 0, sizeof(codec->patch_ops)); snd_hda_jack_tbl_clear(codec); - free_init_pincfgs(codec); -#ifdef CONFIG_PM - cancel_delayed_work(&codec->power_work); - flush_workqueue(codec->bus->workq); -#endif - list_del(&codec->list); - snd_array_free(&codec->mixers); - snd_array_free(&codec->nids); + codec->proc_widget_hook = NULL; + codec->spec = NULL; + + free_hda_cache(&codec->amp_cache); + free_hda_cache(&codec->cmd_cache); + init_hda_cache(&codec->amp_cache, sizeof(struct hda_amp_info)); + init_hda_cache(&codec->cmd_cache, sizeof(struct hda_cache_head)); + + /* free only driver_pins so that init_pins + user_pins are restored */ + snd_array_free(&codec->driver_pins); snd_array_free(&codec->cvt_setups); snd_array_free(&codec->spdif_out); + snd_array_free(&codec->verbs); + codec->preset = NULL; + codec->slave_dig_outs = NULL; + codec->spdif_status_reset = 0; + snd_array_free(&codec->mixers); + snd_array_free(&codec->nids); remove_conn_list(codec); - codec->bus->caddr_tbl[codec->addr] = NULL; - if (codec->patch_ops.free) - codec->patch_ops.free(codec); - hda_call_pm_notify(codec, false); /* cancel leftover refcounts */ - snd_hda_sysfs_clear(codec); - unload_parser(codec); - module_put(codec->owner); - free_hda_cache(&codec->amp_cache); - free_hda_cache(&codec->cmd_cache); - kfree(codec->vendor_name); - kfree(codec->chip_name); - kfree(codec->modelname); - kfree(codec->wcaps); - codec->bus->num_codecs--; - put_device(&codec->dev); } static bool snd_hda_codec_get_supported_ps(struct hda_codec *codec, @@ -1398,11 +1200,12 @@ static unsigned int hda_set_power_state(struct hda_codec *codec, static int snd_hda_codec_dev_register(struct snd_device *device) { struct hda_codec *codec = device->device_data; - int err = device_add(&codec->dev); - if (err < 0) - return err; snd_hda_register_beep_device(codec); + if (device_is_registered(hda_codec_dev(codec))) + pm_runtime_enable(hda_codec_dev(codec)); + /* it was powered up in snd_hda_codec_new(), now all done */ + snd_hda_power_down(codec); return 0; } @@ -1411,20 +1214,37 @@ static int snd_hda_codec_dev_disconnect(struct snd_device *device) struct hda_codec *codec = device->device_data; snd_hda_detach_beep_device(codec); - device_del(&codec->dev); return 0; } static int snd_hda_codec_dev_free(struct snd_device *device) { - snd_hda_codec_free(device->device_data); + struct hda_codec *codec = device->device_data; + + codec->in_freeing = 1; + if (device_is_registered(hda_codec_dev(codec))) + device_del(hda_codec_dev(codec)); + put_device(hda_codec_dev(codec)); return 0; } -/* just free the container */ static void snd_hda_codec_dev_release(struct device *dev) { - kfree(container_of(dev, struct hda_codec, dev)); + struct hda_codec *codec = dev_to_hda_codec(dev); + + free_init_pincfgs(codec); + list_del(&codec->list); + codec->bus->caddr_tbl[codec->addr] = NULL; + clear_bit(codec->addr, &codec->bus->codec_powered); + snd_hda_sysfs_clear(codec); + free_hda_cache(&codec->amp_cache); + free_hda_cache(&codec->cmd_cache); + kfree(codec->vendor_name); + kfree(codec->chip_name); + kfree(codec->modelname); + kfree(codec->wcaps); + codec->bus->num_codecs--; + kfree(codec); } /** @@ -1435,11 +1255,11 @@ static void snd_hda_codec_dev_release(struct device *dev) * * Returns 0 if successful, or a negative error code. */ -int snd_hda_codec_new(struct hda_bus *bus, - unsigned int codec_addr, - struct hda_codec **codecp) +int snd_hda_codec_new(struct hda_bus *bus, struct snd_card *card, + unsigned int codec_addr, struct hda_codec **codecp) { struct hda_codec *codec; + struct device *dev; char component[31]; hda_nid_t fg; int err; @@ -1455,28 +1275,28 @@ int snd_hda_codec_new(struct hda_bus *bus, return -EINVAL; if (bus->caddr_tbl[codec_addr]) { - dev_err(bus->card->dev, + dev_err(card->dev, "address 0x%x is already occupied\n", codec_addr); return -EBUSY; } codec = kzalloc(sizeof(*codec), GFP_KERNEL); - if (codec == NULL) { - dev_err(bus->card->dev, "can't allocate struct hda_codec\n"); + if (!codec) return -ENOMEM; - } - device_initialize(&codec->dev); - codec->dev.parent = &bus->card->card_dev; - codec->dev.class = sound_class; - codec->dev.release = snd_hda_codec_dev_release; - codec->dev.groups = snd_hda_dev_attr_groups; - dev_set_name(&codec->dev, "hdaudioC%dD%d", bus->card->number, - codec_addr); - dev_set_drvdata(&codec->dev, codec); /* for sysfs */ + dev = hda_codec_dev(codec); + device_initialize(dev); + dev->parent = card->dev; + dev->bus = &snd_hda_bus_type; + dev->release = snd_hda_codec_dev_release; + dev->groups = snd_hda_dev_attr_groups; + dev_set_name(dev, "hdaudioC%dD%d", card->number, codec_addr); + dev_set_drvdata(dev, codec); /* for sysfs */ + device_enable_async_suspend(dev); codec->bus = bus; + codec->card = card; codec->addr = codec_addr; mutex_init(&codec->spdif_mutex); mutex_init(&codec->control_mutex); @@ -1492,19 +1312,20 @@ int snd_hda_codec_new(struct hda_bus *bus, snd_array_init(&codec->jacktbl, sizeof(struct hda_jack_tbl), 16); snd_array_init(&codec->verbs, sizeof(struct hda_verb *), 8); INIT_LIST_HEAD(&codec->conn_list); + INIT_LIST_HEAD(&codec->pcm_list_head); INIT_DELAYED_WORK(&codec->jackpoll_work, hda_jackpoll_work); codec->depop_delay = -1; codec->fixup_id = HDA_FIXUP_ID_NOT_SET; #ifdef CONFIG_PM - spin_lock_init(&codec->power_lock); - INIT_DELAYED_WORK(&codec->power_work, hda_power_work); /* snd_hda_codec_new() marks the codec as power-up, and leave it as is. - * the caller has to power down appropriatley after initialization - * phase. + * it's powered down later in snd_hda_codec_dev_register(). */ - hda_keep_power_on(codec); + set_bit(codec->addr, &bus->codec_powered); + pm_runtime_set_active(hda_codec_dev(codec)); + pm_runtime_get_noresume(hda_codec_dev(codec)); + codec->power_jiffies = jiffies; #endif snd_hda_sysfs_init(codec); @@ -1537,17 +1358,15 @@ int snd_hda_codec_new(struct hda_bus *bus, setup_fg_nodes(codec); if (!codec->afg && !codec->mfg) { - dev_err(bus->card->dev, "no AFG or MFG node found\n"); + codec_err(codec, "no AFG or MFG node found\n"); err = -ENODEV; goto error; } fg = codec->afg ? codec->afg : codec->mfg; err = read_widget_caps(codec, fg); - if (err < 0) { - dev_err(bus->card->dev, "cannot malloc\n"); + if (err < 0) goto error; - } err = read_pin_defaults(codec); if (err < 0) goto error; @@ -1564,11 +1383,6 @@ int snd_hda_codec_new(struct hda_bus *bus, #endif codec->epss = snd_hda_codec_get_supported_ps(codec, fg, AC_PWRST_EPSS); -#ifdef CONFIG_PM - if (!codec->d3_stop_clk || !codec->epss) - bus->power_keep_link_on = 1; -#endif - /* power-up all before initialization */ hda_set_power_state(codec, AC_PWRST_D0); @@ -1579,9 +1393,9 @@ int snd_hda_codec_new(struct hda_bus *bus, sprintf(component, "HDA:%08x,%08x,%08x", codec->vendor_id, codec->subsystem_id, codec->revision_id); - snd_component_add(codec->bus->card, component); + snd_component_add(card, component); - err = snd_device_new(bus->card, SNDRV_DEV_CODEC, codec, &dev_ops); + err = snd_device_new(card, SNDRV_DEV_CODEC, codec, &dev_ops); if (err < 0) goto error; @@ -1590,7 +1404,7 @@ int snd_hda_codec_new(struct hda_bus *bus, return 0; error: - snd_hda_codec_free(codec); + put_device(hda_codec_dev(codec)); return err; } EXPORT_SYMBOL_GPL(snd_hda_codec_new); @@ -1613,10 +1427,8 @@ int snd_hda_codec_update_widgets(struct hda_codec *codec) kfree(codec->wcaps); fg = codec->afg ? codec->afg : codec->mfg; err = read_widget_caps(codec, fg); - if (err < 0) { - codec_err(codec, "cannot malloc\n"); + if (err < 0) return err; - } snd_array_free(&codec->init_pins); err = read_pin_defaults(codec); @@ -1625,98 +1437,6 @@ int snd_hda_codec_update_widgets(struct hda_codec *codec) } EXPORT_SYMBOL_GPL(snd_hda_codec_update_widgets); - -#if IS_ENABLED(CONFIG_SND_HDA_CODEC_HDMI) -/* if all audio out widgets are digital, let's assume the codec as a HDMI/DP */ -static bool is_likely_hdmi_codec(struct hda_codec *codec) -{ - hda_nid_t nid = codec->start_nid; - int i; - - for (i = 0; i < codec->num_nodes; i++, nid++) { - unsigned int wcaps = get_wcaps(codec, nid); - switch (get_wcaps_type(wcaps)) { - case AC_WID_AUD_IN: - return false; /* HDMI parser supports only HDMI out */ - case AC_WID_AUD_OUT: - if (!(wcaps & AC_WCAP_DIGITAL)) - return false; - break; - } - } - return true; -} -#else -/* no HDMI codec parser support */ -#define is_likely_hdmi_codec(codec) false -#endif /* CONFIG_SND_HDA_CODEC_HDMI */ - -/** - * snd_hda_codec_configure - (Re-)configure the HD-audio codec - * @codec: the HDA codec - * - * Start parsing of the given codec tree and (re-)initialize the whole - * patch instance. - * - * Returns 0 if successful or a negative error code. - */ -int snd_hda_codec_configure(struct hda_codec *codec) -{ - int (*patch)(struct hda_codec *) = NULL; - int err; - - codec->preset = find_codec_preset(codec); - if (!codec->vendor_name || !codec->chip_name) { - err = get_codec_name(codec); - if (err < 0) - return err; - } - - if (!is_generic_config(codec) && codec->preset) - patch = codec->preset->patch; - if (!patch) { - unload_parser(codec); /* to be sure */ - if (is_likely_hdmi_codec(codec)) { -#if IS_MODULE(CONFIG_SND_HDA_CODEC_HDMI) - patch = load_parser(codec, snd_hda_parse_hdmi_codec); -#elif IS_BUILTIN(CONFIG_SND_HDA_CODEC_HDMI) - patch = snd_hda_parse_hdmi_codec; -#endif - } - if (!patch) { -#if IS_MODULE(CONFIG_SND_HDA_GENERIC) - patch = load_parser(codec, snd_hda_parse_generic_codec); -#elif IS_BUILTIN(CONFIG_SND_HDA_GENERIC) - patch = snd_hda_parse_generic_codec; -#endif - } - if (!patch) { - codec_err(codec, "No codec parser is available\n"); - return -ENODEV; - } - } - - err = patch(codec); - if (err < 0) { - unload_parser(codec); - return err; - } - - if (codec->patch_ops.unsol_event) { - err = init_unsol_queue(codec->bus); - if (err < 0) - return err; - } - - /* audio codec should override the mixer name */ - if (codec->afg || !*codec->bus->card->mixername) - snprintf(codec->bus->card->mixername, - sizeof(codec->bus->card->mixername), - "%s %s", codec->vendor_name, codec->chip_name); - return 0; -} -EXPORT_SYMBOL_GPL(snd_hda_codec_configure); - /* update the stream-id if changed */ static void update_pcm_stream_id(struct hda_codec *codec, struct hda_cvt_setup *p, hda_nid_t nid, @@ -1782,6 +1502,8 @@ void snd_hda_codec_setup_stream(struct hda_codec *codec, hda_nid_t nid, if (!p) return; + if (codec->patch_ops.stream_pm) + codec->patch_ops.stream_pm(codec, nid, true); if (codec->pcm_format_first) update_pcm_format(codec, p, nid, format); update_pcm_stream_id(codec, p, nid, stream_tag, channel_id); @@ -1850,6 +1572,8 @@ static void really_cleanup_stream(struct hda_codec *codec, ); memset(q, 0, sizeof(*q)); q->nid = nid; + if (codec->patch_ops.stream_pm) + codec->patch_ops.stream_pm(codec, nid, false); } /* clean up the all conflicting obsolete streams */ @@ -2192,11 +1916,10 @@ EXPORT_SYMBOL_GPL(snd_hda_codec_amp_read); static int codec_amp_update(struct hda_codec *codec, hda_nid_t nid, int ch, int direction, int idx, int mask, int val, - bool init_only) + bool init_only, bool cache_only) { struct hda_amp_info *info; unsigned int caps; - unsigned int cache_only; if (snd_BUG_ON(mask & ~0xff)) mask &= 0xff; @@ -2214,7 +1937,7 @@ static int codec_amp_update(struct hda_codec *codec, hda_nid_t nid, int ch, return 0; } info->vol[ch] = val; - cache_only = info->head.dirty = codec->cached_write; + info->head.dirty |= cache_only; caps = info->amp_caps; mutex_unlock(&codec->hash_mutex); if (!cache_only) @@ -2238,7 +1961,8 @@ static int codec_amp_update(struct hda_codec *codec, hda_nid_t nid, int ch, int snd_hda_codec_amp_update(struct hda_codec *codec, hda_nid_t nid, int ch, int direction, int idx, int mask, int val) { - return codec_amp_update(codec, nid, ch, direction, idx, mask, val, false); + return codec_amp_update(codec, nid, ch, direction, idx, mask, val, + false, codec->cached_write); } EXPORT_SYMBOL_GPL(snd_hda_codec_amp_update); @@ -2285,7 +2009,8 @@ EXPORT_SYMBOL_GPL(snd_hda_codec_amp_stereo); int snd_hda_codec_amp_init(struct hda_codec *codec, hda_nid_t nid, int ch, int dir, int idx, int mask, int val) { - return codec_amp_update(codec, nid, ch, dir, idx, mask, val, true); + return codec_amp_update(codec, nid, ch, dir, idx, mask, val, true, + codec->cached_write); } EXPORT_SYMBOL_GPL(snd_hda_codec_amp_init); @@ -2427,8 +2152,8 @@ update_amp_value(struct hda_codec *codec, hda_nid_t nid, maxval = get_amp_max_value(codec, nid, dir, 0); if (val > maxval) val = maxval; - return snd_hda_codec_amp_update(codec, nid, ch, dir, idx, - HDA_AMP_VOLMASK, val); + return codec_amp_update(codec, nid, ch, dir, idx, HDA_AMP_VOLMASK, val, + false, !hda_codec_is_power_on(codec)); } /** @@ -2478,14 +2203,12 @@ int snd_hda_mixer_amp_volume_put(struct snd_kcontrol *kcontrol, long *valp = ucontrol->value.integer.value; int change = 0; - snd_hda_power_up(codec); if (chs & 1) { change = update_amp_value(codec, nid, 0, dir, idx, ofs, *valp); valp++; } if (chs & 2) change |= update_amp_value(codec, nid, 1, dir, idx, ofs, *valp); - snd_hda_power_down(codec); return change; } EXPORT_SYMBOL_GPL(snd_hda_mixer_amp_volume_put); @@ -2572,7 +2295,7 @@ find_mixer_ctl(struct hda_codec *codec, const char *name, int dev, int idx) if (snd_BUG_ON(strlen(name) >= sizeof(id.name))) return NULL; strcpy(id.name, name); - return snd_ctl_find_id(codec->bus->card, &id); + return snd_ctl_find_id(codec->card, &id); } /** @@ -2636,7 +2359,7 @@ int snd_hda_ctl_add(struct hda_codec *codec, hda_nid_t nid, nid = kctl->id.subdevice & 0xffff; if (kctl->id.subdevice & (HDA_SUBDEV_NID_FLAG|HDA_SUBDEV_AMP_FLAG)) kctl->id.subdevice = 0; - err = snd_ctl_add(codec->bus->card, kctl); + err = snd_ctl_add(codec->card, kctl); if (err < 0) return err; item = snd_array_new(&codec->mixers); @@ -2689,7 +2412,7 @@ void snd_hda_ctls_clear(struct hda_codec *codec) int i; struct hda_nid_item *items = codec->mixers.list; for (i = 0; i < codec->mixers.used; i++) - snd_ctl_remove(codec->bus->card, items[i].kctl); + snd_ctl_remove(codec->card, items[i].kctl); snd_array_free(&codec->mixers); snd_array_free(&codec->nids); } @@ -2713,9 +2436,8 @@ int snd_hda_lock_devices(struct hda_bus *bus) goto err_clear; list_for_each_entry(codec, &bus->codec_list, list) { - int pcm; - for (pcm = 0; pcm < codec->num_pcms; pcm++) { - struct hda_pcm *cpcm = &codec->pcm_info[pcm]; + struct hda_pcm *cpcm; + list_for_each_entry(cpcm, &codec->pcm_list_head, list) { if (!cpcm->pcm) continue; if (cpcm->pcm->streams[0].substream_opened || @@ -2742,7 +2464,6 @@ void snd_hda_unlock_devices(struct hda_bus *bus) { struct snd_card *card = bus->card; - card = bus->card; spin_lock(&card->files_lock); card->shutdown = 0; spin_unlock(&card->files_lock); @@ -2762,51 +2483,13 @@ EXPORT_SYMBOL_GPL(snd_hda_unlock_devices); int snd_hda_codec_reset(struct hda_codec *codec) { struct hda_bus *bus = codec->bus; - struct snd_card *card = bus->card; - int i; if (snd_hda_lock_devices(bus) < 0) return -EBUSY; /* OK, let it free */ - cancel_delayed_work_sync(&codec->jackpoll_work); -#ifdef CONFIG_PM - cancel_delayed_work_sync(&codec->power_work); - flush_workqueue(bus->workq); -#endif - snd_hda_ctls_clear(codec); - /* release PCMs */ - for (i = 0; i < codec->num_pcms; i++) { - if (codec->pcm_info[i].pcm) { - snd_device_free(card, codec->pcm_info[i].pcm); - clear_bit(codec->pcm_info[i].device, - bus->pcm_dev_bits); - } - } - snd_hda_detach_beep_device(codec); - if (codec->patch_ops.free) - codec->patch_ops.free(codec); - memset(&codec->patch_ops, 0, sizeof(codec->patch_ops)); - snd_hda_jack_tbl_clear(codec); - codec->proc_widget_hook = NULL; - codec->spec = NULL; - free_hda_cache(&codec->amp_cache); - free_hda_cache(&codec->cmd_cache); - init_hda_cache(&codec->amp_cache, sizeof(struct hda_amp_info)); - init_hda_cache(&codec->cmd_cache, sizeof(struct hda_cache_head)); - /* free only driver_pins so that init_pins + user_pins are restored */ - snd_array_free(&codec->driver_pins); - snd_array_free(&codec->cvt_setups); - snd_array_free(&codec->spdif_out); - snd_array_free(&codec->verbs); - codec->num_pcms = 0; - codec->pcm_info = NULL; - codec->preset = NULL; - codec->slave_dig_outs = NULL; - codec->spdif_status_reset = 0; - unload_parser(codec); - module_put(codec->owner); - codec->owner = NULL; + if (device_is_registered(hda_codec_dev(codec))) + device_del(hda_codec_dev(codec)); /* allow device access again */ snd_hda_unlock_devices(bus); @@ -3153,19 +2836,19 @@ int snd_hda_mixer_amp_switch_put(struct snd_kcontrol *kcontrol, long *valp = ucontrol->value.integer.value; int change = 0; - snd_hda_power_up(codec); if (chs & 1) { - change = snd_hda_codec_amp_update(codec, nid, 0, dir, idx, - HDA_AMP_MUTE, - *valp ? 0 : HDA_AMP_MUTE); + change = codec_amp_update(codec, nid, 0, dir, idx, + HDA_AMP_MUTE, + *valp ? 0 : HDA_AMP_MUTE, false, + !hda_codec_is_power_on(codec)); valp++; } if (chs & 2) - change |= snd_hda_codec_amp_update(codec, nid, 1, dir, idx, - HDA_AMP_MUTE, - *valp ? 0 : HDA_AMP_MUTE); + change |= codec_amp_update(codec, nid, 1, dir, idx, + HDA_AMP_MUTE, + *valp ? 0 : HDA_AMP_MUTE, false, + !hda_codec_is_power_on(codec)); hda_call_check_power_status(codec, nid); - snd_hda_power_down(codec); return change; } EXPORT_SYMBOL_GPL(snd_hda_mixer_amp_switch_put); @@ -4212,31 +3895,40 @@ static inline void hda_exec_init_verbs(struct hda_codec *codec) {} #endif #ifdef CONFIG_PM +/* update the power on/off account with the current jiffies */ +static void update_power_acct(struct hda_codec *codec, bool on) +{ + unsigned long delta = jiffies - codec->power_jiffies; + + if (on) + codec->power_on_acct += delta; + else + codec->power_off_acct += delta; + codec->power_jiffies += delta; +} + +void snd_hda_update_power_acct(struct hda_codec *codec) +{ + update_power_acct(codec, hda_codec_is_power_on(codec)); +} + /* * call suspend and power-down; used both from PM and power-save * this function returns the power state in the end */ -static unsigned int hda_call_codec_suspend(struct hda_codec *codec, bool in_wq) +static unsigned int hda_call_codec_suspend(struct hda_codec *codec) { unsigned int state; - codec->in_pm = 1; + atomic_inc(&codec->in_pm); if (codec->patch_ops.suspend) codec->patch_ops.suspend(codec); hda_cleanup_all_streams(codec); state = hda_set_power_state(codec, AC_PWRST_D3); - /* Cancel delayed work if we aren't currently running from it. */ - if (!in_wq) - cancel_delayed_work_sync(&codec->power_work); - spin_lock(&codec->power_lock); - snd_hda_update_power_acct(codec); trace_hda_power_down(codec); - codec->power_on = 0; - codec->power_transition = 0; - codec->power_jiffies = jiffies; - spin_unlock(&codec->power_lock); - codec->in_pm = 0; + update_power_acct(codec, true); + atomic_dec(&codec->in_pm); return state; } @@ -4261,14 +3953,13 @@ static void hda_mark_cmd_cache_dirty(struct hda_codec *codec) */ static void hda_call_codec_resume(struct hda_codec *codec) { - codec->in_pm = 1; + atomic_inc(&codec->in_pm); + trace_hda_power_up(codec); hda_mark_cmd_cache_dirty(codec); - /* set as if powered on for avoiding re-entering the resume - * in the resume / power-save sequence - */ - hda_keep_power_on(codec); + codec->power_jiffies = jiffies; + hda_set_power_state(codec, AC_PWRST_D0); restore_shutup_pins(codec); hda_exec_init_verbs(codec); @@ -4286,64 +3977,63 @@ static void hda_call_codec_resume(struct hda_codec *codec) hda_jackpoll_work(&codec->jackpoll_work.work); else snd_hda_jack_report_sync(codec); - - codec->in_pm = 0; - snd_hda_power_down(codec); /* flag down before returning */ + atomic_dec(&codec->in_pm); } -#endif /* CONFIG_PM */ +static int hda_codec_runtime_suspend(struct device *dev) +{ + struct hda_codec *codec = dev_to_hda_codec(dev); + struct hda_pcm *pcm; + unsigned int state; -/** - * snd_hda_build_controls - build mixer controls - * @bus: the BUS - * - * Creates mixer controls for each codec included in the bus. - * - * Returns 0 if successful, otherwise a negative error code. - */ -int snd_hda_build_controls(struct hda_bus *bus) + cancel_delayed_work_sync(&codec->jackpoll_work); + list_for_each_entry(pcm, &codec->pcm_list_head, list) + snd_pcm_suspend_all(pcm->pcm); + state = hda_call_codec_suspend(codec); + if (codec->d3_stop_clk && codec->epss && (state & AC_PWRST_CLK_STOP_OK)) + clear_bit(codec->addr, &codec->bus->codec_powered); + return 0; +} + +static int hda_codec_runtime_resume(struct device *dev) { - struct hda_codec *codec; + struct hda_codec *codec = dev_to_hda_codec(dev); - list_for_each_entry(codec, &bus->codec_list, list) { - int err = snd_hda_codec_build_controls(codec); - if (err < 0) { - codec_err(codec, - "cannot build controls for #%d (error %d)\n", - codec->addr, err); - err = snd_hda_codec_reset(codec); - if (err < 0) { - codec_err(codec, - "cannot revert codec\n"); - return err; - } - } - } + set_bit(codec->addr, &codec->bus->codec_powered); + hda_call_codec_resume(codec); + pm_runtime_mark_last_busy(dev); return 0; } -EXPORT_SYMBOL_GPL(snd_hda_build_controls); +#endif /* CONFIG_PM */ + +/* referred in hda_bind.c */ +const struct dev_pm_ops hda_codec_driver_pm = { + SET_SYSTEM_SLEEP_PM_OPS(pm_runtime_force_suspend, + pm_runtime_force_resume) + SET_RUNTIME_PM_OPS(hda_codec_runtime_suspend, hda_codec_runtime_resume, + NULL) +}; /* * add standard channel maps if not specified */ static int add_std_chmaps(struct hda_codec *codec) { - int i, str, err; + struct hda_pcm *pcm; + int str, err; - for (i = 0; i < codec->num_pcms; i++) { + list_for_each_entry(pcm, &codec->pcm_list_head, list) { for (str = 0; str < 2; str++) { - struct snd_pcm *pcm = codec->pcm_info[i].pcm; - struct hda_pcm_stream *hinfo = - &codec->pcm_info[i].stream[str]; + struct hda_pcm_stream *hinfo = &pcm->stream[str]; struct snd_pcm_chmap *chmap; const struct snd_pcm_chmap_elem *elem; - if (codec->pcm_info[i].own_chmap) + if (pcm->own_chmap) continue; if (!pcm || !hinfo->substreams) continue; elem = hinfo->chmap ? hinfo->chmap : snd_pcm_std_chmaps; - err = snd_pcm_add_chmap_ctls(pcm, str, elem, + err = snd_pcm_add_chmap_ctls(pcm->pcm, str, elem, hinfo->channels_max, 0, &chmap); if (err < 0) @@ -4792,7 +4482,11 @@ int snd_hda_codec_prepare(struct hda_codec *codec, { int ret; mutex_lock(&codec->bus->prepare_mutex); - ret = hinfo->ops.prepare(hinfo, codec, stream, format, substream); + if (hinfo->ops.prepare) + ret = hinfo->ops.prepare(hinfo, codec, stream, format, + substream); + else + ret = -ENODEV; if (ret >= 0) purify_inactive_streams(codec); mutex_unlock(&codec->bus->prepare_mutex); @@ -4813,7 +4507,8 @@ void snd_hda_codec_cleanup(struct hda_codec *codec, struct snd_pcm_substream *substream) { mutex_lock(&codec->bus->prepare_mutex); - hinfo->ops.cleanup(hinfo, codec, substream); + if (hinfo->ops.cleanup) + hinfo->ops.cleanup(hinfo, codec, substream); mutex_unlock(&codec->bus->prepare_mutex); } EXPORT_SYMBOL_GPL(snd_hda_codec_cleanup); @@ -4871,112 +4566,84 @@ static int get_empty_pcm_device(struct hda_bus *bus, unsigned int type) return -EAGAIN; } -/* - * attach a new PCM stream - */ -static int snd_hda_attach_pcm(struct hda_codec *codec, struct hda_pcm *pcm) +/* call build_pcms ops of the given codec and set up the default parameters */ +int snd_hda_codec_parse_pcms(struct hda_codec *codec) { - struct hda_bus *bus = codec->bus; - struct hda_pcm_stream *info; - int stream, err; + struct hda_pcm *cpcm; + int err; - if (snd_BUG_ON(!pcm->name)) - return -EINVAL; - for (stream = 0; stream < 2; stream++) { - info = &pcm->stream[stream]; - if (info->substreams) { + if (!list_empty(&codec->pcm_list_head)) + return 0; /* already parsed */ + + if (!codec->patch_ops.build_pcms) + return 0; + + err = codec->patch_ops.build_pcms(codec); + if (err < 0) { + codec_err(codec, "cannot build PCMs for #%d (error %d)\n", + codec->addr, err); + return err; + } + + list_for_each_entry(cpcm, &codec->pcm_list_head, list) { + int stream; + + for (stream = 0; stream < 2; stream++) { + struct hda_pcm_stream *info = &cpcm->stream[stream]; + + if (!info->substreams) + continue; err = set_pcm_default_values(codec, info); - if (err < 0) + if (err < 0) { + codec_warn(codec, + "fail to setup default for PCM %s\n", + cpcm->name); return err; + } } } - return bus->ops.attach_pcm(bus, codec, pcm); + + return 0; } /* assign all PCMs of the given codec */ int snd_hda_codec_build_pcms(struct hda_codec *codec) { - unsigned int pcm; - int err; + struct hda_bus *bus = codec->bus; + struct hda_pcm *cpcm; + int dev, err; - if (!codec->num_pcms) { - if (!codec->patch_ops.build_pcms) - return 0; - err = codec->patch_ops.build_pcms(codec); - if (err < 0) { - codec_err(codec, - "cannot build PCMs for #%d (error %d)\n", - codec->addr, err); - err = snd_hda_codec_reset(codec); - if (err < 0) { - codec_err(codec, - "cannot revert codec\n"); - return err; - } - } + if (snd_BUG_ON(!bus->ops.attach_pcm)) + return -EINVAL; + + err = snd_hda_codec_parse_pcms(codec); + if (err < 0) { + snd_hda_codec_reset(codec); + return err; } - for (pcm = 0; pcm < codec->num_pcms; pcm++) { - struct hda_pcm *cpcm = &codec->pcm_info[pcm]; - int dev; + /* attach a new PCM streams */ + list_for_each_entry(cpcm, &codec->pcm_list_head, list) { + if (cpcm->pcm) + continue; /* already attached */ if (!cpcm->stream[0].substreams && !cpcm->stream[1].substreams) continue; /* no substreams assigned */ - if (!cpcm->pcm) { - dev = get_empty_pcm_device(codec->bus, cpcm->pcm_type); - if (dev < 0) - continue; /* no fatal error */ - cpcm->device = dev; - err = snd_hda_attach_pcm(codec, cpcm); - if (err < 0) { - codec_err(codec, - "cannot attach PCM stream %d for codec #%d\n", - dev, codec->addr); - continue; /* no fatal error */ - } + dev = get_empty_pcm_device(bus, cpcm->pcm_type); + if (dev < 0) + continue; /* no fatal error */ + cpcm->device = dev; + err = bus->ops.attach_pcm(bus, codec, cpcm); + if (err < 0) { + codec_err(codec, + "cannot attach PCM stream %d for codec #%d\n", + dev, codec->addr); + continue; /* no fatal error */ } } - return 0; -} -/** - * snd_hda_build_pcms - build PCM information - * @bus: the BUS - * - * Create PCM information for each codec included in the bus. - * - * The build_pcms codec patch is requested to set up codec->num_pcms and - * codec->pcm_info properly. The array is referred by the top-level driver - * to create its PCM instances. - * The allocated codec->pcm_info should be released in codec->patch_ops.free - * callback. - * - * At least, substreams, channels_min and channels_max must be filled for - * each stream. substreams = 0 indicates that the stream doesn't exist. - * When rates and/or formats are zero, the supported values are queried - * from the given nid. The nid is used also by the default ops.prepare - * and ops.cleanup callbacks. - * - * The driver needs to call ops.open in its open callback. Similarly, - * ops.close is supposed to be called in the close callback. - * ops.prepare should be called in the prepare or hw_params callback - * with the proper parameters for set up. - * ops.cleanup should be called in hw_free for clean up of streams. - * - * This function returns 0 if successful, or a negative error code. - */ -int snd_hda_build_pcms(struct hda_bus *bus) -{ - struct hda_codec *codec; - - list_for_each_entry(codec, &bus->codec_list, list) { - int err = snd_hda_codec_build_pcms(codec); - if (err < 0) - return err; - } return 0; } -EXPORT_SYMBOL_GPL(snd_hda_build_pcms); /** * snd_hda_add_new_ctls - create controls from the array @@ -5029,127 +4696,70 @@ int snd_hda_add_new_ctls(struct hda_codec *codec, EXPORT_SYMBOL_GPL(snd_hda_add_new_ctls); #ifdef CONFIG_PM -static void hda_power_work(struct work_struct *work) +/** + * snd_hda_power_up - Power-up the codec + * @codec: HD-audio codec + * + * Increment the usage counter and resume the device if not done yet. + */ +void snd_hda_power_up(struct hda_codec *codec) { - struct hda_codec *codec = - container_of(work, struct hda_codec, power_work.work); - struct hda_bus *bus = codec->bus; - unsigned int state; + struct device *dev = hda_codec_dev(codec); - spin_lock(&codec->power_lock); - if (codec->power_transition > 0) { /* during power-up sequence? */ - spin_unlock(&codec->power_lock); - return; - } - if (!codec->power_on || codec->power_count) { - codec->power_transition = 0; - spin_unlock(&codec->power_lock); + if (codec_in_pm(codec)) return; - } - spin_unlock(&codec->power_lock); - - state = hda_call_codec_suspend(codec, true); - if (!bus->power_keep_link_on && (state & AC_PWRST_CLK_STOP_OK)) - hda_call_pm_notify(codec, false); + pm_runtime_get_sync(dev); } +EXPORT_SYMBOL_GPL(snd_hda_power_up); -static void hda_keep_power_on(struct hda_codec *codec) -{ - spin_lock(&codec->power_lock); - codec->power_count++; - codec->power_on = 1; - codec->power_jiffies = jiffies; - spin_unlock(&codec->power_lock); - hda_call_pm_notify(codec, true); -} - -/* update the power on/off account with the current jiffies */ -void snd_hda_update_power_acct(struct hda_codec *codec) -{ - unsigned long delta = jiffies - codec->power_jiffies; - if (codec->power_on) - codec->power_on_acct += delta; - else - codec->power_off_acct += delta; - codec->power_jiffies += delta; -} - -/* Transition to powered up, if wait_power_down then wait for a pending - * transition to D3 to complete. A pending D3 transition is indicated - * with power_transition == -1. */ -/* call this with codec->power_lock held! */ -static void __snd_hda_power_up(struct hda_codec *codec, bool wait_power_down) +/** + * snd_hda_power_down - Power-down the codec + * @codec: HD-audio codec + * + * Decrement the usage counter and schedules the autosuspend if none used. + */ +void snd_hda_power_down(struct hda_codec *codec) { - /* Return if power_on or transitioning to power_on, unless currently - * powering down. */ - if ((codec->power_on || codec->power_transition > 0) && - !(wait_power_down && codec->power_transition < 0)) - return; - spin_unlock(&codec->power_lock); - - cancel_delayed_work_sync(&codec->power_work); + struct device *dev = hda_codec_dev(codec); - spin_lock(&codec->power_lock); - /* If the power down delayed work was cancelled above before starting, - * then there is no need to go through power up here. - */ - if (codec->power_on) { - if (codec->power_transition < 0) - codec->power_transition = 0; + if (codec_in_pm(codec)) return; - } - - trace_hda_power_up(codec); - snd_hda_update_power_acct(codec); - codec->power_on = 1; - codec->power_jiffies = jiffies; - codec->power_transition = 1; /* avoid reentrance */ - spin_unlock(&codec->power_lock); - - hda_call_codec_resume(codec); - - spin_lock(&codec->power_lock); - codec->power_transition = 0; + pm_runtime_mark_last_busy(dev); + pm_runtime_put_autosuspend(dev); } +EXPORT_SYMBOL_GPL(snd_hda_power_down); -#define power_save(codec) \ - ((codec)->bus->power_save ? *(codec)->bus->power_save : 0) - -/* Transition to powered down */ -static void __snd_hda_power_down(struct hda_codec *codec) +static void codec_set_power_save(struct hda_codec *codec, int delay) { - if (!codec->power_on || codec->power_count || codec->power_transition) - return; + struct device *dev = hda_codec_dev(codec); - if (power_save(codec)) { - codec->power_transition = -1; /* avoid reentrance */ - queue_delayed_work(codec->bus->workq, &codec->power_work, - msecs_to_jiffies(power_save(codec) * 1000)); + if (delay > 0) { + pm_runtime_set_autosuspend_delay(dev, delay); + pm_runtime_use_autosuspend(dev); + pm_runtime_allow(dev); + if (!pm_runtime_suspended(dev)) + pm_runtime_mark_last_busy(dev); + } else { + pm_runtime_dont_use_autosuspend(dev); + pm_runtime_forbid(dev); } } /** - * snd_hda_power_save - Power-up/down/sync the codec - * @codec: HD-audio codec - * @delta: the counter delta to change - * @d3wait: sync for D3 transition complete + * snd_hda_set_power_save - reprogram autosuspend for the given delay + * @bus: HD-audio bus + * @delay: autosuspend delay in msec, 0 = off * - * Change the power-up counter via @delta, and power up or down the hardware - * appropriately. For the power-down, queue to the delayed action. - * Passing zero to @delta means to synchronize the power state. + * Synchronize the runtime PM autosuspend state from the power_save option. */ -void snd_hda_power_save(struct hda_codec *codec, int delta, bool d3wait) +void snd_hda_set_power_save(struct hda_bus *bus, int delay) { - spin_lock(&codec->power_lock); - codec->power_count += delta; - trace_hda_power_count(codec); - if (delta > 0) - __snd_hda_power_up(codec, d3wait); - else - __snd_hda_power_down(codec); - spin_unlock(&codec->power_lock); + struct hda_codec *c; + + list_for_each_entry(c, &bus->codec_list, list) + codec_set_power_save(c, delay); } -EXPORT_SYMBOL_GPL(snd_hda_power_save); +EXPORT_SYMBOL_GPL(snd_hda_set_power_save); /** * snd_hda_check_amp_list_power - Check the amp list and update the power @@ -5203,88 +4813,6 @@ EXPORT_SYMBOL_GPL(snd_hda_check_amp_list_power); #endif /* - * Channel mode helper - */ - -/** - * snd_hda_ch_mode_info - Info callback helper for the channel mode enum - * @codec: the HDA codec - * @uinfo: pointer to get/store the data - * @chmode: channel mode array - * @num_chmodes: channel mode array size - */ -int snd_hda_ch_mode_info(struct hda_codec *codec, - struct snd_ctl_elem_info *uinfo, - const struct hda_channel_mode *chmode, - int num_chmodes) -{ - uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; - uinfo->count = 1; - uinfo->value.enumerated.items = num_chmodes; - if (uinfo->value.enumerated.item >= num_chmodes) - uinfo->value.enumerated.item = num_chmodes - 1; - sprintf(uinfo->value.enumerated.name, "%dch", - chmode[uinfo->value.enumerated.item].channels); - return 0; -} -EXPORT_SYMBOL_GPL(snd_hda_ch_mode_info); - -/** - * snd_hda_ch_mode_get - Get callback helper for the channel mode enum - * @codec: the HDA codec - * @ucontrol: pointer to get/store the data - * @chmode: channel mode array - * @num_chmodes: channel mode array size - * @max_channels: max number of channels - */ -int snd_hda_ch_mode_get(struct hda_codec *codec, - struct snd_ctl_elem_value *ucontrol, - const struct hda_channel_mode *chmode, - int num_chmodes, - int max_channels) -{ - int i; - - for (i = 0; i < num_chmodes; i++) { - if (max_channels == chmode[i].channels) { - ucontrol->value.enumerated.item[0] = i; - break; - } - } - return 0; -} -EXPORT_SYMBOL_GPL(snd_hda_ch_mode_get); - -/** - * snd_hda_ch_mode_put - Put callback helper for the channel mode enum - * @codec: the HDA codec - * @ucontrol: pointer to get/store the data - * @chmode: channel mode array - * @num_chmodes: channel mode array size - * @max_channelsp: pointer to store the max channels - */ -int snd_hda_ch_mode_put(struct hda_codec *codec, - struct snd_ctl_elem_value *ucontrol, - const struct hda_channel_mode *chmode, - int num_chmodes, - int *max_channelsp) -{ - unsigned int mode; - - mode = ucontrol->value.enumerated.item[0]; - if (mode >= num_chmodes) - return -EINVAL; - if (*max_channelsp == chmode[mode].channels) - return 0; - /* change the current channel setting */ - *max_channelsp = chmode[mode].channels; - if (chmode[mode].sequence) - snd_hda_sequence_write_cache(codec, chmode[mode].sequence); - return 1; -} -EXPORT_SYMBOL_GPL(snd_hda_ch_mode_put); - -/* * input MUX helper */ @@ -5418,24 +4946,6 @@ static void cleanup_dig_out_stream(struct hda_codec *codec, hda_nid_t nid) } /** - * snd_hda_bus_reboot_notify - call the reboot notifier of each codec - * @bus: HD-audio bus - */ -void snd_hda_bus_reboot_notify(struct hda_bus *bus) -{ - struct hda_codec *codec; - - if (!bus) - return; - list_for_each_entry(codec, &bus->codec_list, list) { - if (hda_codec_is_power_on(codec) && - codec->patch_ops.reboot_notify) - codec->patch_ops.reboot_notify(codec); - } -} -EXPORT_SYMBOL_GPL(snd_hda_bus_reboot_notify); - -/** * snd_hda_multi_out_dig_open - open the digital out in the exclusive mode * @codec: the HDA codec * @mout: hda_multi_out object @@ -5825,77 +5335,26 @@ int snd_hda_add_imux_item(struct hda_codec *codec, } EXPORT_SYMBOL_GPL(snd_hda_add_imux_item); - -#ifdef CONFIG_PM -/* - * power management - */ - - -static void hda_async_suspend(void *data, async_cookie_t cookie) -{ - hda_call_codec_suspend(data, false); -} - -static void hda_async_resume(void *data, async_cookie_t cookie) -{ - hda_call_codec_resume(data); -} - /** - * snd_hda_suspend - suspend the codecs - * @bus: the HDA bus - * - * Returns 0 if successful. + * snd_hda_bus_reset - Reset the bus + * @bus: HD-audio bus */ -int snd_hda_suspend(struct hda_bus *bus) +void snd_hda_bus_reset(struct hda_bus *bus) { struct hda_codec *codec; - ASYNC_DOMAIN_EXCLUSIVE(domain); list_for_each_entry(codec, &bus->codec_list, list) { + /* FIXME: maybe a better way needed for forced reset */ cancel_delayed_work_sync(&codec->jackpoll_work); +#ifdef CONFIG_PM if (hda_codec_is_power_on(codec)) { - if (bus->num_codecs > 1) - async_schedule_domain(hda_async_suspend, codec, - &domain); - else - hda_call_codec_suspend(codec, false); - } - } - - if (bus->num_codecs > 1) - async_synchronize_full_domain(&domain); - - return 0; -} -EXPORT_SYMBOL_GPL(snd_hda_suspend); - -/** - * snd_hda_resume - resume the codecs - * @bus: the HDA bus - * - * Returns 0 if successful. - */ -int snd_hda_resume(struct hda_bus *bus) -{ - struct hda_codec *codec; - ASYNC_DOMAIN_EXCLUSIVE(domain); - - list_for_each_entry(codec, &bus->codec_list, list) { - if (bus->num_codecs > 1) - async_schedule_domain(hda_async_resume, codec, &domain); - else + hda_call_codec_suspend(codec); hda_call_codec_resume(codec); + } +#endif } - - if (bus->num_codecs > 1) - async_synchronize_full_domain(&domain); - - return 0; } -EXPORT_SYMBOL_GPL(snd_hda_resume); -#endif /* CONFIG_PM */ +EXPORT_SYMBOL_GPL(snd_hda_bus_reset); /* * generic arrays diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index 9c8820f..ccf355d4 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -21,6 +21,7 @@ #ifndef __SOUND_HDA_CODEC_H #define __SOUND_HDA_CODEC_H +#include <linux/kref.h> #include <sound/info.h> #include <sound/control.h> #include <sound/pcm.h> @@ -66,7 +67,6 @@ struct hda_beep; struct hda_codec; struct hda_pcm; struct hda_pcm_stream; -struct hda_bus_unsolicited; /* NID type */ typedef u16 hda_nid_t; @@ -84,10 +84,6 @@ struct hda_bus_ops { struct hda_pcm *pcm); /* reset bus for retry verb */ void (*bus_reset)(struct hda_bus *bus); -#ifdef CONFIG_PM - /* notify power-up/down from codec to controller */ - void (*pm_notify)(struct hda_bus *bus, bool power_up); -#endif #ifdef CONFIG_SND_HDA_DSP_LOADER /* prepare DSP transfer */ int (*load_dsp_prepare)(struct hda_bus *bus, unsigned int format, @@ -101,13 +97,14 @@ struct hda_bus_ops { #endif }; -/* template to pass to the bus constructor */ -struct hda_bus_template { - void *private_data; - struct pci_dev *pci; - const char *modelname; - int *power_save; - struct hda_bus_ops ops; +/* unsolicited event handler */ +#define HDA_UNSOL_QUEUE_SIZE 64 +struct hda_bus_unsolicited { + /* ring buffer */ + u32 queue[HDA_UNSOL_QUEUE_SIZE * 2]; + unsigned int rp, wp; + /* workqueue */ + struct work_struct work; }; /* @@ -119,11 +116,9 @@ struct hda_bus_template { struct hda_bus { struct snd_card *card; - /* copied from template */ void *private_data; struct pci_dev *pci; const char *modelname; - int *power_save; struct hda_bus_ops ops; /* codec linked list */ @@ -136,9 +131,7 @@ struct hda_bus { struct mutex prepare_mutex; /* unsolicited event queue */ - struct hda_bus_unsolicited *unsol; - char workq_name[16]; - struct workqueue_struct *workq; /* common workqueue for codecs */ + struct hda_bus_unsolicited unsol; /* assigned PCMs */ DECLARE_BITMAP(pcm_dev_bits, SNDRV_PCM_DEVICES); @@ -152,10 +145,10 @@ struct hda_bus { unsigned int rirb_error:1; /* error in codec communication */ unsigned int response_reset:1; /* controller was reset */ unsigned int in_reset:1; /* during reset operation */ - unsigned int power_keep_link_on:1; /* don't power off HDA link */ unsigned int no_response_fallback:1; /* don't fallback at RIRB error */ int primary_dig_out_type; /* primary digital out PCM type */ + unsigned long codec_powered; /* bit flags of powered codecs */ }; /* @@ -175,15 +168,22 @@ struct hda_codec_preset { int (*patch)(struct hda_codec *codec); }; -struct hda_codec_preset_list { +#define HDA_CODEC_ID_GENERIC_HDMI 0x00000101 +#define HDA_CODEC_ID_GENERIC 0x00000201 + +struct hda_codec_driver { + struct device_driver driver; const struct hda_codec_preset *preset; - struct module *owner; - struct list_head list; }; -/* initial hook */ -int snd_hda_add_codec_preset(struct hda_codec_preset_list *preset); -int snd_hda_delete_codec_preset(struct hda_codec_preset_list *preset); +int __hda_codec_driver_register(struct hda_codec_driver *drv, const char *name, + struct module *owner); +#define hda_codec_driver_register(drv) \ + __hda_codec_driver_register(drv, KBUILD_MODNAME, THIS_MODULE) +void hda_codec_driver_unregister(struct hda_codec_driver *drv); +#define module_hda_codec_driver(drv) \ + module_driver(drv, hda_codec_driver_register, \ + hda_codec_driver_unregister) /* ops set by the preset patch */ struct hda_codec_ops { @@ -200,6 +200,7 @@ struct hda_codec_ops { int (*check_power_status)(struct hda_codec *codec, hda_nid_t nid); #endif void (*reboot_notify)(struct hda_codec *codec); + void (*stream_pm)(struct hda_codec *codec, hda_nid_t nid, bool on); }; /* record for amp information cache */ @@ -267,12 +268,17 @@ struct hda_pcm { int device; /* device number to assign */ struct snd_pcm *pcm; /* assigned PCM instance */ bool own_chmap; /* codec driver provides own channel maps */ + /* private: */ + struct hda_codec *codec; + struct kref kref; + struct list_head list; }; /* codec information */ struct hda_codec { struct device dev; struct hda_bus *bus; + struct snd_card *card; unsigned int addr; /* codec addr*/ struct list_head list; /* list point */ @@ -287,11 +293,10 @@ struct hda_codec { u32 vendor_id; u32 subsystem_id; u32 revision_id; + u32 probe_id; /* overridden id for probing */ /* detected preset */ const struct hda_codec_preset *preset; - struct module *owner; - int (*parser)(struct hda_codec *codec); const char *vendor_name; /* codec vendor name */ const char *chip_name; /* codec chip name */ const char *modelname; /* model name for preset */ @@ -300,8 +305,7 @@ struct hda_codec { struct hda_codec_ops patch_ops; /* PCM to create, set by patch_ops.build_pcms callback */ - unsigned int num_pcms; - struct hda_pcm *pcm_info; + struct list_head pcm_list_head; /* codec specific info */ void *spec; @@ -345,6 +349,7 @@ struct hda_codec { #endif /* misc flags */ + unsigned int in_freeing:1; /* being released */ unsigned int spdif_status_reset :1; /* needs to toggle SPDIF for each * status change * (e.g. Realtek codecs) @@ -366,18 +371,13 @@ struct hda_codec { unsigned int cached_write:1; /* write only to caches */ unsigned int dp_mst:1; /* support DP1.2 Multi-stream transport */ unsigned int dump_coef:1; /* dump processing coefs in codec proc file */ + unsigned int power_save_node:1; /* advanced PM for each widget */ #ifdef CONFIG_PM - unsigned int power_on :1; /* current (global) power-state */ unsigned int d3_stop_clk:1; /* support D3 operation without BCLK */ - unsigned int pm_up_notified:1; /* PM notified to controller */ - unsigned int in_pm:1; /* suspend/resume being performed */ - int power_transition; /* power-state in transition */ - int power_count; /* current (global) power refcount */ - struct delayed_work power_work; /* delayed task for powerdown */ + atomic_t in_pm; /* suspend/resume being performed */ unsigned long power_on_acct; unsigned long power_off_acct; unsigned long power_jiffies; - spinlock_t power_lock; #endif /* filter the requested power state per nid */ @@ -409,6 +409,11 @@ struct hda_codec { struct snd_array verbs; }; +#define dev_to_hda_codec(_dev) container_of(_dev, struct hda_codec, dev) +#define hda_codec_dev(_dev) (&(_dev)->dev) + +extern struct bus_type snd_hda_bus_type; + /* direction */ enum { HDA_INPUT, HDA_OUTPUT @@ -420,10 +425,9 @@ enum { /* * constructors */ -int snd_hda_bus_new(struct snd_card *card, const struct hda_bus_template *temp, - struct hda_bus **busp); -int snd_hda_codec_new(struct hda_bus *bus, unsigned int codec_addr, - struct hda_codec **codecp); +int snd_hda_bus_new(struct snd_card *card, struct hda_bus **busp); +int snd_hda_codec_new(struct hda_bus *bus, struct snd_card *card, + unsigned int codec_addr, struct hda_codec **codecp); int snd_hda_codec_configure(struct hda_codec *codec); int snd_hda_codec_update_widgets(struct hda_codec *codec); @@ -512,15 +516,24 @@ void snd_hda_spdif_ctls_assign(struct hda_codec *codec, int idx, hda_nid_t nid); /* * Mixer */ -int snd_hda_build_controls(struct hda_bus *bus); int snd_hda_codec_build_controls(struct hda_codec *codec); /* * PCM */ -int snd_hda_build_pcms(struct hda_bus *bus); +int snd_hda_codec_parse_pcms(struct hda_codec *codec); int snd_hda_codec_build_pcms(struct hda_codec *codec); +__printf(2, 3) +struct hda_pcm *snd_hda_codec_pcm_new(struct hda_codec *codec, + const char *fmt, ...); + +static inline void snd_hda_codec_pcm_get(struct hda_pcm *pcm) +{ + kref_get(&pcm->kref); +} +void snd_hda_codec_pcm_put(struct hda_pcm *pcm); + int snd_hda_codec_prepare(struct hda_codec *codec, struct hda_pcm_stream *hinfo, unsigned int stream, @@ -552,20 +565,17 @@ extern const struct snd_pcm_chmap_elem snd_pcm_2_1_chmaps[]; * Misc */ void snd_hda_get_codec_name(struct hda_codec *codec, char *name, int namelen); -void snd_hda_bus_reboot_notify(struct hda_bus *bus); void snd_hda_codec_set_power_to_all(struct hda_codec *codec, hda_nid_t fg, unsigned int power_state); int snd_hda_lock_devices(struct hda_bus *bus); void snd_hda_unlock_devices(struct hda_bus *bus); +void snd_hda_bus_reset(struct hda_bus *bus); /* * power management */ -#ifdef CONFIG_PM -int snd_hda_suspend(struct hda_bus *bus); -int snd_hda_resume(struct hda_bus *bus); -#endif +extern const struct dev_pm_ops hda_codec_driver_pm; static inline int hda_call_check_power_status(struct hda_codec *codec, hda_nid_t nid) @@ -588,64 +598,16 @@ const char *snd_hda_get_jack_location(u32 cfg); * power saving */ #ifdef CONFIG_PM -void snd_hda_power_save(struct hda_codec *codec, int delta, bool d3wait); +void snd_hda_power_up(struct hda_codec *codec); +void snd_hda_power_down(struct hda_codec *codec); +void snd_hda_set_power_save(struct hda_bus *bus, int delay); void snd_hda_update_power_acct(struct hda_codec *codec); #else -static inline void snd_hda_power_save(struct hda_codec *codec, int delta, - bool d3wait) {} +static inline void snd_hda_power_up(struct hda_codec *codec) {} +static inline void snd_hda_power_down(struct hda_codec *codec) {} +static inline void snd_hda_set_power_save(struct hda_bus *bus, int delay) {} #endif -/** - * snd_hda_power_up - Power-up the codec - * @codec: HD-audio codec - * - * Increment the power-up counter and power up the hardware really when - * not turned on yet. - */ -static inline void snd_hda_power_up(struct hda_codec *codec) -{ - snd_hda_power_save(codec, 1, false); -} - -/** - * snd_hda_power_up_d3wait - Power-up the codec after waiting for any pending - * D3 transition to complete. This differs from snd_hda_power_up() when - * power_transition == -1. snd_hda_power_up sees this case as a nop, - * snd_hda_power_up_d3wait waits for the D3 transition to complete then powers - * back up. - * @codec: HD-audio codec - * - * Cancel any power down operation hapenning on the work queue, then power up. - */ -static inline void snd_hda_power_up_d3wait(struct hda_codec *codec) -{ - snd_hda_power_save(codec, 1, true); -} - -/** - * snd_hda_power_down - Power-down the codec - * @codec: HD-audio codec - * - * Decrement the power-up counter and schedules the power-off work if - * the counter rearches to zero. - */ -static inline void snd_hda_power_down(struct hda_codec *codec) -{ - snd_hda_power_save(codec, -1, false); -} - -/** - * snd_hda_power_sync - Synchronize the power-save status - * @codec: HD-audio codec - * - * Synchronize the actual power state with the power account; - * called when power_save parameter is changed - */ -static inline void snd_hda_power_sync(struct hda_codec *codec) -{ - snd_hda_power_save(codec, 0, false); -} - #ifdef CONFIG_SND_HDA_PATCH_LOADER /* * patch firmware diff --git a/sound/pci/hda/hda_controller.c b/sound/pci/hda/hda_controller.c index 17c2637..4fd0b2e 100644 --- a/sound/pci/hda/hda_controller.c +++ b/sound/pci/hda/hda_controller.c @@ -27,10 +27,8 @@ #include <linux/module.h> #include <linux/pm_runtime.h> #include <linux/slab.h> -#include <linux/reboot.h> #include <sound/core.h> #include <sound/initval.h> -#include "hda_priv.h" #include "hda_controller.h" #define CREATE_TRACE_POINTS @@ -259,11 +257,18 @@ static void azx_timecounter_init(struct snd_pcm_substream *substream, tc->cycle_last = last; } +static inline struct hda_pcm_stream * +to_hda_pcm_stream(struct snd_pcm_substream *substream) +{ + struct azx_pcm *apcm = snd_pcm_substream_chip(substream); + return &apcm->info->stream[substream->stream]; +} + static u64 azx_adjust_codec_delay(struct snd_pcm_substream *substream, u64 nsec) { struct azx_pcm *apcm = snd_pcm_substream_chip(substream); - struct hda_pcm_stream *hinfo = apcm->hinfo[substream->stream]; + struct hda_pcm_stream *hinfo = to_hda_pcm_stream(substream); u64 codec_frames, codec_nsecs; if (!hinfo->ops.get_delay) @@ -399,7 +404,7 @@ static int azx_setup_periods(struct azx *chip, static int azx_pcm_close(struct snd_pcm_substream *substream) { struct azx_pcm *apcm = snd_pcm_substream_chip(substream); - struct hda_pcm_stream *hinfo = apcm->hinfo[substream->stream]; + struct hda_pcm_stream *hinfo = to_hda_pcm_stream(substream); struct azx *chip = apcm->chip; struct azx_dev *azx_dev = get_azx_dev(substream); unsigned long flags; @@ -410,9 +415,11 @@ static int azx_pcm_close(struct snd_pcm_substream *substream) azx_dev->running = 0; spin_unlock_irqrestore(&chip->reg_lock, flags); azx_release_device(azx_dev); - hinfo->ops.close(hinfo, apcm->codec, substream); + if (hinfo->ops.close) + hinfo->ops.close(hinfo, apcm->codec, substream); snd_hda_power_down(apcm->codec); mutex_unlock(&chip->open_mutex); + snd_hda_codec_pcm_put(apcm->info); return 0; } @@ -441,7 +448,7 @@ static int azx_pcm_hw_free(struct snd_pcm_substream *substream) struct azx_pcm *apcm = snd_pcm_substream_chip(substream); struct azx_dev *azx_dev = get_azx_dev(substream); struct azx *chip = apcm->chip; - struct hda_pcm_stream *hinfo = apcm->hinfo[substream->stream]; + struct hda_pcm_stream *hinfo = to_hda_pcm_stream(substream); int err; /* reset BDL address */ @@ -468,7 +475,7 @@ static int azx_pcm_prepare(struct snd_pcm_substream *substream) struct azx_pcm *apcm = snd_pcm_substream_chip(substream); struct azx *chip = apcm->chip; struct azx_dev *azx_dev = get_azx_dev(substream); - struct hda_pcm_stream *hinfo = apcm->hinfo[substream->stream]; + struct hda_pcm_stream *hinfo = to_hda_pcm_stream(substream); struct snd_pcm_runtime *runtime = substream->runtime; unsigned int bufsize, period_bytes, format_val, stream_tag; int err; @@ -708,7 +715,7 @@ unsigned int azx_get_position(struct azx *chip, if (substream->runtime) { struct azx_pcm *apcm = snd_pcm_substream_chip(substream); - struct hda_pcm_stream *hinfo = apcm->hinfo[stream]; + struct hda_pcm_stream *hinfo = to_hda_pcm_stream(substream); if (chip->get_delay[stream]) delay += chip->get_delay[stream](chip, azx_dev, pos); @@ -732,17 +739,32 @@ static snd_pcm_uframes_t azx_pcm_pointer(struct snd_pcm_substream *substream) azx_get_position(chip, azx_dev)); } -static int azx_get_wallclock_tstamp(struct snd_pcm_substream *substream, - struct timespec *ts) +static int azx_get_time_info(struct snd_pcm_substream *substream, + struct timespec *system_ts, struct timespec *audio_ts, + struct snd_pcm_audio_tstamp_config *audio_tstamp_config, + struct snd_pcm_audio_tstamp_report *audio_tstamp_report) { struct azx_dev *azx_dev = get_azx_dev(substream); u64 nsec; - nsec = timecounter_read(&azx_dev->azx_tc); - nsec = div_u64(nsec, 3); /* can be optimized */ - nsec = azx_adjust_codec_delay(substream, nsec); + if ((substream->runtime->hw.info & SNDRV_PCM_INFO_HAS_LINK_ATIME) && + (audio_tstamp_config->type_requested == SNDRV_PCM_AUDIO_TSTAMP_TYPE_LINK)) { + + snd_pcm_gettime(substream->runtime, system_ts); - *ts = ns_to_timespec(nsec); + nsec = timecounter_read(&azx_dev->azx_tc); + nsec = div_u64(nsec, 3); /* can be optimized */ + if (audio_tstamp_config->report_delay) + nsec = azx_adjust_codec_delay(substream, nsec); + + *audio_ts = ns_to_timespec(nsec); + + audio_tstamp_report->actual_type = SNDRV_PCM_AUDIO_TSTAMP_TYPE_LINK; + audio_tstamp_report->accuracy_report = 1; /* rest of structure is valid */ + audio_tstamp_report->accuracy = 42; /* 24 MHz WallClock == 42ns resolution */ + + } else + audio_tstamp_report->actual_type = SNDRV_PCM_AUDIO_TSTAMP_TYPE_DEFAULT; return 0; } @@ -756,7 +778,8 @@ static struct snd_pcm_hardware azx_pcm_hw = { /* SNDRV_PCM_INFO_RESUME |*/ SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_SYNC_START | - SNDRV_PCM_INFO_HAS_WALL_CLOCK | + SNDRV_PCM_INFO_HAS_WALL_CLOCK | /* legacy */ + SNDRV_PCM_INFO_HAS_LINK_ATIME | SNDRV_PCM_INFO_NO_PERIOD_WAKEUP), .formats = SNDRV_PCM_FMTBIT_S16_LE, .rates = SNDRV_PCM_RATE_48000, @@ -775,7 +798,7 @@ static struct snd_pcm_hardware azx_pcm_hw = { static int azx_pcm_open(struct snd_pcm_substream *substream) { struct azx_pcm *apcm = snd_pcm_substream_chip(substream); - struct hda_pcm_stream *hinfo = apcm->hinfo[substream->stream]; + struct hda_pcm_stream *hinfo = to_hda_pcm_stream(substream); struct azx *chip = apcm->chip; struct azx_dev *azx_dev; struct snd_pcm_runtime *runtime = substream->runtime; @@ -783,11 +806,12 @@ static int azx_pcm_open(struct snd_pcm_substream *substream) int err; int buff_step; + snd_hda_codec_pcm_get(apcm->info); mutex_lock(&chip->open_mutex); azx_dev = azx_assign_device(chip, substream); if (azx_dev == NULL) { - mutex_unlock(&chip->open_mutex); - return -EBUSY; + err = -EBUSY; + goto unlock; } runtime->hw = azx_pcm_hw; runtime->hw.channels_min = hinfo->channels_min; @@ -821,13 +845,14 @@ static int azx_pcm_open(struct snd_pcm_substream *substream) buff_step); snd_pcm_hw_constraint_step(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_BYTES, buff_step); - snd_hda_power_up_d3wait(apcm->codec); - err = hinfo->ops.open(hinfo, apcm->codec, substream); + snd_hda_power_up(apcm->codec); + if (hinfo->ops.open) + err = hinfo->ops.open(hinfo, apcm->codec, substream); + else + err = -ENODEV; if (err < 0) { azx_release_device(azx_dev); - snd_hda_power_down(apcm->codec); - mutex_unlock(&chip->open_mutex); - return err; + goto powerdown; } snd_pcm_limit_hw_rates(runtime); /* sanity check */ @@ -836,16 +861,18 @@ static int azx_pcm_open(struct snd_pcm_substream *substream) snd_BUG_ON(!runtime->hw.formats) || snd_BUG_ON(!runtime->hw.rates)) { azx_release_device(azx_dev); - hinfo->ops.close(hinfo, apcm->codec, substream); - snd_hda_power_down(apcm->codec); - mutex_unlock(&chip->open_mutex); - return -EINVAL; + if (hinfo->ops.close) + hinfo->ops.close(hinfo, apcm->codec, substream); + err = -EINVAL; + goto powerdown; } - /* disable WALLCLOCK timestamps for capture streams + /* disable LINK_ATIME timestamps for capture streams until we figure out how to handle digital inputs */ - if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) - runtime->hw.info &= ~SNDRV_PCM_INFO_HAS_WALL_CLOCK; + if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) { + runtime->hw.info &= ~SNDRV_PCM_INFO_HAS_WALL_CLOCK; /* legacy */ + runtime->hw.info &= ~SNDRV_PCM_INFO_HAS_LINK_ATIME; + } spin_lock_irqsave(&chip->reg_lock, flags); azx_dev->substream = substream; @@ -856,6 +883,13 @@ static int azx_pcm_open(struct snd_pcm_substream *substream) snd_pcm_set_sync(substream); mutex_unlock(&chip->open_mutex); return 0; + + powerdown: + snd_hda_power_down(apcm->codec); + unlock: + mutex_unlock(&chip->open_mutex); + snd_hda_codec_pcm_put(apcm->info); + return err; } static int azx_pcm_mmap(struct snd_pcm_substream *substream, @@ -877,7 +911,7 @@ static struct snd_pcm_ops azx_pcm_ops = { .prepare = azx_pcm_prepare, .trigger = azx_pcm_trigger, .pointer = azx_pcm_pointer, - .wall_clock = azx_get_wallclock_tstamp, + .get_time_info = azx_get_time_info, .mmap = azx_pcm_mmap, .page = snd_pcm_sgbuf_ops_page, }; @@ -887,6 +921,7 @@ static void azx_pcm_free(struct snd_pcm *pcm) struct azx_pcm *apcm = pcm->private_data; if (apcm) { list_del(&apcm->list); + apcm->info->pcm = NULL; kfree(apcm); } } @@ -923,6 +958,7 @@ static int azx_attach_pcm_stream(struct hda_bus *bus, struct hda_codec *codec, apcm->chip = chip; apcm->pcm = pcm; apcm->codec = codec; + apcm->info = cpcm; pcm->private_data = apcm; pcm->private_free = azx_pcm_free; if (cpcm->pcm_type == HDA_PCM_TYPE_MODEM) @@ -930,7 +966,6 @@ static int azx_attach_pcm_stream(struct hda_bus *bus, struct hda_codec *codec, list_add_tail(&apcm->list, &chip->pcm_list); cpcm->pcm = pcm; for (s = 0; s < 2; s++) { - apcm->hinfo[s] = &cpcm->stream[s]; if (cpcm->stream[s].substreams) snd_pcm_set_ops(pcm, s, &azx_pcm_ops); } @@ -941,9 +976,6 @@ static int azx_attach_pcm_stream(struct hda_bus *bus, struct hda_codec *codec, snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV_SG, chip->card->dev, size, MAX_PREALLOC_SIZE); - /* link to codec */ - for (s = 0; s < 2; s++) - pcm->streams[s].dev.parent = &codec->dev; return 0; } @@ -952,14 +984,9 @@ static int azx_attach_pcm_stream(struct hda_bus *bus, struct hda_codec *codec, */ static int azx_alloc_cmd_io(struct azx *chip) { - int err; - /* single page (at least 4096 bytes) must suffice for both ringbuffes */ - err = chip->ops->dma_alloc_pages(chip, SNDRV_DMA_TYPE_DEV, - PAGE_SIZE, &chip->rb); - if (err < 0) - dev_err(chip->card->dev, "cannot allocate CORB/RIRB\n"); - return err; + return chip->ops->dma_alloc_pages(chip, SNDRV_DMA_TYPE_DEV, + PAGE_SIZE, &chip->rb); } static void azx_init_cmd_io(struct azx *chip) @@ -1445,7 +1472,6 @@ static void azx_load_dsp_cleanup(struct hda_bus *bus, int azx_alloc_stream_pages(struct azx *chip) { int i, err; - struct snd_card *card = chip->card; for (i = 0; i < chip->num_streams; i++) { dsp_lock_init(&chip->azx_dev[i]); @@ -1453,18 +1479,14 @@ int azx_alloc_stream_pages(struct azx *chip) err = chip->ops->dma_alloc_pages(chip, SNDRV_DMA_TYPE_DEV, BDL_SIZE, &chip->azx_dev[i].bdl); - if (err < 0) { - dev_err(card->dev, "cannot allocate BDL\n"); + if (err < 0) return -ENOMEM; - } } /* allocate memory for the position buffer */ err = chip->ops->dma_alloc_pages(chip, SNDRV_DMA_TYPE_DEV, chip->num_streams * 8, &chip->posbuf); - if (err < 0) { - dev_err(card->dev, "cannot allocate posbuf\n"); + if (err < 0) return -ENOMEM; - } /* allocate CORB/RIRB */ err = azx_alloc_cmd_io(chip); @@ -1676,7 +1698,7 @@ irqreturn_t azx_interrupt(int irq, void *dev_id) int i; #ifdef CONFIG_PM - if (chip->driver_caps & AZX_DCAPS_PM_RUNTIME) + if (azx_has_pm_runtime(chip)) if (!pm_runtime_active(chip->card->dev)) return IRQ_NONE; #endif @@ -1761,34 +1783,11 @@ static void azx_bus_reset(struct hda_bus *bus) bus->in_reset = 1; azx_stop_chip(chip); azx_init_chip(chip, true); -#ifdef CONFIG_PM - if (chip->initialized) { - struct azx_pcm *p; - list_for_each_entry(p, &chip->pcm_list, list) - snd_pcm_suspend_all(p->pcm); - snd_hda_suspend(chip->bus); - snd_hda_resume(chip->bus); - } -#endif + if (chip->initialized) + snd_hda_bus_reset(chip->bus); bus->in_reset = 0; } -#ifdef CONFIG_PM -/* power-up/down the controller */ -static void azx_power_notify(struct hda_bus *bus, bool power_up) -{ - struct azx *chip = bus->private_data; - - if (!(chip->driver_caps & AZX_DCAPS_PM_RUNTIME)) - return; - - if (power_up) - pm_runtime_get_sync(chip->card->dev); - else - pm_runtime_put_sync(chip->card->dev); -} -#endif - static int get_jackpoll_interval(struct azx *chip) { int i; @@ -1810,41 +1809,59 @@ static int get_jackpoll_interval(struct azx *chip) return j; } -/* Codec initialization */ -int azx_codec_create(struct azx *chip, const char *model, - unsigned int max_slots, - int *power_save_to) -{ - struct hda_bus_template bus_temp; - int c, codecs, err; - - memset(&bus_temp, 0, sizeof(bus_temp)); - bus_temp.private_data = chip; - bus_temp.modelname = model; - bus_temp.pci = chip->pci; - bus_temp.ops.command = azx_send_cmd; - bus_temp.ops.get_response = azx_get_response; - bus_temp.ops.attach_pcm = azx_attach_pcm_stream; - bus_temp.ops.bus_reset = azx_bus_reset; -#ifdef CONFIG_PM - bus_temp.power_save = power_save_to; - bus_temp.ops.pm_notify = azx_power_notify; -#endif +static struct hda_bus_ops bus_ops = { + .command = azx_send_cmd, + .get_response = azx_get_response, + .attach_pcm = azx_attach_pcm_stream, + .bus_reset = azx_bus_reset, #ifdef CONFIG_SND_HDA_DSP_LOADER - bus_temp.ops.load_dsp_prepare = azx_load_dsp_prepare; - bus_temp.ops.load_dsp_trigger = azx_load_dsp_trigger; - bus_temp.ops.load_dsp_cleanup = azx_load_dsp_cleanup; + .load_dsp_prepare = azx_load_dsp_prepare, + .load_dsp_trigger = azx_load_dsp_trigger, + .load_dsp_cleanup = azx_load_dsp_cleanup, #endif +}; + +/* HD-audio bus initialization */ +int azx_bus_create(struct azx *chip, const char *model) +{ + struct hda_bus *bus; + int err; - err = snd_hda_bus_new(chip->card, &bus_temp, &chip->bus); + err = snd_hda_bus_new(chip->card, &bus); if (err < 0) return err; + chip->bus = bus; + bus->private_data = chip; + bus->pci = chip->pci; + bus->modelname = model; + bus->ops = bus_ops; + if (chip->driver_caps & AZX_DCAPS_RIRB_DELAY) { dev_dbg(chip->card->dev, "Enable delay in RIRB handling\n"); - chip->bus->needs_damn_long_delay = 1; + bus->needs_damn_long_delay = 1; + } + + /* AMD chipsets often cause the communication stalls upon certain + * sequence like the pin-detection. It seems that forcing the synced + * access works around the stall. Grrr... + */ + if (chip->driver_caps & AZX_DCAPS_SYNC_WRITE) { + dev_dbg(chip->card->dev, "Enable sync_write for stable communication\n"); + bus->sync_write = 1; + bus->allow_bus_reset = 1; } + return 0; +} +EXPORT_SYMBOL_GPL(azx_bus_create); + +/* Probe codecs */ +int azx_probe_codecs(struct azx *chip, unsigned int max_slots) +{ + struct hda_bus *bus = chip->bus; + int c, codecs, err; + codecs = 0; if (!max_slots) max_slots = AZX_DEFAULT_CODECS; @@ -1872,21 +1889,11 @@ int azx_codec_create(struct azx *chip, const char *model, } } - /* AMD chipsets often cause the communication stalls upon certain - * sequence like the pin-detection. It seems that forcing the synced - * access works around the stall. Grrr... - */ - if (chip->driver_caps & AZX_DCAPS_SYNC_WRITE) { - dev_dbg(chip->card->dev, "Enable sync_write for stable communication\n"); - chip->bus->sync_write = 1; - chip->bus->allow_bus_reset = 1; - } - /* Then create codec instances */ for (c = 0; c < max_slots; c++) { if ((chip->codec_mask & (1 << c)) & chip->codec_probe_mask) { struct hda_codec *codec; - err = snd_hda_codec_new(chip->bus, c, &codec); + err = snd_hda_codec_new(bus, bus->card, c, &codec); if (err < 0) continue; codec->jackpoll_interval = get_jackpoll_interval(chip); @@ -1900,7 +1907,7 @@ int azx_codec_create(struct azx *chip, const char *model, } return 0; } -EXPORT_SYMBOL_GPL(azx_codec_create); +EXPORT_SYMBOL_GPL(azx_probe_codecs); /* configure each codec instance */ int azx_codec_configure(struct azx *chip) @@ -1913,13 +1920,6 @@ int azx_codec_configure(struct azx *chip) } EXPORT_SYMBOL_GPL(azx_codec_configure); -/* mixer creation - all stuff is implemented in hda module */ -int azx_mixer_create(struct azx *chip) -{ - return snd_hda_build_controls(chip->bus); -} -EXPORT_SYMBOL_GPL(azx_mixer_create); - static bool is_input_stream(struct azx *chip, unsigned char index) { @@ -1966,30 +1966,5 @@ int azx_init_stream(struct azx *chip) } EXPORT_SYMBOL_GPL(azx_init_stream); -/* - * reboot notifier for hang-up problem at power-down - */ -static int azx_halt(struct notifier_block *nb, unsigned long event, void *buf) -{ - struct azx *chip = container_of(nb, struct azx, reboot_notifier); - snd_hda_bus_reboot_notify(chip->bus); - azx_stop_chip(chip); - return NOTIFY_OK; -} - -void azx_notifier_register(struct azx *chip) -{ - chip->reboot_notifier.notifier_call = azx_halt; - register_reboot_notifier(&chip->reboot_notifier); -} -EXPORT_SYMBOL_GPL(azx_notifier_register); - -void azx_notifier_unregister(struct azx *chip) -{ - if (chip->reboot_notifier.notifier_call) - unregister_reboot_notifier(&chip->reboot_notifier); -} -EXPORT_SYMBOL_GPL(azx_notifier_unregister); - MODULE_LICENSE("GPL"); MODULE_DESCRIPTION("Common HDA driver functions"); diff --git a/sound/pci/hda/hda_controller.h b/sound/pci/hda/hda_controller.h index c90d10f..be1b7de 100644 --- a/sound/pci/hda/hda_controller.h +++ b/sound/pci/hda/hda_controller.h @@ -15,10 +15,396 @@ #ifndef __SOUND_HDA_CONTROLLER_H #define __SOUND_HDA_CONTROLLER_H +#include <linux/timecounter.h> +#include <linux/interrupt.h> #include <sound/core.h> +#include <sound/pcm.h> #include <sound/initval.h> #include "hda_codec.h" -#include "hda_priv.h" + +/* + * registers + */ +#define AZX_REG_GCAP 0x00 +#define AZX_GCAP_64OK (1 << 0) /* 64bit address support */ +#define AZX_GCAP_NSDO (3 << 1) /* # of serial data out signals */ +#define AZX_GCAP_BSS (31 << 3) /* # of bidirectional streams */ +#define AZX_GCAP_ISS (15 << 8) /* # of input streams */ +#define AZX_GCAP_OSS (15 << 12) /* # of output streams */ +#define AZX_REG_VMIN 0x02 +#define AZX_REG_VMAJ 0x03 +#define AZX_REG_OUTPAY 0x04 +#define AZX_REG_INPAY 0x06 +#define AZX_REG_GCTL 0x08 +#define AZX_GCTL_RESET (1 << 0) /* controller reset */ +#define AZX_GCTL_FCNTRL (1 << 1) /* flush control */ +#define AZX_GCTL_UNSOL (1 << 8) /* accept unsol. response enable */ +#define AZX_REG_WAKEEN 0x0c +#define AZX_REG_STATESTS 0x0e +#define AZX_REG_GSTS 0x10 +#define AZX_GSTS_FSTS (1 << 1) /* flush status */ +#define AZX_REG_INTCTL 0x20 +#define AZX_REG_INTSTS 0x24 +#define AZX_REG_WALLCLK 0x30 /* 24Mhz source */ +#define AZX_REG_OLD_SSYNC 0x34 /* SSYNC for old ICH */ +#define AZX_REG_SSYNC 0x38 +#define AZX_REG_CORBLBASE 0x40 +#define AZX_REG_CORBUBASE 0x44 +#define AZX_REG_CORBWP 0x48 +#define AZX_REG_CORBRP 0x4a +#define AZX_CORBRP_RST (1 << 15) /* read pointer reset */ +#define AZX_REG_CORBCTL 0x4c +#define AZX_CORBCTL_RUN (1 << 1) /* enable DMA */ +#define AZX_CORBCTL_CMEIE (1 << 0) /* enable memory error irq */ +#define AZX_REG_CORBSTS 0x4d +#define AZX_CORBSTS_CMEI (1 << 0) /* memory error indication */ +#define AZX_REG_CORBSIZE 0x4e + +#define AZX_REG_RIRBLBASE 0x50 +#define AZX_REG_RIRBUBASE 0x54 +#define AZX_REG_RIRBWP 0x58 +#define AZX_RIRBWP_RST (1 << 15) /* write pointer reset */ +#define AZX_REG_RINTCNT 0x5a +#define AZX_REG_RIRBCTL 0x5c +#define AZX_RBCTL_IRQ_EN (1 << 0) /* enable IRQ */ +#define AZX_RBCTL_DMA_EN (1 << 1) /* enable DMA */ +#define AZX_RBCTL_OVERRUN_EN (1 << 2) /* enable overrun irq */ +#define AZX_REG_RIRBSTS 0x5d +#define AZX_RBSTS_IRQ (1 << 0) /* response irq */ +#define AZX_RBSTS_OVERRUN (1 << 2) /* overrun irq */ +#define AZX_REG_RIRBSIZE 0x5e + +#define AZX_REG_IC 0x60 +#define AZX_REG_IR 0x64 +#define AZX_REG_IRS 0x68 +#define AZX_IRS_VALID (1<<1) +#define AZX_IRS_BUSY (1<<0) + +#define AZX_REG_DPLBASE 0x70 +#define AZX_REG_DPUBASE 0x74 +#define AZX_DPLBASE_ENABLE 0x1 /* Enable position buffer */ + +/* SD offset: SDI0=0x80, SDI1=0xa0, ... SDO3=0x160 */ +enum { SDI0, SDI1, SDI2, SDI3, SDO0, SDO1, SDO2, SDO3 }; + +/* stream register offsets from stream base */ +#define AZX_REG_SD_CTL 0x00 +#define AZX_REG_SD_STS 0x03 +#define AZX_REG_SD_LPIB 0x04 +#define AZX_REG_SD_CBL 0x08 +#define AZX_REG_SD_LVI 0x0c +#define AZX_REG_SD_FIFOW 0x0e +#define AZX_REG_SD_FIFOSIZE 0x10 +#define AZX_REG_SD_FORMAT 0x12 +#define AZX_REG_SD_BDLPL 0x18 +#define AZX_REG_SD_BDLPU 0x1c + +/* PCI space */ +#define AZX_PCIREG_TCSEL 0x44 + +/* + * other constants + */ + +/* max number of fragments - we may use more if allocating more pages for BDL */ +#define BDL_SIZE 4096 +#define AZX_MAX_BDL_ENTRIES (BDL_SIZE / 16) +#define AZX_MAX_FRAG 32 +/* max buffer size - no h/w limit, you can increase as you like */ +#define AZX_MAX_BUF_SIZE (1024*1024*1024) + +/* RIRB int mask: overrun[2], response[0] */ +#define RIRB_INT_RESPONSE 0x01 +#define RIRB_INT_OVERRUN 0x04 +#define RIRB_INT_MASK 0x05 + +/* STATESTS int mask: S3,SD2,SD1,SD0 */ +#define AZX_MAX_CODECS 8 +#define AZX_DEFAULT_CODECS 4 +#define STATESTS_INT_MASK ((1 << AZX_MAX_CODECS) - 1) + +/* SD_CTL bits */ +#define SD_CTL_STREAM_RESET 0x01 /* stream reset bit */ +#define SD_CTL_DMA_START 0x02 /* stream DMA start bit */ +#define SD_CTL_STRIPE (3 << 16) /* stripe control */ +#define SD_CTL_TRAFFIC_PRIO (1 << 18) /* traffic priority */ +#define SD_CTL_DIR (1 << 19) /* bi-directional stream */ +#define SD_CTL_STREAM_TAG_MASK (0xf << 20) +#define SD_CTL_STREAM_TAG_SHIFT 20 + +/* SD_CTL and SD_STS */ +#define SD_INT_DESC_ERR 0x10 /* descriptor error interrupt */ +#define SD_INT_FIFO_ERR 0x08 /* FIFO error interrupt */ +#define SD_INT_COMPLETE 0x04 /* completion interrupt */ +#define SD_INT_MASK (SD_INT_DESC_ERR|SD_INT_FIFO_ERR|\ + SD_INT_COMPLETE) + +/* SD_STS */ +#define SD_STS_FIFO_READY 0x20 /* FIFO ready */ + +/* INTCTL and INTSTS */ +#define AZX_INT_ALL_STREAM 0xff /* all stream interrupts */ +#define AZX_INT_CTRL_EN 0x40000000 /* controller interrupt enable bit */ +#define AZX_INT_GLOBAL_EN 0x80000000 /* global interrupt enable bit */ + +/* below are so far hardcoded - should read registers in future */ +#define AZX_MAX_CORB_ENTRIES 256 +#define AZX_MAX_RIRB_ENTRIES 256 + +/* driver quirks (capabilities) */ +/* bits 0-7 are used for indicating driver type */ +#define AZX_DCAPS_NO_TCSEL (1 << 8) /* No Intel TCSEL bit */ +#define AZX_DCAPS_NO_MSI (1 << 9) /* No MSI support */ +#define AZX_DCAPS_SNOOP_MASK (3 << 10) /* snoop type mask */ +#define AZX_DCAPS_SNOOP_OFF (1 << 12) /* snoop default off */ +#define AZX_DCAPS_RIRB_DELAY (1 << 13) /* Long delay in read loop */ +#define AZX_DCAPS_RIRB_PRE_DELAY (1 << 14) /* Put a delay before read */ +#define AZX_DCAPS_CTX_WORKAROUND (1 << 15) /* X-Fi workaround */ +#define AZX_DCAPS_POSFIX_LPIB (1 << 16) /* Use LPIB as default */ +#define AZX_DCAPS_POSFIX_VIA (1 << 17) /* Use VIACOMBO as default */ +#define AZX_DCAPS_NO_64BIT (1 << 18) /* No 64bit address */ +#define AZX_DCAPS_SYNC_WRITE (1 << 19) /* sync each cmd write */ +#define AZX_DCAPS_OLD_SSYNC (1 << 20) /* Old SSYNC reg for ICH */ +#define AZX_DCAPS_NO_ALIGN_BUFSIZE (1 << 21) /* no buffer size alignment */ +/* 22 unused */ +#define AZX_DCAPS_4K_BDLE_BOUNDARY (1 << 23) /* BDLE in 4k boundary */ +#define AZX_DCAPS_REVERSE_ASSIGN (1 << 24) /* Assign devices in reverse order */ +#define AZX_DCAPS_COUNT_LPIB_DELAY (1 << 25) /* Take LPIB as delay */ +#define AZX_DCAPS_PM_RUNTIME (1 << 26) /* runtime PM support */ +#define AZX_DCAPS_I915_POWERWELL (1 << 27) /* HSW i915 powerwell support */ +#define AZX_DCAPS_CORBRP_SELF_CLEAR (1 << 28) /* CORBRP clears itself after reset */ +#define AZX_DCAPS_NO_MSI64 (1 << 29) /* Stick to 32-bit MSIs */ +#define AZX_DCAPS_SEPARATE_STREAM_TAG (1 << 30) /* capture and playback use separate stream tag */ + +enum { + AZX_SNOOP_TYPE_NONE, + AZX_SNOOP_TYPE_SCH, + AZX_SNOOP_TYPE_ATI, + AZX_SNOOP_TYPE_NVIDIA, +}; + +/* HD Audio class code */ +#define PCI_CLASS_MULTIMEDIA_HD_AUDIO 0x0403 + +struct azx_dev { + struct snd_dma_buffer bdl; /* BDL buffer */ + u32 *posbuf; /* position buffer pointer */ + + unsigned int bufsize; /* size of the play buffer in bytes */ + unsigned int period_bytes; /* size of the period in bytes */ + unsigned int frags; /* number for period in the play buffer */ + unsigned int fifo_size; /* FIFO size */ + unsigned long start_wallclk; /* start + minimum wallclk */ + unsigned long period_wallclk; /* wallclk for period */ + + void __iomem *sd_addr; /* stream descriptor pointer */ + + u32 sd_int_sta_mask; /* stream int status mask */ + + /* pcm support */ + struct snd_pcm_substream *substream; /* assigned substream, + * set in PCM open + */ + unsigned int format_val; /* format value to be set in the + * controller and the codec + */ + unsigned char stream_tag; /* assigned stream */ + unsigned char index; /* stream index */ + int assigned_key; /* last device# key assigned to */ + + unsigned int opened:1; + unsigned int running:1; + unsigned int irq_pending:1; + unsigned int prepared:1; + unsigned int locked:1; + /* + * For VIA: + * A flag to ensure DMA position is 0 + * when link position is not greater than FIFO size + */ + unsigned int insufficient:1; + unsigned int wc_marked:1; + unsigned int no_period_wakeup:1; + + struct timecounter azx_tc; + struct cyclecounter azx_cc; + + int delay_negative_threshold; + +#ifdef CONFIG_SND_HDA_DSP_LOADER + /* Allows dsp load to have sole access to the playback stream. */ + struct mutex dsp_mutex; +#endif +}; + +/* CORB/RIRB */ +struct azx_rb { + u32 *buf; /* CORB/RIRB buffer + * Each CORB entry is 4byte, RIRB is 8byte + */ + dma_addr_t addr; /* physical address of CORB/RIRB buffer */ + /* for RIRB */ + unsigned short rp, wp; /* read/write pointers */ + int cmds[AZX_MAX_CODECS]; /* number of pending requests */ + u32 res[AZX_MAX_CODECS]; /* last read value */ +}; + +struct azx; + +/* Functions to read/write to hda registers. */ +struct hda_controller_ops { + /* Register Access */ + void (*reg_writel)(u32 value, u32 __iomem *addr); + u32 (*reg_readl)(u32 __iomem *addr); + void (*reg_writew)(u16 value, u16 __iomem *addr); + u16 (*reg_readw)(u16 __iomem *addr); + void (*reg_writeb)(u8 value, u8 __iomem *addr); + u8 (*reg_readb)(u8 __iomem *addr); + /* Disable msi if supported, PCI only */ + int (*disable_msi_reset_irq)(struct azx *); + /* Allocation ops */ + int (*dma_alloc_pages)(struct azx *chip, + int type, + size_t size, + struct snd_dma_buffer *buf); + void (*dma_free_pages)(struct azx *chip, struct snd_dma_buffer *buf); + int (*substream_alloc_pages)(struct azx *chip, + struct snd_pcm_substream *substream, + size_t size); + int (*substream_free_pages)(struct azx *chip, + struct snd_pcm_substream *substream); + void (*pcm_mmap_prepare)(struct snd_pcm_substream *substream, + struct vm_area_struct *area); + /* Check if current position is acceptable */ + int (*position_check)(struct azx *chip, struct azx_dev *azx_dev); +}; + +struct azx_pcm { + struct azx *chip; + struct snd_pcm *pcm; + struct hda_codec *codec; + struct hda_pcm *info; + struct list_head list; +}; + +typedef unsigned int (*azx_get_pos_callback_t)(struct azx *, struct azx_dev *); +typedef int (*azx_get_delay_callback_t)(struct azx *, struct azx_dev *, unsigned int pos); + +struct azx { + struct snd_card *card; + struct pci_dev *pci; + int dev_index; + + /* chip type specific */ + int driver_type; + unsigned int driver_caps; + int playback_streams; + int playback_index_offset; + int capture_streams; + int capture_index_offset; + int num_streams; + const int *jackpoll_ms; /* per-card jack poll interval */ + + /* Register interaction. */ + const struct hda_controller_ops *ops; + + /* position adjustment callbacks */ + azx_get_pos_callback_t get_position[2]; + azx_get_delay_callback_t get_delay[2]; + + /* pci resources */ + unsigned long addr; + void __iomem *remap_addr; + int irq; + + /* locks */ + spinlock_t reg_lock; + struct mutex open_mutex; /* Prevents concurrent open/close operations */ + + /* streams (x num_streams) */ + struct azx_dev *azx_dev; + + /* PCM */ + struct list_head pcm_list; /* azx_pcm list */ + + /* HD codec */ + unsigned short codec_mask; + int codec_probe_mask; /* copied from probe_mask option */ + struct hda_bus *bus; + unsigned int beep_mode; + + /* CORB/RIRB */ + struct azx_rb corb; + struct azx_rb rirb; + + /* CORB/RIRB and position buffers */ + struct snd_dma_buffer rb; + struct snd_dma_buffer posbuf; + +#ifdef CONFIG_SND_HDA_PATCH_LOADER + const struct firmware *fw; +#endif + + /* flags */ + const int *bdl_pos_adj; + int poll_count; + unsigned int running:1; + unsigned int initialized:1; + unsigned int single_cmd:1; + unsigned int polling_mode:1; + unsigned int msi:1; + unsigned int probing:1; /* codec probing phase */ + unsigned int snoop:1; + unsigned int align_buffer_size:1; + unsigned int region_requested:1; + unsigned int disabled:1; /* disabled by VGA-switcher */ + + /* for debugging */ + unsigned int last_cmd[AZX_MAX_CODECS]; + +#ifdef CONFIG_SND_HDA_DSP_LOADER + struct azx_dev saved_azx_dev; +#endif +}; + +#ifdef CONFIG_X86 +#define azx_snoop(chip) ((chip)->snoop) +#else +#define azx_snoop(chip) true +#endif + +/* + * macros for easy use + */ + +#define azx_writel(chip, reg, value) \ + ((chip)->ops->reg_writel(value, (chip)->remap_addr + AZX_REG_##reg)) +#define azx_readl(chip, reg) \ + ((chip)->ops->reg_readl((chip)->remap_addr + AZX_REG_##reg)) +#define azx_writew(chip, reg, value) \ + ((chip)->ops->reg_writew(value, (chip)->remap_addr + AZX_REG_##reg)) +#define azx_readw(chip, reg) \ + ((chip)->ops->reg_readw((chip)->remap_addr + AZX_REG_##reg)) +#define azx_writeb(chip, reg, value) \ + ((chip)->ops->reg_writeb(value, (chip)->remap_addr + AZX_REG_##reg)) +#define azx_readb(chip, reg) \ + ((chip)->ops->reg_readb((chip)->remap_addr + AZX_REG_##reg)) + +#define azx_sd_writel(chip, dev, reg, value) \ + ((chip)->ops->reg_writel(value, (dev)->sd_addr + AZX_REG_##reg)) +#define azx_sd_readl(chip, dev, reg) \ + ((chip)->ops->reg_readl((dev)->sd_addr + AZX_REG_##reg)) +#define azx_sd_writew(chip, dev, reg, value) \ + ((chip)->ops->reg_writew(value, (dev)->sd_addr + AZX_REG_##reg)) +#define azx_sd_readw(chip, dev, reg) \ + ((chip)->ops->reg_readw((dev)->sd_addr + AZX_REG_##reg)) +#define azx_sd_writeb(chip, dev, reg, value) \ + ((chip)->ops->reg_writeb(value, (dev)->sd_addr + AZX_REG_##reg)) +#define azx_sd_readb(chip, dev, reg) \ + ((chip)->ops->reg_readb((dev)->sd_addr + AZX_REG_##reg)) + +#define azx_has_pm_runtime(chip) \ + (!AZX_DCAPS_PM_RUNTIME || ((chip)->driver_caps & AZX_DCAPS_PM_RUNTIME)) /* PCM setup */ static inline struct azx_dev *get_azx_dev(struct snd_pcm_substream *substream) @@ -43,14 +429,9 @@ void azx_enter_link_reset(struct azx *chip); irqreturn_t azx_interrupt(int irq, void *dev_id); /* Codec interface */ -int azx_codec_create(struct azx *chip, const char *model, - unsigned int max_slots, - int *power_save_to); +int azx_bus_create(struct azx *chip, const char *model); +int azx_probe_codecs(struct azx *chip, unsigned int max_slots); int azx_codec_configure(struct azx *chip); -int azx_mixer_create(struct azx *chip); int azx_init_stream(struct azx *chip); -void azx_notifier_register(struct azx *chip); -void azx_notifier_unregister(struct azx *chip); - #endif /* __SOUND_HDA_CONTROLLER_H */ diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index 8ec5289..0ef2459 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -140,6 +140,9 @@ static void parse_user_hints(struct hda_codec *codec) val = snd_hda_get_bool_hint(codec, "single_adc_amp"); if (val >= 0) codec->single_adc_amp = !!val; + val = snd_hda_get_bool_hint(codec, "power_save_node"); + if (val >= 0) + codec->power_save_node = !!val; val = snd_hda_get_bool_hint(codec, "auto_mute"); if (val >= 0) @@ -648,12 +651,24 @@ static bool is_active_nid(struct hda_codec *codec, hda_nid_t nid, unsigned int dir, unsigned int idx) { struct hda_gen_spec *spec = codec->spec; + int type = get_wcaps_type(get_wcaps(codec, nid)); int i, n; + if (nid == codec->afg) + return true; + for (n = 0; n < spec->paths.used; n++) { struct nid_path *path = snd_array_elem(&spec->paths, n); if (!path->active) continue; + if (codec->power_save_node) { + if (!path->stream_enabled) + continue; + /* ignore unplugged paths except for DAC/ADC */ + if (!(path->pin_enabled || path->pin_fixed) && + type != AC_WID_AUD_OUT && type != AC_WID_AUD_IN) + continue; + } for (i = 0; i < path->depth; i++) { if (path->path[i] == nid) { if (dir == HDA_OUTPUT || path->idx[i] == idx) @@ -807,6 +822,44 @@ static void activate_amp_in(struct hda_codec *codec, struct nid_path *path, } } +/* sync power of each widget in the the given path */ +static hda_nid_t path_power_update(struct hda_codec *codec, + struct nid_path *path, + bool allow_powerdown) +{ + hda_nid_t nid, changed = 0; + int i, state; + + for (i = 0; i < path->depth; i++) { + nid = path->path[i]; + if (nid == codec->afg) + continue; + if (!allow_powerdown || is_active_nid_for_any(codec, nid)) + state = AC_PWRST_D0; + else + state = AC_PWRST_D3; + if (!snd_hda_check_power_state(codec, nid, state)) { + snd_hda_codec_write(codec, nid, 0, + AC_VERB_SET_POWER_STATE, state); + changed = nid; + /* here we assume that widget attributes (e.g. amp, + * pinctl connection) don't change with local power + * state change. If not, need to sync the cache. + */ + } + } + return changed; +} + +/* do sync with the last power state change */ +static void sync_power_state_change(struct hda_codec *codec, hda_nid_t nid) +{ + if (nid) { + msleep(10); + snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_POWER_STATE, 0); + } +} + /** * snd_hda_activate_path - activate or deactivate the given path * @codec: the HDA codec @@ -825,15 +878,13 @@ void snd_hda_activate_path(struct hda_codec *codec, struct nid_path *path, if (!enable) path->active = false; + /* make sure the widget is powered up */ + if (enable && (spec->power_down_unused || codec->power_save_node)) + path_power_update(codec, path, codec->power_save_node); + for (i = path->depth - 1; i >= 0; i--) { hda_nid_t nid = path->path[i]; - if (enable && spec->power_down_unused) { - /* make sure the widget is powered up */ - if (!snd_hda_check_power_state(codec, nid, AC_PWRST_D0)) - snd_hda_codec_write(codec, nid, 0, - AC_VERB_SET_POWER_STATE, - AC_PWRST_D0); - } + if (enable && path->multi[i]) snd_hda_codec_update_cache(codec, nid, 0, AC_VERB_SET_CONNECT_SEL, @@ -853,28 +904,10 @@ EXPORT_SYMBOL_GPL(snd_hda_activate_path); static void path_power_down_sync(struct hda_codec *codec, struct nid_path *path) { struct hda_gen_spec *spec = codec->spec; - bool changed = false; - int i; - if (!spec->power_down_unused || path->active) + if (!(spec->power_down_unused || codec->power_save_node) || path->active) return; - - for (i = 0; i < path->depth; i++) { - hda_nid_t nid = path->path[i]; - if (!snd_hda_check_power_state(codec, nid, AC_PWRST_D3) && - !is_active_nid_for_any(codec, nid)) { - snd_hda_codec_write(codec, nid, 0, - AC_VERB_SET_POWER_STATE, - AC_PWRST_D3); - changed = true; - } - } - - if (changed) { - msleep(10); - snd_hda_codec_read(codec, path->path[0], 0, - AC_VERB_GET_POWER_STATE, 0); - } + sync_power_state_change(codec, path_power_update(codec, path, true)); } /* turn on/off EAPD on the given pin */ @@ -1574,6 +1607,7 @@ static int check_aamix_out_path(struct hda_codec *codec, int path_idx) return 0; /* print_nid_path(codec, "output-aamix", path); */ path->active = false; /* unused as default */ + path->pin_fixed = true; /* static route */ return snd_hda_get_path_idx(codec, path); } @@ -2998,6 +3032,7 @@ static int new_analog_input(struct hda_codec *codec, int input_idx, } path->active = true; + path->stream_enabled = true; /* no DAC/ADC involved */ err = add_loopback_list(spec, mix_nid, idx); if (err < 0) return err; @@ -3009,6 +3044,8 @@ static int new_analog_input(struct hda_codec *codec, int input_idx, if (path) { print_nid_path(codec, "loopback-merge", path); path->active = true; + path->pin_fixed = true; /* static route */ + path->stream_enabled = true; /* no DAC/ADC involved */ spec->loopback_merge_path = snd_hda_get_path_idx(codec, path); } @@ -3810,6 +3847,7 @@ static void parse_digital(struct hda_codec *codec) continue; print_nid_path(codec, "digout", path); path->active = true; + path->pin_fixed = true; /* no jack detection */ spec->digout_paths[i] = snd_hda_get_path_idx(codec, path); set_pin_target(codec, pin, PIN_OUT, false); if (!nums) { @@ -3837,6 +3875,7 @@ static void parse_digital(struct hda_codec *codec) if (path) { print_nid_path(codec, "digin", path); path->active = true; + path->pin_fixed = true; /* no jack */ spec->dig_in_nid = dig_nid; spec->digin_path = snd_hda_get_path_idx(codec, path); set_pin_target(codec, pin, PIN_IN, false); @@ -3896,6 +3935,229 @@ static int mux_select(struct hda_codec *codec, unsigned int adc_idx, return 1; } +/* power up/down widgets in the all paths that match with the given NID + * as terminals (either start- or endpoint) + * + * returns the last changed NID, or zero if unchanged. + */ +static hda_nid_t set_path_power(struct hda_codec *codec, hda_nid_t nid, + int pin_state, int stream_state) +{ + struct hda_gen_spec *spec = codec->spec; + hda_nid_t last, changed = 0; + struct nid_path *path; + int n; + + for (n = 0; n < spec->paths.used; n++) { + path = snd_array_elem(&spec->paths, n); + if (path->path[0] == nid || + path->path[path->depth - 1] == nid) { + bool pin_old = path->pin_enabled; + bool stream_old = path->stream_enabled; + + if (pin_state >= 0) + path->pin_enabled = pin_state; + if (stream_state >= 0) + path->stream_enabled = stream_state; + if ((!path->pin_fixed && path->pin_enabled != pin_old) + || path->stream_enabled != stream_old) { + last = path_power_update(codec, path, true); + if (last) + changed = last; + } + } + } + return changed; +} + +/* power up/down the paths of the given pin according to the jack state; + * power = 0/1 : only power up/down if it matches with the jack state, + * < 0 : force power up/down to follow the jack sate + * + * returns the last changed NID, or zero if unchanged. + */ +static hda_nid_t set_pin_power_jack(struct hda_codec *codec, hda_nid_t pin, + int power) +{ + bool on; + + if (!codec->power_save_node) + return 0; + + on = snd_hda_jack_detect_state(codec, pin) != HDA_JACK_NOT_PRESENT; + if (power >= 0 && on != power) + return 0; + return set_path_power(codec, pin, on, -1); +} + +static void pin_power_callback(struct hda_codec *codec, + struct hda_jack_callback *jack, + bool on) +{ + if (jack && jack->tbl->nid) + sync_power_state_change(codec, + set_pin_power_jack(codec, jack->tbl->nid, on)); +} + +/* callback only doing power up -- called at first */ +static void pin_power_up_callback(struct hda_codec *codec, + struct hda_jack_callback *jack) +{ + pin_power_callback(codec, jack, true); +} + +/* callback only doing power down -- called at last */ +static void pin_power_down_callback(struct hda_codec *codec, + struct hda_jack_callback *jack) +{ + pin_power_callback(codec, jack, false); +} + +/* set up the power up/down callbacks */ +static void add_pin_power_ctls(struct hda_codec *codec, int num_pins, + const hda_nid_t *pins, bool on) +{ + int i; + hda_jack_callback_fn cb = + on ? pin_power_up_callback : pin_power_down_callback; + + for (i = 0; i < num_pins && pins[i]; i++) { + if (is_jack_detectable(codec, pins[i])) + snd_hda_jack_detect_enable_callback(codec, pins[i], cb); + else + set_path_power(codec, pins[i], true, -1); + } +} + +/* enabled power callback to each available I/O pin with jack detections; + * the digital I/O pins are excluded because of the unreliable detectsion + */ +static void add_all_pin_power_ctls(struct hda_codec *codec, bool on) +{ + struct hda_gen_spec *spec = codec->spec; + struct auto_pin_cfg *cfg = &spec->autocfg; + int i; + + if (!codec->power_save_node) + return; + add_pin_power_ctls(codec, cfg->line_outs, cfg->line_out_pins, on); + if (cfg->line_out_type != AUTO_PIN_HP_OUT) + add_pin_power_ctls(codec, cfg->hp_outs, cfg->hp_pins, on); + if (cfg->line_out_type != AUTO_PIN_SPEAKER_OUT) + add_pin_power_ctls(codec, cfg->speaker_outs, cfg->speaker_pins, on); + for (i = 0; i < cfg->num_inputs; i++) + add_pin_power_ctls(codec, 1, &cfg->inputs[i].pin, on); +} + +/* sync path power up/down with the jack states of given pins */ +static void sync_pin_power_ctls(struct hda_codec *codec, int num_pins, + const hda_nid_t *pins) +{ + int i; + + for (i = 0; i < num_pins && pins[i]; i++) + if (is_jack_detectable(codec, pins[i])) + set_pin_power_jack(codec, pins[i], -1); +} + +/* sync path power up/down with pins; called at init and resume */ +static void sync_all_pin_power_ctls(struct hda_codec *codec) +{ + struct hda_gen_spec *spec = codec->spec; + struct auto_pin_cfg *cfg = &spec->autocfg; + int i; + + if (!codec->power_save_node) + return; + sync_pin_power_ctls(codec, cfg->line_outs, cfg->line_out_pins); + if (cfg->line_out_type != AUTO_PIN_HP_OUT) + sync_pin_power_ctls(codec, cfg->hp_outs, cfg->hp_pins); + if (cfg->line_out_type != AUTO_PIN_SPEAKER_OUT) + sync_pin_power_ctls(codec, cfg->speaker_outs, cfg->speaker_pins); + for (i = 0; i < cfg->num_inputs; i++) + sync_pin_power_ctls(codec, 1, &cfg->inputs[i].pin); +} + +/* add fake paths if not present yet */ +static int add_fake_paths(struct hda_codec *codec, hda_nid_t nid, + int num_pins, const hda_nid_t *pins) +{ + struct hda_gen_spec *spec = codec->spec; + struct nid_path *path; + int i; + + for (i = 0; i < num_pins; i++) { + if (!pins[i]) + break; + if (get_nid_path(codec, nid, pins[i], 0)) + continue; + path = snd_array_new(&spec->paths); + if (!path) + return -ENOMEM; + memset(path, 0, sizeof(*path)); + path->depth = 2; + path->path[0] = nid; + path->path[1] = pins[i]; + path->active = true; + } + return 0; +} + +/* create fake paths to all outputs from beep */ +static int add_fake_beep_paths(struct hda_codec *codec) +{ + struct hda_gen_spec *spec = codec->spec; + struct auto_pin_cfg *cfg = &spec->autocfg; + hda_nid_t nid = spec->beep_nid; + int err; + + if (!codec->power_save_node || !nid) + return 0; + err = add_fake_paths(codec, nid, cfg->line_outs, cfg->line_out_pins); + if (err < 0) + return err; + if (cfg->line_out_type != AUTO_PIN_HP_OUT) { + err = add_fake_paths(codec, nid, cfg->hp_outs, cfg->hp_pins); + if (err < 0) + return err; + } + if (cfg->line_out_type != AUTO_PIN_SPEAKER_OUT) { + err = add_fake_paths(codec, nid, cfg->speaker_outs, + cfg->speaker_pins); + if (err < 0) + return err; + } + return 0; +} + +/* power up/down beep widget and its output paths */ +static void beep_power_hook(struct hda_beep *beep, bool on) +{ + set_path_power(beep->codec, beep->nid, -1, on); +} + +/** + * snd_hda_gen_fix_pin_power - Fix the power of the given pin widget to D0 + * @codec: the HDA codec + * @pin: NID of pin to fix + */ +int snd_hda_gen_fix_pin_power(struct hda_codec *codec, hda_nid_t pin) +{ + struct hda_gen_spec *spec = codec->spec; + struct nid_path *path; + + path = snd_array_new(&spec->paths); + if (!path) + return -ENOMEM; + memset(path, 0, sizeof(*path)); + path->depth = 1; + path->path[0] = pin; + path->active = true; + path->pin_fixed = true; + path->stream_enabled = true; + return 0; +} +EXPORT_SYMBOL_GPL(snd_hda_gen_fix_pin_power); /* * Jack detections for HP auto-mute and mic-switch @@ -3933,6 +4195,10 @@ static void do_automute(struct hda_codec *codec, int num_pins, hda_nid_t *pins, if (!nid) break; + oldval = snd_hda_codec_get_pin_target(codec, nid); + if (oldval & PIN_IN) + continue; /* no mute for inputs */ + if (spec->auto_mute_via_amp) { struct nid_path *path; hda_nid_t mute_nid; @@ -3947,29 +4213,33 @@ static void do_automute(struct hda_codec *codec, int num_pins, hda_nid_t *pins, spec->mute_bits |= (1ULL << mute_nid); else spec->mute_bits &= ~(1ULL << mute_nid); - set_pin_eapd(codec, nid, !mute); continue; + } else { + /* don't reset VREF value in case it's controlling + * the amp (see alc861_fixup_asus_amp_vref_0f()) + */ + if (spec->keep_vref_in_automute) + val = oldval & ~PIN_HP; + else + val = 0; + if (!mute) + val |= oldval; + /* here we call update_pin_ctl() so that the pinctl is + * changed without changing the pinctl target value; + * the original target value will be still referred at + * the init / resume again + */ + update_pin_ctl(codec, nid, val); } - oldval = snd_hda_codec_get_pin_target(codec, nid); - if (oldval & PIN_IN) - continue; /* no mute for inputs */ - /* don't reset VREF value in case it's controlling - * the amp (see alc861_fixup_asus_amp_vref_0f()) - */ - if (spec->keep_vref_in_automute) - val = oldval & ~PIN_HP; - else - val = 0; - if (!mute) - val |= oldval; - /* here we call update_pin_ctl() so that the pinctl is changed - * without changing the pinctl target value; - * the original target value will be still referred at the - * init / resume again - */ - update_pin_ctl(codec, nid, val); set_pin_eapd(codec, nid, !mute); + if (codec->power_save_node) { + bool on = !mute; + if (on) + on = snd_hda_jack_detect_state(codec, nid) + != HDA_JACK_NOT_PRESENT; + set_path_power(codec, nid, on, -1); + } } } @@ -4466,6 +4736,21 @@ static void mute_all_mixer_nid(struct hda_codec *codec, hda_nid_t mix) } /** + * snd_hda_gen_stream_pm - Stream power management callback + * @codec: the HDA codec + * @nid: audio widget + * @on: power on/off flag + * + * Set this in patch_ops.stream_pm. Only valid with power_save_node flag. + */ +void snd_hda_gen_stream_pm(struct hda_codec *codec, hda_nid_t nid, bool on) +{ + if (codec->power_save_node) + set_path_power(codec, nid, -1, on); +} +EXPORT_SYMBOL_GPL(snd_hda_gen_stream_pm); + +/** * snd_hda_gen_parse_auto_config - Parse the given BIOS configuration and * set up the hda_gen_spec * @codec: the HDA codec @@ -4549,6 +4834,9 @@ int snd_hda_gen_parse_auto_config(struct hda_codec *codec, if (err < 0) return err; + /* add power-down pin callbacks at first */ + add_all_pin_power_ctls(codec, false); + spec->const_channel_count = spec->ext_channel_count; /* check the multiple speaker and headphone pins */ if (cfg->line_out_type != AUTO_PIN_SPEAKER_OUT) @@ -4618,6 +4906,9 @@ int snd_hda_gen_parse_auto_config(struct hda_codec *codec, } } + /* add power-up pin callbacks at last */ + add_all_pin_power_ctls(codec, true); + /* mute all aamix input initially */ if (spec->mixer_nid) mute_all_mixer_nid(codec, spec->mixer_nid); @@ -4625,13 +4916,19 @@ int snd_hda_gen_parse_auto_config(struct hda_codec *codec, dig_only: parse_digital(codec); - if (spec->power_down_unused) + if (spec->power_down_unused || codec->power_save_node) codec->power_filter = snd_hda_gen_path_power_filter; if (!spec->no_analog && spec->beep_nid) { err = snd_hda_attach_beep_device(codec, spec->beep_nid); if (err < 0) return err; + if (codec->beep && codec->power_save_node) { + err = add_fake_beep_paths(codec); + if (err < 0) + return err; + codec->beep->power_hook = beep_power_hook; + } } return 1; @@ -4675,7 +4972,7 @@ int snd_hda_gen_build_controls(struct hda_codec *codec) err = snd_hda_create_dig_out_ctls(codec, spec->multiout.dig_out_nid, spec->multiout.dig_out_nid, - spec->pcm_rec[1].pcm_type); + spec->pcm_rec[1]->pcm_type); if (err < 0) return err; if (!spec->no_analog) { @@ -5137,6 +5434,33 @@ static void fill_pcm_stream_name(char *str, size_t len, const char *sfx, strlcat(str, sfx, len); } +/* copy PCM stream info from @default_str, and override non-NULL entries + * from @spec_str and @nid + */ +static void setup_pcm_stream(struct hda_pcm_stream *str, + const struct hda_pcm_stream *default_str, + const struct hda_pcm_stream *spec_str, + hda_nid_t nid) +{ + *str = *default_str; + if (nid) + str->nid = nid; + if (spec_str) { + if (spec_str->substreams) + str->substreams = spec_str->substreams; + if (spec_str->channels_min) + str->channels_min = spec_str->channels_min; + if (spec_str->channels_max) + str->channels_max = spec_str->channels_max; + if (spec_str->rates) + str->rates = spec_str->rates; + if (spec_str->formats) + str->formats = spec_str->formats; + if (spec_str->maxbps) + str->maxbps = spec_str->maxbps; + } +} + /** * snd_hda_gen_build_pcms - build PCM streams based on the parsed results * @codec: the HDA codec @@ -5146,27 +5470,25 @@ static void fill_pcm_stream_name(char *str, size_t len, const char *sfx, int snd_hda_gen_build_pcms(struct hda_codec *codec) { struct hda_gen_spec *spec = codec->spec; - struct hda_pcm *info = spec->pcm_rec; - const struct hda_pcm_stream *p; + struct hda_pcm *info; bool have_multi_adcs; - codec->num_pcms = 1; - codec->pcm_info = info; - if (spec->no_analog) goto skip_analog; fill_pcm_stream_name(spec->stream_name_analog, sizeof(spec->stream_name_analog), " Analog", codec->chip_name); - info->name = spec->stream_name_analog; + info = snd_hda_codec_pcm_new(codec, "%s", spec->stream_name_analog); + if (!info) + return -ENOMEM; + spec->pcm_rec[0] = info; if (spec->multiout.num_dacs > 0) { - p = spec->stream_analog_playback; - if (!p) - p = &pcm_analog_playback; - info->stream[SNDRV_PCM_STREAM_PLAYBACK] = *p; - info->stream[SNDRV_PCM_STREAM_PLAYBACK].nid = spec->multiout.dac_nids[0]; + setup_pcm_stream(&info->stream[SNDRV_PCM_STREAM_PLAYBACK], + &pcm_analog_playback, + spec->stream_analog_playback, + spec->multiout.dac_nids[0]); info->stream[SNDRV_PCM_STREAM_PLAYBACK].channels_max = spec->multiout.max_channels; if (spec->autocfg.line_out_type == AUTO_PIN_SPEAKER_OUT && @@ -5175,15 +5497,11 @@ int snd_hda_gen_build_pcms(struct hda_codec *codec) snd_pcm_2_1_chmaps; } if (spec->num_adc_nids) { - p = spec->stream_analog_capture; - if (!p) { - if (spec->dyn_adc_switch) - p = &dyn_adc_pcm_analog_capture; - else - p = &pcm_analog_capture; - } - info->stream[SNDRV_PCM_STREAM_CAPTURE] = *p; - info->stream[SNDRV_PCM_STREAM_CAPTURE].nid = spec->adc_nids[0]; + setup_pcm_stream(&info->stream[SNDRV_PCM_STREAM_CAPTURE], + (spec->dyn_adc_switch ? + &dyn_adc_pcm_analog_capture : &pcm_analog_capture), + spec->stream_analog_capture, + spec->adc_nids[0]); } skip_analog: @@ -5192,28 +5510,26 @@ int snd_hda_gen_build_pcms(struct hda_codec *codec) fill_pcm_stream_name(spec->stream_name_digital, sizeof(spec->stream_name_digital), " Digital", codec->chip_name); - codec->num_pcms = 2; + info = snd_hda_codec_pcm_new(codec, "%s", + spec->stream_name_digital); + if (!info) + return -ENOMEM; codec->slave_dig_outs = spec->multiout.slave_dig_outs; - info = spec->pcm_rec + 1; - info->name = spec->stream_name_digital; + spec->pcm_rec[1] = info; if (spec->dig_out_type) info->pcm_type = spec->dig_out_type; else info->pcm_type = HDA_PCM_TYPE_SPDIF; - if (spec->multiout.dig_out_nid) { - p = spec->stream_digital_playback; - if (!p) - p = &pcm_digital_playback; - info->stream[SNDRV_PCM_STREAM_PLAYBACK] = *p; - info->stream[SNDRV_PCM_STREAM_PLAYBACK].nid = spec->multiout.dig_out_nid; - } - if (spec->dig_in_nid) { - p = spec->stream_digital_capture; - if (!p) - p = &pcm_digital_capture; - info->stream[SNDRV_PCM_STREAM_CAPTURE] = *p; - info->stream[SNDRV_PCM_STREAM_CAPTURE].nid = spec->dig_in_nid; - } + if (spec->multiout.dig_out_nid) + setup_pcm_stream(&info->stream[SNDRV_PCM_STREAM_PLAYBACK], + &pcm_digital_playback, + spec->stream_digital_playback, + spec->multiout.dig_out_nid); + if (spec->dig_in_nid) + setup_pcm_stream(&info->stream[SNDRV_PCM_STREAM_CAPTURE], + &pcm_digital_capture, + spec->stream_digital_capture, + spec->dig_in_nid); } if (spec->no_analog) @@ -5229,34 +5545,29 @@ int snd_hda_gen_build_pcms(struct hda_codec *codec) fill_pcm_stream_name(spec->stream_name_alt_analog, sizeof(spec->stream_name_alt_analog), " Alt Analog", codec->chip_name); - codec->num_pcms = 3; - info = spec->pcm_rec + 2; - info->name = spec->stream_name_alt_analog; - if (spec->alt_dac_nid) { - p = spec->stream_analog_alt_playback; - if (!p) - p = &pcm_analog_alt_playback; - info->stream[SNDRV_PCM_STREAM_PLAYBACK] = *p; - info->stream[SNDRV_PCM_STREAM_PLAYBACK].nid = - spec->alt_dac_nid; - } else { - info->stream[SNDRV_PCM_STREAM_PLAYBACK] = - pcm_null_stream; - info->stream[SNDRV_PCM_STREAM_PLAYBACK].nid = 0; - } + info = snd_hda_codec_pcm_new(codec, "%s", + spec->stream_name_alt_analog); + if (!info) + return -ENOMEM; + spec->pcm_rec[2] = info; + if (spec->alt_dac_nid) + setup_pcm_stream(&info->stream[SNDRV_PCM_STREAM_PLAYBACK], + &pcm_analog_alt_playback, + spec->stream_analog_alt_playback, + spec->alt_dac_nid); + else + setup_pcm_stream(&info->stream[SNDRV_PCM_STREAM_PLAYBACK], + &pcm_null_stream, NULL, 0); if (have_multi_adcs) { - p = spec->stream_analog_alt_capture; - if (!p) - p = &pcm_analog_alt_capture; - info->stream[SNDRV_PCM_STREAM_CAPTURE] = *p; - info->stream[SNDRV_PCM_STREAM_CAPTURE].nid = - spec->adc_nids[1]; + setup_pcm_stream(&info->stream[SNDRV_PCM_STREAM_CAPTURE], + &pcm_analog_alt_capture, + spec->stream_analog_alt_capture, + spec->adc_nids[1]); info->stream[SNDRV_PCM_STREAM_CAPTURE].substreams = spec->num_adc_nids - 1; } else { - info->stream[SNDRV_PCM_STREAM_CAPTURE] = - pcm_null_stream; - info->stream[SNDRV_PCM_STREAM_CAPTURE].nid = 0; + setup_pcm_stream(&info->stream[SNDRV_PCM_STREAM_CAPTURE], + &pcm_null_stream, NULL, 0); } } @@ -5464,6 +5775,8 @@ int snd_hda_gen_init(struct hda_codec *codec) clear_unsol_on_unused_pins(codec); + sync_all_pin_power_ctls(codec); + /* call init functions of standard auto-mute helpers */ update_automute_all(codec); @@ -5524,13 +5837,11 @@ static const struct hda_codec_ops generic_patch_ops = { #endif }; -/** +/* * snd_hda_parse_generic_codec - Generic codec parser * @codec: the HDA codec - * - * This should be called from the HDA codec core. */ -int snd_hda_parse_generic_codec(struct hda_codec *codec) +static int snd_hda_parse_generic_codec(struct hda_codec *codec) { struct hda_gen_spec *spec; int err; @@ -5556,7 +5867,17 @@ error: snd_hda_gen_free(codec); return err; } -EXPORT_SYMBOL_GPL(snd_hda_parse_generic_codec); + +static const struct hda_codec_preset snd_hda_preset_generic[] = { + { .id = HDA_CODEC_ID_GENERIC, .patch = snd_hda_parse_generic_codec }, + {} /* terminator */ +}; + +static struct hda_codec_driver generic_driver = { + .preset = snd_hda_preset_generic, +}; + +module_hda_codec_driver(generic_driver); MODULE_LICENSE("GPL"); MODULE_DESCRIPTION("Generic HD-audio codec parser"); diff --git a/sound/pci/hda/hda_generic.h b/sound/pci/hda/hda_generic.h index 3d85266..56e4139 100644 --- a/sound/pci/hda/hda_generic.h +++ b/sound/pci/hda/hda_generic.h @@ -46,7 +46,10 @@ struct nid_path { unsigned char idx[MAX_NID_PATH_DEPTH]; unsigned char multi[MAX_NID_PATH_DEPTH]; unsigned int ctls[NID_PATH_NUM_CTLS]; /* NID_PATH_XXX_CTL */ - bool active; + bool active:1; /* activated by driver */ + bool pin_enabled:1; /* pins are enabled */ + bool pin_fixed:1; /* path with fixed pin */ + bool stream_enabled:1; /* stream is active */ }; /* mic/line-in auto switching entry */ @@ -144,7 +147,7 @@ struct hda_gen_spec { int const_channel_count; /* channel count for all */ /* PCM information */ - struct hda_pcm pcm_rec[3]; /* used in build_pcms() */ + struct hda_pcm *pcm_rec[3]; /* used in build_pcms() */ /* dynamic controls, init_verbs and input_mux */ struct auto_pin_cfg autocfg; @@ -340,5 +343,7 @@ int snd_hda_gen_check_power_status(struct hda_codec *codec, hda_nid_t nid); unsigned int snd_hda_gen_path_power_filter(struct hda_codec *codec, hda_nid_t nid, unsigned int power_state); +void snd_hda_gen_stream_pm(struct hda_codec *codec, hda_nid_t nid, bool on); +int snd_hda_gen_fix_pin_power(struct hda_codec *codec, hda_nid_t pin); #endif /* __SOUND_HDA_GENERIC_H */ diff --git a/sound/pci/hda/hda_hwdep.c b/sound/pci/hda/hda_hwdep.c index 11b5a42..57df06e 100644 --- a/sound/pci/hda/hda_hwdep.c +++ b/sound/pci/hda/hda_hwdep.c @@ -101,7 +101,7 @@ int snd_hda_create_hwdep(struct hda_codec *codec) int err; sprintf(hwname, "HDA Codec %d", codec->addr); - err = snd_hwdep_new(codec->bus->card, hwname, codec->addr, &hwdep); + err = snd_hwdep_new(codec->card, hwname, codec->addr, &hwdep); if (err < 0) return err; codec->hwdep = hwdep; @@ -116,9 +116,6 @@ int snd_hda_create_hwdep(struct hda_codec *codec) hwdep->ops.ioctl_compat = hda_hwdep_ioctl_compat; #endif - /* link to codec */ - hwdep->dev.parent = &codec->dev; - /* for sysfs */ hwdep->dev.groups = snd_hda_dev_attr_groups; dev_set_drvdata(&hwdep->dev, codec); diff --git a/sound/pci/hda/hda_i915.c b/sound/pci/hda/hda_i915.c index 7148945..52a85d8 100644 --- a/sound/pci/hda/hda_i915.c +++ b/sound/pci/hda/hda_i915.c @@ -22,7 +22,7 @@ #include <linux/component.h> #include <drm/i915_component.h> #include <sound/core.h> -#include "hda_priv.h" +#include "hda_controller.h" #include "hda_intel.h" /* Intel HSW/BDW display HDA controller Extended Mode registers. diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 4ca3d5d..060f7a2 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -62,7 +62,6 @@ #include <linux/firmware.h> #include "hda_codec.h" #include "hda_controller.h" -#include "hda_priv.h" #include "hda_intel.h" /* position fix mode */ @@ -174,7 +173,6 @@ static struct kernel_param_ops param_ops_xint = { #define param_check_xint param_check_int static int power_save = CONFIG_SND_HDA_POWER_SAVE_DEFAULT; -static int *power_save_addr = &power_save; module_param(power_save, xint, 0644); MODULE_PARM_DESC(power_save, "Automatic power-saving timeout " "(in second, 0 = disable)."); @@ -187,7 +185,7 @@ static bool power_save_controller = 1; module_param(power_save_controller, bool, 0644); MODULE_PARM_DESC(power_save_controller, "Reset controller in power save mode."); #else -static int *power_save_addr; +#define power_save 0 #endif /* CONFIG_PM */ static int align_buffer_size = -1; @@ -530,10 +528,10 @@ static int azx_position_check(struct azx *chip, struct azx_dev *azx_dev) if (ok == 1) { azx_dev->irq_pending = 0; return ok; - } else if (ok == 0 && chip->bus && chip->bus->workq) { + } else if (ok == 0) { /* bogus IRQ, process it later */ azx_dev->irq_pending = 1; - queue_work(chip->bus->workq, &hda->irq_pending_work); + schedule_work(&hda->irq_pending_work); } return 0; } @@ -741,7 +739,6 @@ static int param_set_xint(const char *val, const struct kernel_param *kp) { struct hda_intel *hda; struct azx *chip; - struct hda_codec *c; int prev = power_save; int ret = param_set_int(val, kp); @@ -753,8 +750,7 @@ static int param_set_xint(const char *val, const struct kernel_param *kp) chip = &hda->chip; if (!chip->bus || chip->disabled) continue; - list_for_each_entry(c, &chip->bus->codec_list, list) - snd_hda_power_sync(c); + snd_hda_set_power_save(chip->bus, power_save * 1000); } mutex_unlock(&card_list_lock); return 0; @@ -773,7 +769,6 @@ static int azx_suspend(struct device *dev) struct snd_card *card = dev_get_drvdata(dev); struct azx *chip; struct hda_intel *hda; - struct azx_pcm *p; if (!card) return 0; @@ -785,10 +780,6 @@ static int azx_suspend(struct device *dev) snd_power_change_state(card, SNDRV_CTL_POWER_D3hot); azx_clear_irq_pending(chip); - list_for_each_entry(p, &chip->pcm_list, list) - snd_pcm_suspend_all(p->pcm); - if (chip->initialized) - snd_hda_suspend(chip->bus); azx_stop_chip(chip); azx_enter_link_reset(chip); if (chip->irq >= 0) { @@ -831,7 +822,6 @@ static int azx_resume(struct device *dev) azx_init_chip(chip, true); - snd_hda_resume(chip->bus); snd_power_change_state(card, SNDRV_CTL_POWER_D0); return 0; } @@ -852,7 +842,7 @@ static int azx_runtime_suspend(struct device *dev) if (chip->disabled || hda->init_failed) return 0; - if (!(chip->driver_caps & AZX_DCAPS_PM_RUNTIME)) + if (!azx_has_pm_runtime(chip)) return 0; /* enable controller wake up event */ @@ -885,7 +875,7 @@ static int azx_runtime_resume(struct device *dev) if (chip->disabled || hda->init_failed) return 0; - if (!(chip->driver_caps & AZX_DCAPS_PM_RUNTIME)) + if (!azx_has_pm_runtime(chip)) return 0; if (chip->driver_caps & AZX_DCAPS_I915_POWERWELL) { @@ -903,8 +893,8 @@ static int azx_runtime_resume(struct device *dev) if (status && bus) { list_for_each_entry(codec, &bus->codec_list, list) if (status & (1 << codec->addr)) - queue_delayed_work(codec->bus->workq, - &codec->jackpoll_work, codec->jackpoll_interval); + schedule_delayed_work(&codec->jackpoll_work, + codec->jackpoll_interval); } /* disable controller Wake Up event*/ @@ -928,8 +918,8 @@ static int azx_runtime_idle(struct device *dev) if (chip->disabled || hda->init_failed) return 0; - if (!power_save_controller || - !(chip->driver_caps & AZX_DCAPS_PM_RUNTIME)) + if (!power_save_controller || !azx_has_pm_runtime(chip) || + chip->bus->codec_powered) return -EBUSY; return 0; @@ -1071,14 +1061,11 @@ static int azx_free(struct azx *chip) struct hda_intel *hda = container_of(chip, struct hda_intel, chip); int i; - if ((chip->driver_caps & AZX_DCAPS_PM_RUNTIME) - && chip->running) + if (azx_has_pm_runtime(chip) && chip->running) pm_runtime_get_noresume(&pci->dev); azx_del_card_list(chip); - azx_notifier_unregister(chip); - hda->init_failed = 1; /* to be sure */ complete_all(&hda->probe_wait); @@ -1394,7 +1381,6 @@ static int azx_create(struct snd_card *card, struct pci_dev *pci, hda = kzalloc(sizeof(*hda), GFP_KERNEL); if (!hda) { - dev_err(card->dev, "Cannot allocate hda\n"); pci_disable_device(pci); return -ENOMEM; } @@ -1575,10 +1561,8 @@ static int azx_first_init(struct azx *chip) chip->num_streams = chip->playback_streams + chip->capture_streams; chip->azx_dev = kcalloc(chip->num_streams, sizeof(*chip->azx_dev), GFP_KERNEL); - if (!chip->azx_dev) { - dev_err(card->dev, "cannot malloc azx_dev\n"); + if (!chip->azx_dev) return -ENOMEM; - } err = azx_alloc_stream_pages(chip); if (err < 0) @@ -1615,19 +1599,6 @@ static int azx_first_init(struct azx *chip) return 0; } -static void power_down_all_codecs(struct azx *chip) -{ -#ifdef CONFIG_PM - /* The codecs were powered up in snd_hda_codec_new(). - * Now all initialization done, so turn them down if possible - */ - struct hda_codec *codec; - list_for_each_entry(codec, &chip->bus->codec_list, list) { - snd_hda_power_down(codec); - } -#endif -} - #ifdef CONFIG_SND_HDA_PATCH_LOADER /* callback from request_firmware_nowait() */ static void azx_firmware_cb(const struct firmware *fw, void *context) @@ -1896,12 +1867,14 @@ static int azx_probe_continue(struct azx *chip) #endif /* create codec instances */ - err = azx_codec_create(chip, model[dev], - azx_max_codecs[chip->driver_type], - power_save_addr); + err = azx_bus_create(chip, model[dev]); + if (err < 0) + goto out_free; + err = azx_probe_codecs(chip, azx_max_codecs[chip->driver_type]); if (err < 0) goto out_free; + #ifdef CONFIG_SND_HDA_PATCH_LOADER if (chip->fw) { err = snd_hda_load_patch(chip->bus, chip->fw->size, @@ -1920,25 +1893,14 @@ static int azx_probe_continue(struct azx *chip) goto out_free; } - /* create PCM streams */ - err = snd_hda_build_pcms(chip->bus); - if (err < 0) - goto out_free; - - /* create mixer controls */ - err = azx_mixer_create(chip); - if (err < 0) - goto out_free; - err = snd_card_register(chip->card); if (err < 0) goto out_free; chip->running = 1; - power_down_all_codecs(chip); - azx_notifier_register(chip); azx_add_card_list(chip); - if ((chip->driver_caps & AZX_DCAPS_PM_RUNTIME) || hda->use_vga_switcheroo) + snd_hda_set_power_save(chip->bus, power_save * 1000); + if (azx_has_pm_runtime(chip) || hda->use_vga_switcheroo) pm_runtime_put_noidle(&pci->dev); out_free: @@ -1956,6 +1918,18 @@ static void azx_remove(struct pci_dev *pci) snd_card_free(card); } +static void azx_shutdown(struct pci_dev *pci) +{ + struct snd_card *card = pci_get_drvdata(pci); + struct azx *chip; + + if (!card) + return; + chip = card->private_data; + if (chip && chip->running) + azx_stop_chip(chip); +} + /* PCI IDs */ static const struct pci_device_id azx_ids[] = { /* CPT */ @@ -2178,6 +2152,7 @@ static struct pci_driver azx_driver = { .id_table = azx_ids, .probe = azx_probe, .remove = azx_remove, + .shutdown = azx_shutdown, .driver = { .pm = AZX_PM_OPS, }, diff --git a/sound/pci/hda/hda_intel.h b/sound/pci/hda/hda_intel.h index 3486118..d5231f7 100644 --- a/sound/pci/hda/hda_intel.h +++ b/sound/pci/hda/hda_intel.h @@ -17,7 +17,7 @@ #define __SOUND_HDA_INTEL_H #include <drm/i915_component.h> -#include "hda_priv.h" +#include "hda_controller.h" struct hda_intel { struct azx chip; diff --git a/sound/pci/hda/hda_jack.c b/sound/pci/hda/hda_jack.c index e664307..d7cfe7b 100644 --- a/sound/pci/hda/hda_jack.c +++ b/sound/pci/hda/hda_jack.c @@ -135,7 +135,7 @@ void snd_hda_jack_tbl_clear(struct hda_codec *codec) #ifdef CONFIG_SND_HDA_INPUT_JACK /* free jack instances manually when clearing/reconfiguring */ if (!codec->bus->shutdown && jack->jack) - snd_device_free(codec->bus->card, jack->jack); + snd_device_free(codec->card, jack->jack); #endif for (cb = jack->callback; cb; cb = next) { next = cb->next; @@ -340,7 +340,7 @@ void snd_hda_jack_report_sync(struct hda_codec *codec) if (!jack->kctl || jack->block_report) continue; state = get_jack_plug_state(jack->pin_sense); - snd_kctl_jack_report(codec->bus->card, jack->kctl, state); + snd_kctl_jack_report(codec->card, jack->kctl, state); #ifdef CONFIG_SND_HDA_INPUT_JACK if (jack->jack) snd_jack_report(jack->jack, @@ -412,11 +412,11 @@ static int __snd_hda_jack_add_kctl(struct hda_codec *codec, hda_nid_t nid, jack->phantom_jack = !!phantom_jack; state = snd_hda_jack_detect(codec, nid); - snd_kctl_jack_report(codec->bus->card, kctl, state); + snd_kctl_jack_report(codec->card, kctl, state); #ifdef CONFIG_SND_HDA_INPUT_JACK if (!phantom_jack) { jack->type = get_input_jack_type(codec, nid); - err = snd_jack_new(codec->bus->card, name, jack->type, + err = snd_jack_new(codec->card, name, jack->type, &jack->jack); if (err < 0) return err; diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h index 62658f2..1d00164 100644 --- a/sound/pci/hda/hda_local.h +++ b/sound/pci/hda/hda_local.h @@ -150,6 +150,7 @@ int __snd_hda_add_vmaster(struct hda_codec *codec, char *name, #define snd_hda_add_vmaster(codec, name, tlv, slaves, suffix) \ __snd_hda_add_vmaster(codec, name, tlv, slaves, suffix, true, NULL) int snd_hda_codec_reset(struct hda_codec *codec); +void snd_hda_codec_cleanup_for_unbind(struct hda_codec *codec); enum { HDA_VMUTE_OFF, @@ -273,29 +274,6 @@ int snd_hda_add_imux_item(struct hda_codec *codec, int index, int *type_index_ret); /* - * Channel mode helper - */ -struct hda_channel_mode { - int channels; - const struct hda_verb *sequence; -}; - -int snd_hda_ch_mode_info(struct hda_codec *codec, - struct snd_ctl_elem_info *uinfo, - const struct hda_channel_mode *chmode, - int num_chmodes); -int snd_hda_ch_mode_get(struct hda_codec *codec, - struct snd_ctl_elem_value *ucontrol, - const struct hda_channel_mode *chmode, - int num_chmodes, - int max_channels); -int snd_hda_ch_mode_put(struct hda_codec *codec, - struct snd_ctl_elem_value *ucontrol, - const struct hda_channel_mode *chmode, - int num_chmodes, - int *max_channelsp); - -/* * Multi-channel / digital-out PCM helper */ @@ -351,12 +329,6 @@ int snd_hda_multi_out_analog_cleanup(struct hda_codec *codec, struct hda_multi_out *mout); /* - * generic codec parser - */ -int snd_hda_parse_generic_codec(struct hda_codec *codec); -int snd_hda_parse_hdmi_codec(struct hda_codec *codec); - -/* * generic proc interface */ #ifdef CONFIG_PROC_FS @@ -466,23 +438,6 @@ void snd_hda_pick_pin_fixup(struct hda_codec *codec, const struct snd_hda_pin_quirk *pin_quirk, const struct hda_fixup *fixlist); - -/* - * unsolicited event handler - */ - -#define HDA_UNSOL_QUEUE_SIZE 64 - -struct hda_bus_unsolicited { - /* ring buffer */ - u32 queue[HDA_UNSOL_QUEUE_SIZE * 2]; - unsigned int rp, wp; - - /* workqueue */ - struct work_struct work; - struct hda_bus *bus; -}; - /* helper macros to retrieve pin default-config values */ #define get_defcfg_connect(cfg) \ ((cfg & AC_DEFCFG_PORT_CONN) >> AC_DEFCFG_PORT_CONN_SHIFT) @@ -800,9 +755,13 @@ void snd_print_channel_allocation(int spk_alloc, char *buf, int buflen); /* */ -#define codec_err(codec, fmt, args...) dev_err(&(codec)->dev, fmt, ##args) -#define codec_warn(codec, fmt, args...) dev_warn(&(codec)->dev, fmt, ##args) -#define codec_info(codec, fmt, args...) dev_info(&(codec)->dev, fmt, ##args) -#define codec_dbg(codec, fmt, args...) dev_dbg(&(codec)->dev, fmt, ##args) +#define codec_err(codec, fmt, args...) \ + dev_err(hda_codec_dev(codec), fmt, ##args) +#define codec_warn(codec, fmt, args...) \ + dev_warn(hda_codec_dev(codec), fmt, ##args) +#define codec_info(codec, fmt, args...) \ + dev_info(hda_codec_dev(codec), fmt, ##args) +#define codec_dbg(codec, fmt, args...) \ + dev_dbg(hda_codec_dev(codec), fmt, ##args) #endif /* __SOUND_HDA_LOCAL_H */ diff --git a/sound/pci/hda/hda_priv.h b/sound/pci/hda/hda_priv.h deleted file mode 100644 index daf4582..0000000 --- a/sound/pci/hda/hda_priv.h +++ /dev/null @@ -1,406 +0,0 @@ -/* - * Common defines for the alsa driver code base for HD Audio. - * - * This program is free software; you can redistribute it and/or modify it - * under the terms of the GNU General Public License as published by the Free - * Software Foundation; either version 2 of the License, or (at your option) - * any later version. - * - * This program is distributed in the hope that it will be useful, but WITHOUT - * ANY WARRANTY; without even the implied warranty of MERCHANTABILITY or - * FITNESS FOR A PARTICULAR PURPOSE. See the GNU General Public License for - * more details. - */ - -#ifndef __SOUND_HDA_PRIV_H -#define __SOUND_HDA_PRIV_H - -#include <linux/timecounter.h> -#include <sound/core.h> -#include <sound/pcm.h> - -/* - * registers - */ -#define AZX_REG_GCAP 0x00 -#define AZX_GCAP_64OK (1 << 0) /* 64bit address support */ -#define AZX_GCAP_NSDO (3 << 1) /* # of serial data out signals */ -#define AZX_GCAP_BSS (31 << 3) /* # of bidirectional streams */ -#define AZX_GCAP_ISS (15 << 8) /* # of input streams */ -#define AZX_GCAP_OSS (15 << 12) /* # of output streams */ -#define AZX_REG_VMIN 0x02 -#define AZX_REG_VMAJ 0x03 -#define AZX_REG_OUTPAY 0x04 -#define AZX_REG_INPAY 0x06 -#define AZX_REG_GCTL 0x08 -#define AZX_GCTL_RESET (1 << 0) /* controller reset */ -#define AZX_GCTL_FCNTRL (1 << 1) /* flush control */ -#define AZX_GCTL_UNSOL (1 << 8) /* accept unsol. response enable */ -#define AZX_REG_WAKEEN 0x0c -#define AZX_REG_STATESTS 0x0e -#define AZX_REG_GSTS 0x10 -#define AZX_GSTS_FSTS (1 << 1) /* flush status */ -#define AZX_REG_INTCTL 0x20 -#define AZX_REG_INTSTS 0x24 -#define AZX_REG_WALLCLK 0x30 /* 24Mhz source */ -#define AZX_REG_OLD_SSYNC 0x34 /* SSYNC for old ICH */ -#define AZX_REG_SSYNC 0x38 -#define AZX_REG_CORBLBASE 0x40 -#define AZX_REG_CORBUBASE 0x44 -#define AZX_REG_CORBWP 0x48 -#define AZX_REG_CORBRP 0x4a -#define AZX_CORBRP_RST (1 << 15) /* read pointer reset */ -#define AZX_REG_CORBCTL 0x4c -#define AZX_CORBCTL_RUN (1 << 1) /* enable DMA */ -#define AZX_CORBCTL_CMEIE (1 << 0) /* enable memory error irq */ -#define AZX_REG_CORBSTS 0x4d -#define AZX_CORBSTS_CMEI (1 << 0) /* memory error indication */ -#define AZX_REG_CORBSIZE 0x4e - -#define AZX_REG_RIRBLBASE 0x50 -#define AZX_REG_RIRBUBASE 0x54 -#define AZX_REG_RIRBWP 0x58 -#define AZX_RIRBWP_RST (1 << 15) /* write pointer reset */ -#define AZX_REG_RINTCNT 0x5a -#define AZX_REG_RIRBCTL 0x5c -#define AZX_RBCTL_IRQ_EN (1 << 0) /* enable IRQ */ -#define AZX_RBCTL_DMA_EN (1 << 1) /* enable DMA */ -#define AZX_RBCTL_OVERRUN_EN (1 << 2) /* enable overrun irq */ -#define AZX_REG_RIRBSTS 0x5d -#define AZX_RBSTS_IRQ (1 << 0) /* response irq */ -#define AZX_RBSTS_OVERRUN (1 << 2) /* overrun irq */ -#define AZX_REG_RIRBSIZE 0x5e - -#define AZX_REG_IC 0x60 -#define AZX_REG_IR 0x64 -#define AZX_REG_IRS 0x68 -#define AZX_IRS_VALID (1<<1) -#define AZX_IRS_BUSY (1<<0) - -#define AZX_REG_DPLBASE 0x70 -#define AZX_REG_DPUBASE 0x74 -#define AZX_DPLBASE_ENABLE 0x1 /* Enable position buffer */ - -/* SD offset: SDI0=0x80, SDI1=0xa0, ... SDO3=0x160 */ -enum { SDI0, SDI1, SDI2, SDI3, SDO0, SDO1, SDO2, SDO3 }; - -/* stream register offsets from stream base */ -#define AZX_REG_SD_CTL 0x00 -#define AZX_REG_SD_STS 0x03 -#define AZX_REG_SD_LPIB 0x04 -#define AZX_REG_SD_CBL 0x08 -#define AZX_REG_SD_LVI 0x0c -#define AZX_REG_SD_FIFOW 0x0e -#define AZX_REG_SD_FIFOSIZE 0x10 -#define AZX_REG_SD_FORMAT 0x12 -#define AZX_REG_SD_BDLPL 0x18 -#define AZX_REG_SD_BDLPU 0x1c - -/* PCI space */ -#define AZX_PCIREG_TCSEL 0x44 - -/* - * other constants - */ - -/* max number of fragments - we may use more if allocating more pages for BDL */ -#define BDL_SIZE 4096 -#define AZX_MAX_BDL_ENTRIES (BDL_SIZE / 16) -#define AZX_MAX_FRAG 32 -/* max buffer size - no h/w limit, you can increase as you like */ -#define AZX_MAX_BUF_SIZE (1024*1024*1024) - -/* RIRB int mask: overrun[2], response[0] */ -#define RIRB_INT_RESPONSE 0x01 -#define RIRB_INT_OVERRUN 0x04 -#define RIRB_INT_MASK 0x05 - -/* STATESTS int mask: S3,SD2,SD1,SD0 */ -#define AZX_MAX_CODECS 8 -#define AZX_DEFAULT_CODECS 4 -#define STATESTS_INT_MASK ((1 << AZX_MAX_CODECS) - 1) - -/* SD_CTL bits */ -#define SD_CTL_STREAM_RESET 0x01 /* stream reset bit */ -#define SD_CTL_DMA_START 0x02 /* stream DMA start bit */ -#define SD_CTL_STRIPE (3 << 16) /* stripe control */ -#define SD_CTL_TRAFFIC_PRIO (1 << 18) /* traffic priority */ -#define SD_CTL_DIR (1 << 19) /* bi-directional stream */ -#define SD_CTL_STREAM_TAG_MASK (0xf << 20) -#define SD_CTL_STREAM_TAG_SHIFT 20 - -/* SD_CTL and SD_STS */ -#define SD_INT_DESC_ERR 0x10 /* descriptor error interrupt */ -#define SD_INT_FIFO_ERR 0x08 /* FIFO error interrupt */ -#define SD_INT_COMPLETE 0x04 /* completion interrupt */ -#define SD_INT_MASK (SD_INT_DESC_ERR|SD_INT_FIFO_ERR|\ - SD_INT_COMPLETE) - -/* SD_STS */ -#define SD_STS_FIFO_READY 0x20 /* FIFO ready */ - -/* INTCTL and INTSTS */ -#define AZX_INT_ALL_STREAM 0xff /* all stream interrupts */ -#define AZX_INT_CTRL_EN 0x40000000 /* controller interrupt enable bit */ -#define AZX_INT_GLOBAL_EN 0x80000000 /* global interrupt enable bit */ - -/* below are so far hardcoded - should read registers in future */ -#define AZX_MAX_CORB_ENTRIES 256 -#define AZX_MAX_RIRB_ENTRIES 256 - -/* driver quirks (capabilities) */ -/* bits 0-7 are used for indicating driver type */ -#define AZX_DCAPS_NO_TCSEL (1 << 8) /* No Intel TCSEL bit */ -#define AZX_DCAPS_NO_MSI (1 << 9) /* No MSI support */ -#define AZX_DCAPS_SNOOP_MASK (3 << 10) /* snoop type mask */ -#define AZX_DCAPS_SNOOP_OFF (1 << 12) /* snoop default off */ -#define AZX_DCAPS_RIRB_DELAY (1 << 13) /* Long delay in read loop */ -#define AZX_DCAPS_RIRB_PRE_DELAY (1 << 14) /* Put a delay before read */ -#define AZX_DCAPS_CTX_WORKAROUND (1 << 15) /* X-Fi workaround */ -#define AZX_DCAPS_POSFIX_LPIB (1 << 16) /* Use LPIB as default */ -#define AZX_DCAPS_POSFIX_VIA (1 << 17) /* Use VIACOMBO as default */ -#define AZX_DCAPS_NO_64BIT (1 << 18) /* No 64bit address */ -#define AZX_DCAPS_SYNC_WRITE (1 << 19) /* sync each cmd write */ -#define AZX_DCAPS_OLD_SSYNC (1 << 20) /* Old SSYNC reg for ICH */ -#define AZX_DCAPS_NO_ALIGN_BUFSIZE (1 << 21) /* no buffer size alignment */ -/* 22 unused */ -#define AZX_DCAPS_4K_BDLE_BOUNDARY (1 << 23) /* BDLE in 4k boundary */ -#define AZX_DCAPS_REVERSE_ASSIGN (1 << 24) /* Assign devices in reverse order */ -#define AZX_DCAPS_COUNT_LPIB_DELAY (1 << 25) /* Take LPIB as delay */ -#define AZX_DCAPS_PM_RUNTIME (1 << 26) /* runtime PM support */ -#define AZX_DCAPS_I915_POWERWELL (1 << 27) /* HSW i915 powerwell support */ -#define AZX_DCAPS_CORBRP_SELF_CLEAR (1 << 28) /* CORBRP clears itself after reset */ -#define AZX_DCAPS_NO_MSI64 (1 << 29) /* Stick to 32-bit MSIs */ -#define AZX_DCAPS_SEPARATE_STREAM_TAG (1 << 30) /* capture and playback use separate stream tag */ - -enum { - AZX_SNOOP_TYPE_NONE , - AZX_SNOOP_TYPE_SCH, - AZX_SNOOP_TYPE_ATI, - AZX_SNOOP_TYPE_NVIDIA, -}; - -/* HD Audio class code */ -#define PCI_CLASS_MULTIMEDIA_HD_AUDIO 0x0403 - -struct azx_dev { - struct snd_dma_buffer bdl; /* BDL buffer */ - u32 *posbuf; /* position buffer pointer */ - - unsigned int bufsize; /* size of the play buffer in bytes */ - unsigned int period_bytes; /* size of the period in bytes */ - unsigned int frags; /* number for period in the play buffer */ - unsigned int fifo_size; /* FIFO size */ - unsigned long start_wallclk; /* start + minimum wallclk */ - unsigned long period_wallclk; /* wallclk for period */ - - void __iomem *sd_addr; /* stream descriptor pointer */ - - u32 sd_int_sta_mask; /* stream int status mask */ - - /* pcm support */ - struct snd_pcm_substream *substream; /* assigned substream, - * set in PCM open - */ - unsigned int format_val; /* format value to be set in the - * controller and the codec - */ - unsigned char stream_tag; /* assigned stream */ - unsigned char index; /* stream index */ - int assigned_key; /* last device# key assigned to */ - - unsigned int opened:1; - unsigned int running:1; - unsigned int irq_pending:1; - unsigned int prepared:1; - unsigned int locked:1; - /* - * For VIA: - * A flag to ensure DMA position is 0 - * when link position is not greater than FIFO size - */ - unsigned int insufficient:1; - unsigned int wc_marked:1; - unsigned int no_period_wakeup:1; - - struct timecounter azx_tc; - struct cyclecounter azx_cc; - - int delay_negative_threshold; - -#ifdef CONFIG_SND_HDA_DSP_LOADER - /* Allows dsp load to have sole access to the playback stream. */ - struct mutex dsp_mutex; -#endif -}; - -/* CORB/RIRB */ -struct azx_rb { - u32 *buf; /* CORB/RIRB buffer - * Each CORB entry is 4byte, RIRB is 8byte - */ - dma_addr_t addr; /* physical address of CORB/RIRB buffer */ - /* for RIRB */ - unsigned short rp, wp; /* read/write pointers */ - int cmds[AZX_MAX_CODECS]; /* number of pending requests */ - u32 res[AZX_MAX_CODECS]; /* last read value */ -}; - -struct azx; - -/* Functions to read/write to hda registers. */ -struct hda_controller_ops { - /* Register Access */ - void (*reg_writel)(u32 value, u32 __iomem *addr); - u32 (*reg_readl)(u32 __iomem *addr); - void (*reg_writew)(u16 value, u16 __iomem *addr); - u16 (*reg_readw)(u16 __iomem *addr); - void (*reg_writeb)(u8 value, u8 __iomem *addr); - u8 (*reg_readb)(u8 __iomem *addr); - /* Disable msi if supported, PCI only */ - int (*disable_msi_reset_irq)(struct azx *); - /* Allocation ops */ - int (*dma_alloc_pages)(struct azx *chip, - int type, - size_t size, - struct snd_dma_buffer *buf); - void (*dma_free_pages)(struct azx *chip, struct snd_dma_buffer *buf); - int (*substream_alloc_pages)(struct azx *chip, - struct snd_pcm_substream *substream, - size_t size); - int (*substream_free_pages)(struct azx *chip, - struct snd_pcm_substream *substream); - void (*pcm_mmap_prepare)(struct snd_pcm_substream *substream, - struct vm_area_struct *area); - /* Check if current position is acceptable */ - int (*position_check)(struct azx *chip, struct azx_dev *azx_dev); -}; - -struct azx_pcm { - struct azx *chip; - struct snd_pcm *pcm; - struct hda_codec *codec; - struct hda_pcm_stream *hinfo[2]; - struct list_head list; -}; - -typedef unsigned int (*azx_get_pos_callback_t)(struct azx *, struct azx_dev *); -typedef int (*azx_get_delay_callback_t)(struct azx *, struct azx_dev *, unsigned int pos); - -struct azx { - struct snd_card *card; - struct pci_dev *pci; - int dev_index; - - /* chip type specific */ - int driver_type; - unsigned int driver_caps; - int playback_streams; - int playback_index_offset; - int capture_streams; - int capture_index_offset; - int num_streams; - const int *jackpoll_ms; /* per-card jack poll interval */ - - /* Register interaction. */ - const struct hda_controller_ops *ops; - - /* position adjustment callbacks */ - azx_get_pos_callback_t get_position[2]; - azx_get_delay_callback_t get_delay[2]; - - /* pci resources */ - unsigned long addr; - void __iomem *remap_addr; - int irq; - - /* locks */ - spinlock_t reg_lock; - struct mutex open_mutex; /* Prevents concurrent open/close operations */ - - /* streams (x num_streams) */ - struct azx_dev *azx_dev; - - /* PCM */ - struct list_head pcm_list; /* azx_pcm list */ - - /* HD codec */ - unsigned short codec_mask; - int codec_probe_mask; /* copied from probe_mask option */ - struct hda_bus *bus; - unsigned int beep_mode; - - /* CORB/RIRB */ - struct azx_rb corb; - struct azx_rb rirb; - - /* CORB/RIRB and position buffers */ - struct snd_dma_buffer rb; - struct snd_dma_buffer posbuf; - -#ifdef CONFIG_SND_HDA_PATCH_LOADER - const struct firmware *fw; -#endif - - /* flags */ - const int *bdl_pos_adj; - int poll_count; - unsigned int running:1; - unsigned int initialized:1; - unsigned int single_cmd:1; - unsigned int polling_mode:1; - unsigned int msi:1; - unsigned int probing:1; /* codec probing phase */ - unsigned int snoop:1; - unsigned int align_buffer_size:1; - unsigned int region_requested:1; - unsigned int disabled:1; /* disabled by VGA-switcher */ - - /* for debugging */ - unsigned int last_cmd[AZX_MAX_CODECS]; - - /* reboot notifier (for mysterious hangup problem at power-down) */ - struct notifier_block reboot_notifier; - -#ifdef CONFIG_SND_HDA_DSP_LOADER - struct azx_dev saved_azx_dev; -#endif -}; - -#ifdef CONFIG_X86 -#define azx_snoop(chip) ((chip)->snoop) -#else -#define azx_snoop(chip) true -#endif - -/* - * macros for easy use - */ - -#define azx_writel(chip, reg, value) \ - ((chip)->ops->reg_writel(value, (chip)->remap_addr + AZX_REG_##reg)) -#define azx_readl(chip, reg) \ - ((chip)->ops->reg_readl((chip)->remap_addr + AZX_REG_##reg)) -#define azx_writew(chip, reg, value) \ - ((chip)->ops->reg_writew(value, (chip)->remap_addr + AZX_REG_##reg)) -#define azx_readw(chip, reg) \ - ((chip)->ops->reg_readw((chip)->remap_addr + AZX_REG_##reg)) -#define azx_writeb(chip, reg, value) \ - ((chip)->ops->reg_writeb(value, (chip)->remap_addr + AZX_REG_##reg)) -#define azx_readb(chip, reg) \ - ((chip)->ops->reg_readb((chip)->remap_addr + AZX_REG_##reg)) - -#define azx_sd_writel(chip, dev, reg, value) \ - ((chip)->ops->reg_writel(value, (dev)->sd_addr + AZX_REG_##reg)) -#define azx_sd_readl(chip, dev, reg) \ - ((chip)->ops->reg_readl((dev)->sd_addr + AZX_REG_##reg)) -#define azx_sd_writew(chip, dev, reg, value) \ - ((chip)->ops->reg_writew(value, (dev)->sd_addr + AZX_REG_##reg)) -#define azx_sd_readw(chip, dev, reg) \ - ((chip)->ops->reg_readw((dev)->sd_addr + AZX_REG_##reg)) -#define azx_sd_writeb(chip, dev, reg, value) \ - ((chip)->ops->reg_writeb(value, (dev)->sd_addr + AZX_REG_##reg)) -#define azx_sd_readb(chip, dev, reg) \ - ((chip)->ops->reg_readb((dev)->sd_addr + AZX_REG_##reg)) - -#endif /* __SOUND_HDA_PRIV_H */ diff --git a/sound/pci/hda/hda_proc.c b/sound/pci/hda/hda_proc.c index 05e19f7..dacfe74 100644 --- a/sound/pci/hda/hda_proc.c +++ b/sound/pci/hda/hda_proc.c @@ -99,10 +99,10 @@ static void print_nid_array(struct snd_info_buffer *buffer, static void print_nid_pcms(struct snd_info_buffer *buffer, struct hda_codec *codec, hda_nid_t nid) { - int pcm, type; + int type; struct hda_pcm *cpcm; - for (pcm = 0; pcm < codec->num_pcms; pcm++) { - cpcm = &codec->pcm_info[pcm]; + + list_for_each_entry(cpcm, &codec->pcm_list_head, list) { for (type = 0; type < 2; type++) { if (cpcm->stream[type].nid != nid || cpcm->pcm == NULL) continue; @@ -861,7 +861,7 @@ int snd_hda_codec_proc_new(struct hda_codec *codec) int err; snprintf(name, sizeof(name), "codec#%d", codec->addr); - err = snd_card_proc_new(codec->bus->card, name, &entry); + err = snd_card_proc_new(codec->card, name, &entry); if (err < 0) return err; diff --git a/sound/pci/hda/hda_sysfs.c b/sound/pci/hda/hda_sysfs.c index ccc962a..e13c75d 100644 --- a/sound/pci/hda/hda_sysfs.c +++ b/sound/pci/hda/hda_sysfs.c @@ -149,7 +149,7 @@ static int reconfig_codec(struct hda_codec *codec) err = snd_hda_codec_build_controls(codec); if (err < 0) goto error; - err = snd_card_register(codec->bus->card); + err = snd_card_register(codec->card); error: snd_hda_power_down(codec); return err; diff --git a/sound/pci/hda/hda_tegra.c b/sound/pci/hda/hda_tegra.c index 375e94f..2e4fd5c 100644 --- a/sound/pci/hda/hda_tegra.c +++ b/sound/pci/hda/hda_tegra.c @@ -37,7 +37,6 @@ #include "hda_codec.h" #include "hda_controller.h" -#include "hda_priv.h" /* Defines for Nvidia Tegra HDA support */ #define HDA_BAR0 0x8000 @@ -82,7 +81,7 @@ module_param(power_save, bint, 0644); MODULE_PARM_DESC(power_save, "Automatic power-saving timeout (in seconds, 0 = disable)."); #else -static int power_save = 0; +#define power_save 0 #endif /* @@ -250,14 +249,9 @@ static int hda_tegra_suspend(struct device *dev) { struct snd_card *card = dev_get_drvdata(dev); struct azx *chip = card->private_data; - struct azx_pcm *p; struct hda_tegra *hda = container_of(chip, struct hda_tegra, chip); snd_power_change_state(card, SNDRV_CTL_POWER_D3hot); - list_for_each_entry(p, &chip->pcm_list, list) - snd_pcm_suspend_all(p->pcm); - if (chip->initialized) - snd_hda_suspend(chip->bus); azx_stop_chip(chip); azx_enter_link_reset(chip); @@ -278,7 +272,6 @@ static int hda_tegra_resume(struct device *dev) azx_init_chip(chip, 1); - snd_hda_resume(chip->bus); snd_power_change_state(card, SNDRV_CTL_POWER_D0); return 0; @@ -297,8 +290,6 @@ static int hda_tegra_dev_free(struct snd_device *device) int i; struct azx *chip = device->device_data; - azx_notifier_unregister(chip); - if (chip->initialized) { for (i = 0; i < chip->num_streams; i++) azx_stream_stop(chip, &chip->azx_dev[i]); @@ -344,17 +335,6 @@ static int hda_tegra_init_chip(struct azx *chip, struct platform_device *pdev) return 0; } -/* - * The codecs were powered up in snd_hda_codec_new(). - * Now all initialization done, so turn them down if possible - */ -static void power_down_all_codecs(struct azx *chip) -{ - struct hda_codec *codec; - list_for_each_entry(codec, &chip->bus->codec_list, list) - snd_hda_power_down(codec); -} - static int hda_tegra_first_init(struct azx *chip, struct platform_device *pdev) { struct snd_card *card = chip->card; @@ -503,21 +483,15 @@ static int hda_tegra_probe(struct platform_device *pdev) goto out_free; /* create codec instances */ - err = azx_codec_create(chip, NULL, 0, &power_save); + err = azx_bus_create(chip, NULL); if (err < 0) goto out_free; - err = azx_codec_configure(chip); + err = azx_probe_codecs(chip, 0); if (err < 0) goto out_free; - /* create PCM streams */ - err = snd_hda_build_pcms(chip->bus); - if (err < 0) - goto out_free; - - /* create mixer controls */ - err = azx_mixer_create(chip); + err = azx_codec_configure(chip); if (err < 0) goto out_free; @@ -526,8 +500,7 @@ static int hda_tegra_probe(struct platform_device *pdev) goto out_free; chip->running = 1; - power_down_all_codecs(chip); - azx_notifier_register(chip); + snd_hda_set_power_save(chip->bus, power_save * 1000); return 0; @@ -541,6 +514,18 @@ static int hda_tegra_remove(struct platform_device *pdev) return snd_card_free(dev_get_drvdata(&pdev->dev)); } +static void hda_tegra_shutdown(struct platform_device *pdev) +{ + struct snd_card *card = dev_get_drvdata(&pdev->dev); + struct azx *chip; + + if (!card) + return; + chip = card->private_data; + if (chip && chip->running) + azx_stop_chip(chip); +} + static struct platform_driver tegra_platform_hda = { .driver = { .name = "tegra-hda", @@ -549,6 +534,7 @@ static struct platform_driver tegra_platform_hda = { }, .probe = hda_tegra_probe, .remove = hda_tegra_remove, + .shutdown = hda_tegra_shutdown, }; module_platform_driver(tegra_platform_hda); diff --git a/sound/pci/hda/hda_trace.h b/sound/pci/hda/hda_trace.h index 3a1c631..7fedfa8 100644 --- a/sound/pci/hda/hda_trace.h +++ b/sound/pci/hda/hda_trace.h @@ -23,7 +23,7 @@ DECLARE_EVENT_CLASS(hda_cmd, ), TP_fast_assign( - __entry->card = (codec)->bus->card->number; + __entry->card = (codec)->card->number; __entry->addr = (codec)->addr; __entry->val = (val); ), @@ -71,7 +71,7 @@ DECLARE_EVENT_CLASS(hda_power, ), TP_fast_assign( - __entry->card = (codec)->bus->card->number; + __entry->card = (codec)->card->number; __entry->addr = (codec)->addr; ), @@ -87,30 +87,6 @@ DEFINE_EVENT(hda_power, hda_power_up, TP_PROTO(struct hda_codec *codec), TP_ARGS(codec) ); - -TRACE_EVENT(hda_power_count, - TP_PROTO(struct hda_codec *codec), - TP_ARGS(codec), - TP_STRUCT__entry( - __field( unsigned int, card ) - __field( unsigned int, addr ) - __field( int, power_count ) - __field( int, power_on ) - __field( int, power_transition ) - ), - - TP_fast_assign( - __entry->card = (codec)->bus->card->number; - __entry->addr = (codec)->addr; - __entry->power_count = (codec)->power_count; - __entry->power_on = (codec)->power_on; - __entry->power_transition = (codec)->power_transition; - ), - - TP_printk("[%d:%d] power_count=%d, power_on=%d, power_transition=%d", - __entry->card, __entry->addr, __entry->power_count, - __entry->power_on, __entry->power_transition) -); #endif /* CONFIG_PM */ TRACE_EVENT(hda_unsol_event, diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index d285904..af4c7be 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -1194,20 +1194,8 @@ MODULE_ALIAS("snd-hda-codec-id:11d4*"); MODULE_LICENSE("GPL"); MODULE_DESCRIPTION("Analog Devices HD-audio codec"); -static struct hda_codec_preset_list analog_list = { +static struct hda_codec_driver analog_driver = { .preset = snd_hda_preset_analog, - .owner = THIS_MODULE, }; -static int __init patch_analog_init(void) -{ - return snd_hda_add_codec_preset(&analog_list); -} - -static void __exit patch_analog_exit(void) -{ - snd_hda_delete_codec_preset(&analog_list); -} - -module_init(patch_analog_init) -module_exit(patch_analog_exit) +module_hda_codec_driver(analog_driver); diff --git a/sound/pci/hda/patch_ca0110.c b/sound/pci/hda/patch_ca0110.c index 5e65999..4473026 100644 --- a/sound/pci/hda/patch_ca0110.c +++ b/sound/pci/hda/patch_ca0110.c @@ -98,20 +98,8 @@ MODULE_ALIAS("snd-hda-codec-id:1102000d"); MODULE_LICENSE("GPL"); MODULE_DESCRIPTION("Creative CA0110-IBG HD-audio codec"); -static struct hda_codec_preset_list ca0110_list = { +static struct hda_codec_driver ca0110_driver = { .preset = snd_hda_preset_ca0110, - .owner = THIS_MODULE, }; -static int __init patch_ca0110_init(void) -{ - return snd_hda_add_codec_preset(&ca0110_list); -} - -static void __exit patch_ca0110_exit(void) -{ - snd_hda_delete_codec_preset(&ca0110_list); -} - -module_init(patch_ca0110_init) -module_exit(patch_ca0110_exit) +module_hda_codec_driver(ca0110_driver); diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c index e0383ee..72d2065 100644 --- a/sound/pci/hda/patch_ca0132.c +++ b/sound/pci/hda/patch_ca0132.c @@ -719,7 +719,6 @@ struct ca0132_spec { unsigned int num_inputs; hda_nid_t shared_mic_nid; hda_nid_t shared_out_nid; - struct hda_pcm pcm_rec[5]; /* PCM information */ /* chip access */ struct mutex chipio_mutex; /* chip access mutex */ @@ -4036,12 +4035,11 @@ static struct hda_pcm_stream ca0132_pcm_digital_capture = { static int ca0132_build_pcms(struct hda_codec *codec) { struct ca0132_spec *spec = codec->spec; - struct hda_pcm *info = spec->pcm_rec; + struct hda_pcm *info; - codec->pcm_info = info; - codec->num_pcms = 0; - - info->name = "CA0132 Analog"; + info = snd_hda_codec_pcm_new(codec, "CA0132 Analog"); + if (!info) + return -ENOMEM; info->stream[SNDRV_PCM_STREAM_PLAYBACK] = ca0132_pcm_analog_playback; info->stream[SNDRV_PCM_STREAM_PLAYBACK].nid = spec->dacs[0]; info->stream[SNDRV_PCM_STREAM_PLAYBACK].channels_max = @@ -4049,27 +4047,27 @@ static int ca0132_build_pcms(struct hda_codec *codec) info->stream[SNDRV_PCM_STREAM_CAPTURE] = ca0132_pcm_analog_capture; info->stream[SNDRV_PCM_STREAM_CAPTURE].substreams = 1; info->stream[SNDRV_PCM_STREAM_CAPTURE].nid = spec->adcs[0]; - codec->num_pcms++; - info++; - info->name = "CA0132 Analog Mic-In2"; + info = snd_hda_codec_pcm_new(codec, "CA0132 Analog Mic-In2"); + if (!info) + return -ENOMEM; info->stream[SNDRV_PCM_STREAM_CAPTURE] = ca0132_pcm_analog_capture; info->stream[SNDRV_PCM_STREAM_CAPTURE].substreams = 1; info->stream[SNDRV_PCM_STREAM_CAPTURE].nid = spec->adcs[1]; - codec->num_pcms++; - info++; - info->name = "CA0132 What U Hear"; + info = snd_hda_codec_pcm_new(codec, "CA0132 What U Hear"); + if (!info) + return -ENOMEM; info->stream[SNDRV_PCM_STREAM_CAPTURE] = ca0132_pcm_analog_capture; info->stream[SNDRV_PCM_STREAM_CAPTURE].substreams = 1; info->stream[SNDRV_PCM_STREAM_CAPTURE].nid = spec->adcs[2]; - codec->num_pcms++; if (!spec->dig_out && !spec->dig_in) return 0; - info++; - info->name = "CA0132 Digital"; + info = snd_hda_codec_pcm_new(codec, "CA0132 Digital"); + if (!info) + return -ENOMEM; info->pcm_type = HDA_PCM_TYPE_SPDIF; if (spec->dig_out) { info->stream[SNDRV_PCM_STREAM_PLAYBACK] = @@ -4081,7 +4079,6 @@ static int ca0132_build_pcms(struct hda_codec *codec) ca0132_pcm_digital_capture; info->stream[SNDRV_PCM_STREAM_CAPTURE].nid = spec->dig_in; } - codec->num_pcms++; return 0; } @@ -4352,7 +4349,7 @@ static bool ca0132_download_dsp_images(struct hda_codec *codec) const struct dsp_image_seg *dsp_os_image; const struct firmware *fw_entry; - if (request_firmware(&fw_entry, EFX_FILE, codec->bus->card->dev) != 0) + if (request_firmware(&fw_entry, EFX_FILE, codec->card->dev) != 0) return false; dsp_os_image = (struct dsp_image_seg *)(fw_entry->data); @@ -4413,8 +4410,7 @@ static void hp_callback(struct hda_codec *codec, struct hda_jack_callback *cb) * state machine run. */ cancel_delayed_work_sync(&spec->unsol_hp_work); - queue_delayed_work(codec->bus->workq, &spec->unsol_hp_work, - msecs_to_jiffies(500)); + schedule_delayed_work(&spec->unsol_hp_work, msecs_to_jiffies(500)); cb->tbl->block_report = 1; } @@ -4702,20 +4698,8 @@ MODULE_ALIAS("snd-hda-codec-id:11020011"); MODULE_LICENSE("GPL"); MODULE_DESCRIPTION("Creative Sound Core3D codec"); -static struct hda_codec_preset_list ca0132_list = { +static struct hda_codec_driver ca0132_driver = { .preset = snd_hda_preset_ca0132, - .owner = THIS_MODULE, }; -static int __init patch_ca0132_init(void) -{ - return snd_hda_add_codec_preset(&ca0132_list); -} - -static void __exit patch_ca0132_exit(void) -{ - snd_hda_delete_codec_preset(&ca0132_list); -} - -module_init(patch_ca0132_init) -module_exit(patch_ca0132_exit) +module_hda_codec_driver(ca0132_driver); diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c index dd2b3d9..50e9dd6 100644 --- a/sound/pci/hda/patch_cirrus.c +++ b/sound/pci/hda/patch_cirrus.c @@ -1221,20 +1221,8 @@ MODULE_ALIAS("snd-hda-codec-id:10134213"); MODULE_LICENSE("GPL"); MODULE_DESCRIPTION("Cirrus Logic HD-audio codec"); -static struct hda_codec_preset_list cirrus_list = { +static struct hda_codec_driver cirrus_driver = { .preset = snd_hda_preset_cirrus, - .owner = THIS_MODULE, }; -static int __init patch_cirrus_init(void) -{ - return snd_hda_add_codec_preset(&cirrus_list); -} - -static void __exit patch_cirrus_exit(void) -{ - snd_hda_delete_codec_preset(&cirrus_list); -} - -module_init(patch_cirrus_init) -module_exit(patch_cirrus_exit) +module_hda_codec_driver(cirrus_driver); diff --git a/sound/pci/hda/patch_cmedia.c b/sound/pci/hda/patch_cmedia.c index c895a8f..617d901 100644 --- a/sound/pci/hda/patch_cmedia.c +++ b/sound/pci/hda/patch_cmedia.c @@ -137,20 +137,8 @@ MODULE_ALIAS("snd-hda-codec-id:434d4980"); MODULE_LICENSE("GPL"); MODULE_DESCRIPTION("C-Media HD-audio codec"); -static struct hda_codec_preset_list cmedia_list = { +static struct hda_codec_driver cmedia_driver = { .preset = snd_hda_preset_cmedia, - .owner = THIS_MODULE, }; -static int __init patch_cmedia_init(void) -{ - return snd_hda_add_codec_preset(&cmedia_list); -} - -static void __exit patch_cmedia_exit(void) -{ - snd_hda_delete_codec_preset(&cmedia_list); -} - -module_init(patch_cmedia_init) -module_exit(patch_cmedia_exit) +module_hda_codec_driver(cmedia_driver); diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index da67ea8..5aa466a 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -1018,20 +1018,8 @@ MODULE_ALIAS("snd-hda-codec-id:14f151d7"); MODULE_LICENSE("GPL"); MODULE_DESCRIPTION("Conexant HD-audio codec"); -static struct hda_codec_preset_list conexant_list = { +static struct hda_codec_driver conexant_driver = { .preset = snd_hda_preset_conexant, - .owner = THIS_MODULE, }; -static int __init patch_conexant_init(void) -{ - return snd_hda_add_codec_preset(&conexant_list); -} - -static void __exit patch_conexant_exit(void) -{ - snd_hda_delete_codec_preset(&conexant_list); -} - -module_init(patch_conexant_init) -module_exit(patch_conexant_exit) +module_hda_codec_driver(conexant_driver); diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index b422e40..7e9ff7b 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -86,7 +86,6 @@ struct hdmi_spec_per_pin { bool non_pcm; bool chmap_set; /* channel-map override by ALSA API? */ unsigned char chmap[8]; /* ALSA API channel-map */ - char pcm_name[8]; /* filled in build_pcm callbacks */ #ifdef CONFIG_PROC_FS struct snd_info_entry *proc_entry; #endif @@ -132,7 +131,7 @@ struct hdmi_spec { int num_pins; struct snd_array pins; /* struct hdmi_spec_per_pin */ - struct snd_array pcm_rec; /* struct hda_pcm */ + struct hda_pcm *pcm_rec[16]; unsigned int channels_max; /* max over all cvts */ struct hdmi_eld temp_eld; @@ -355,8 +354,7 @@ static struct cea_channel_speaker_allocation channel_allocations[] = { ((struct hdmi_spec_per_pin *)snd_array_elem(&spec->pins, idx)) #define get_cvt(spec, idx) \ ((struct hdmi_spec_per_cvt *)snd_array_elem(&spec->cvts, idx)) -#define get_pcm_rec(spec, idx) \ - ((struct hda_pcm *)snd_array_elem(&spec->pcm_rec, idx)) +#define get_pcm_rec(spec, idx) ((spec)->pcm_rec[idx]) static int pin_nid_to_pin_index(struct hda_codec *codec, hda_nid_t pin_nid) { @@ -579,7 +577,7 @@ static int eld_proc_new(struct hdmi_spec_per_pin *per_pin, int index) int err; snprintf(name, sizeof(name), "eld#%d.%d", codec->addr, index); - err = snd_card_proc_new(codec->bus->card, name, &entry); + err = snd_card_proc_new(codec->card, name, &entry); if (err < 0) return err; @@ -594,7 +592,7 @@ static int eld_proc_new(struct hdmi_spec_per_pin *per_pin, int index) static void eld_proc_free(struct hdmi_spec_per_pin *per_pin) { if (!per_pin->codec->bus->shutdown && per_pin->proc_entry) { - snd_device_free(per_pin->codec->bus->card, per_pin->proc_entry); + snd_device_free(per_pin->codec->card, per_pin->proc_entry); per_pin->proc_entry = NULL; } } @@ -1578,9 +1576,8 @@ static bool hdmi_present_sense(struct hdmi_spec_per_pin *per_pin, int repoll) update_eld = true; } else if (repoll) { - queue_delayed_work(codec->bus->workq, - &per_pin->work, - msecs_to_jiffies(300)); + schedule_delayed_work(&per_pin->work, + msecs_to_jiffies(300)); goto unlock; } } @@ -1624,7 +1621,7 @@ static bool hdmi_present_sense(struct hdmi_spec_per_pin *per_pin, int repoll) } if (eld_changed) - snd_ctl_notify(codec->bus->card, + snd_ctl_notify(codec->card, SNDRV_CTL_EVENT_MASK_VALUE | SNDRV_CTL_EVENT_MASK_INFO, &per_pin->eld_ctl->id); unlock: @@ -2056,11 +2053,10 @@ static int generic_hdmi_build_pcms(struct hda_codec *codec) struct hdmi_spec_per_pin *per_pin; per_pin = get_pin(spec, pin_idx); - sprintf(per_pin->pcm_name, "HDMI %d", pin_idx); - info = snd_array_new(&spec->pcm_rec); + info = snd_hda_codec_pcm_new(codec, "HDMI %d", pin_idx); if (!info) return -ENOMEM; - info->name = per_pin->pcm_name; + spec->pcm_rec[pin_idx] = info; info->pcm_type = HDA_PCM_TYPE_HDMI; info->own_chmap = true; @@ -2070,9 +2066,6 @@ static int generic_hdmi_build_pcms(struct hda_codec *codec) /* other pstr fields are set in open */ } - codec->num_pcms = spec->num_pins; - codec->pcm_info = spec->pcm_rec.list; - return 0; } @@ -2125,13 +2118,15 @@ static int generic_hdmi_build_controls(struct hda_codec *codec) /* add channel maps */ for (pin_idx = 0; pin_idx < spec->num_pins; pin_idx++) { + struct hda_pcm *pcm; struct snd_pcm_chmap *chmap; struct snd_kcontrol *kctl; int i; - if (!codec->pcm_info[pin_idx].pcm) + pcm = spec->pcm_rec[pin_idx]; + if (!pcm || !pcm->pcm) break; - err = snd_pcm_add_chmap_ctls(codec->pcm_info[pin_idx].pcm, + err = snd_pcm_add_chmap_ctls(pcm->pcm, SNDRV_PCM_STREAM_PLAYBACK, NULL, 0, pin_idx, &chmap); if (err < 0) @@ -2186,14 +2181,12 @@ static void hdmi_array_init(struct hdmi_spec *spec, int nums) { snd_array_init(&spec->pins, sizeof(struct hdmi_spec_per_pin), nums); snd_array_init(&spec->cvts, sizeof(struct hdmi_spec_per_cvt), nums); - snd_array_init(&spec->pcm_rec, sizeof(struct hda_pcm), nums); } static void hdmi_array_free(struct hdmi_spec *spec) { snd_array_free(&spec->pins); snd_array_free(&spec->cvts); - snd_array_free(&spec->pcm_rec); } static void generic_hdmi_free(struct hda_codec *codec) @@ -2204,11 +2197,10 @@ static void generic_hdmi_free(struct hda_codec *codec) for (pin_idx = 0; pin_idx < spec->num_pins; pin_idx++) { struct hdmi_spec_per_pin *per_pin = get_pin(spec, pin_idx); - cancel_delayed_work(&per_pin->work); + cancel_delayed_work_sync(&per_pin->work); eld_proc_free(per_pin); } - flush_workqueue(codec->bus->workq); hdmi_array_free(spec); kfree(spec); } @@ -2381,11 +2373,10 @@ static int simple_playback_build_pcms(struct hda_codec *codec) chans = get_wcaps(codec, per_cvt->cvt_nid); chans = get_wcaps_channels(chans); - info = snd_array_new(&spec->pcm_rec); + info = snd_hda_codec_pcm_new(codec, "HDMI 0"); if (!info) return -ENOMEM; - info->name = get_pin(spec, 0)->pcm_name; - sprintf(info->name, "HDMI 0"); + spec->pcm_rec[0] = info; info->pcm_type = HDA_PCM_TYPE_HDMI; pstr = &info->stream[SNDRV_PCM_STREAM_PLAYBACK]; *pstr = spec->pcm_playback; @@ -2393,9 +2384,6 @@ static int simple_playback_build_pcms(struct hda_codec *codec) if (pstr->channels_max <= 2 && chans && chans <= 16) pstr->channels_max = chans; - codec->num_pcms = 1; - codec->pcm_info = info; - return 0; } @@ -3301,15 +3289,6 @@ static int patch_via_hdmi(struct hda_codec *codec) } /* - * called from hda_codec.c for generic HDMI support - */ -int snd_hda_parse_hdmi_codec(struct hda_codec *codec) -{ - return patch_generic_hdmi(codec); -} -EXPORT_SYMBOL_GPL(snd_hda_parse_hdmi_codec); - -/* * patch entries */ static const struct hda_codec_preset snd_hda_preset_hdmi[] = { @@ -3373,6 +3352,8 @@ static const struct hda_codec_preset snd_hda_preset_hdmi[] = { { .id = 0x80862882, .name = "Valleyview2 HDMI", .patch = patch_generic_hdmi }, { .id = 0x80862883, .name = "Braswell HDMI", .patch = patch_generic_hdmi }, { .id = 0x808629fb, .name = "Crestline HDMI", .patch = patch_generic_hdmi }, +/* special ID for generic HDMI */ +{ .id = HDA_CODEC_ID_GENERIC_HDMI, .patch = patch_generic_hdmi }, {} /* terminator */ }; @@ -3442,20 +3423,8 @@ MODULE_ALIAS("snd-hda-codec-intelhdmi"); MODULE_ALIAS("snd-hda-codec-nvhdmi"); MODULE_ALIAS("snd-hda-codec-atihdmi"); -static struct hda_codec_preset_list intel_list = { +static struct hda_codec_driver hdmi_driver = { .preset = snd_hda_preset_hdmi, - .owner = THIS_MODULE, }; -static int __init patch_hdmi_init(void) -{ - return snd_hda_add_codec_preset(&intel_list); -} - -static void __exit patch_hdmi_exit(void) -{ - snd_hda_delete_codec_preset(&intel_list); -} - -module_init(patch_hdmi_init) -module_exit(patch_hdmi_exit) +module_hda_codec_driver(hdmi_driver); diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 526398a..124eacf 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -2602,53 +2602,12 @@ static int patch_alc268(struct hda_codec *codec) * ALC269 */ -static int playback_pcm_open(struct hda_pcm_stream *hinfo, - struct hda_codec *codec, - struct snd_pcm_substream *substream) -{ - struct hda_gen_spec *spec = codec->spec; - return snd_hda_multi_out_analog_open(codec, &spec->multiout, substream, - hinfo); -} - -static int playback_pcm_prepare(struct hda_pcm_stream *hinfo, - struct hda_codec *codec, - unsigned int stream_tag, - unsigned int format, - struct snd_pcm_substream *substream) -{ - struct hda_gen_spec *spec = codec->spec; - return snd_hda_multi_out_analog_prepare(codec, &spec->multiout, - stream_tag, format, substream); -} - -static int playback_pcm_cleanup(struct hda_pcm_stream *hinfo, - struct hda_codec *codec, - struct snd_pcm_substream *substream) -{ - struct hda_gen_spec *spec = codec->spec; - return snd_hda_multi_out_analog_cleanup(codec, &spec->multiout); -} - static const struct hda_pcm_stream alc269_44k_pcm_analog_playback = { - .substreams = 1, - .channels_min = 2, - .channels_max = 8, .rates = SNDRV_PCM_RATE_44100, /* fixed rate */ - /* NID is set in alc_build_pcms */ - .ops = { - .open = playback_pcm_open, - .prepare = playback_pcm_prepare, - .cleanup = playback_pcm_cleanup - }, }; static const struct hda_pcm_stream alc269_44k_pcm_analog_capture = { - .substreams = 1, - .channels_min = 2, - .channels_max = 2, .rates = SNDRV_PCM_RATE_44100, /* fixed rate */ - /* NID is set in alc_build_pcms */ }; /* different alc269-variants */ @@ -5850,7 +5809,7 @@ static void alc_fixup_bass_chmap(struct hda_codec *codec, { if (action == HDA_FIXUP_ACT_BUILD) { struct alc_spec *spec = codec->spec; - spec->gen.pcm_rec[0].stream[0].chmap = asus_pcm_2_1_chmaps; + spec->gen.pcm_rec[0]->stream[0].chmap = asus_pcm_2_1_chmaps; } } @@ -6521,20 +6480,8 @@ MODULE_ALIAS("snd-hda-codec-id:10ec*"); MODULE_LICENSE("GPL"); MODULE_DESCRIPTION("Realtek HD-audio codec"); -static struct hda_codec_preset_list realtek_list = { +static struct hda_codec_driver realtek_driver = { .preset = snd_hda_preset_realtek, - .owner = THIS_MODULE, }; -static int __init patch_realtek_init(void) -{ - return snd_hda_add_codec_preset(&realtek_list); -} - -static void __exit patch_realtek_exit(void) -{ - snd_hda_delete_codec_preset(&realtek_list); -} - -module_init(patch_realtek_init) -module_exit(patch_realtek_exit) +module_hda_codec_driver(realtek_driver); diff --git a/sound/pci/hda/patch_si3054.c b/sound/pci/hda/patch_si3054.c index 3208ad69..df24313 100644 --- a/sound/pci/hda/patch_si3054.c +++ b/sound/pci/hda/patch_si3054.c @@ -83,7 +83,6 @@ struct si3054_spec { unsigned international; - struct hda_pcm pcm; }; @@ -199,11 +198,11 @@ static const struct hda_pcm_stream si3054_pcm = { static int si3054_build_pcms(struct hda_codec *codec) { - struct si3054_spec *spec = codec->spec; - struct hda_pcm *info = &spec->pcm; - codec->num_pcms = 1; - codec->pcm_info = info; - info->name = "Si3054 Modem"; + struct hda_pcm *info; + + info = snd_hda_codec_pcm_new(codec, "Si3054 Modem"); + if (!info) + return -ENOMEM; info->stream[SNDRV_PCM_STREAM_PLAYBACK] = si3054_pcm; info->stream[SNDRV_PCM_STREAM_CAPTURE] = si3054_pcm; info->stream[SNDRV_PCM_STREAM_PLAYBACK].nid = codec->mfg; @@ -319,20 +318,8 @@ MODULE_ALIAS("snd-hda-codec-id:18540018"); MODULE_LICENSE("GPL"); MODULE_DESCRIPTION("Si3054 HD-audio modem codec"); -static struct hda_codec_preset_list si3054_list = { +static struct hda_codec_driver si3054_driver = { .preset = snd_hda_preset_si3054, - .owner = THIS_MODULE, }; -static int __init patch_si3054_init(void) -{ - return snd_hda_add_codec_preset(&si3054_list); -} - -static void __exit patch_si3054_exit(void) -{ - snd_hda_delete_codec_preset(&si3054_list); -} - -module_init(patch_si3054_init) -module_exit(patch_si3054_exit) +module_hda_codec_driver(si3054_driver); diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 87eff31..5b7c173 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -2132,8 +2132,10 @@ static void stac92hd83xxx_fixup_hp_mic_led(struct hda_codec *codec, if (action == HDA_FIXUP_ACT_PRE_PROBE) { spec->mic_mute_led_gpio = 0x08; /* GPIO3 */ +#ifdef CONFIG_PM /* resetting controller clears GPIO, so we need to keep on */ - codec->bus->power_keep_link_on = 1; + codec->d3_stop_clk = 0; +#endif } } @@ -4223,6 +4225,12 @@ static int stac_parse_auto_config(struct hda_codec *codec) if (err < 0) return err; + if (spec->vref_mute_led_nid) { + err = snd_hda_gen_fix_pin_power(codec, spec->vref_mute_led_nid); + if (err < 0) + return err; + } + /* setup analog beep controls */ if (spec->anabeep_nid > 0) { err = stac_auto_create_beep_ctls(codec, @@ -4392,6 +4400,7 @@ static const struct hda_codec_ops stac_patch_ops = { #ifdef CONFIG_PM .suspend = stac_suspend, #endif + .stream_pm = snd_hda_gen_stream_pm, .reboot_notify = stac_shutup, }; @@ -4485,6 +4494,7 @@ static int patch_stac92hd73xx(struct hda_codec *codec) return err; spec = codec->spec; + codec->power_save_node = 1; spec->linear_tone_beep = 0; spec->gen.mixer_nid = 0x1d; spec->have_spdif_mux = 1; @@ -4590,6 +4600,7 @@ static int patch_stac92hd83xxx(struct hda_codec *codec) codec->epss = 0; /* longer delay needed for D3 */ spec = codec->spec; + codec->power_save_node = 1; spec->linear_tone_beep = 0; spec->gen.own_eapd_ctl = 1; spec->gen.power_down_unused = 1; @@ -4639,6 +4650,7 @@ static int patch_stac92hd95(struct hda_codec *codec) codec->epss = 0; /* longer delay needed for D3 */ spec = codec->spec; + codec->power_save_node = 1; spec->linear_tone_beep = 0; spec->gen.own_eapd_ctl = 1; spec->gen.power_down_unused = 1; @@ -4680,6 +4692,7 @@ static int patch_stac92hd71bxx(struct hda_codec *codec) return err; spec = codec->spec; + codec->power_save_node = 1; spec->linear_tone_beep = 0; spec->gen.own_eapd_ctl = 1; spec->gen.power_down_unused = 1; @@ -5091,20 +5104,8 @@ MODULE_ALIAS("snd-hda-codec-id:111d*"); MODULE_LICENSE("GPL"); MODULE_DESCRIPTION("IDT/Sigmatel HD-audio codec"); -static struct hda_codec_preset_list sigmatel_list = { +static struct hda_codec_driver sigmatel_driver = { .preset = snd_hda_preset_sigmatel, - .owner = THIS_MODULE, }; -static int __init patch_sigmatel_init(void) -{ - return snd_hda_add_codec_preset(&sigmatel_list); -} - -static void __exit patch_sigmatel_exit(void) -{ - snd_hda_delete_codec_preset(&sigmatel_list); -} - -module_init(patch_sigmatel_init) -module_exit(patch_sigmatel_exit) +module_hda_codec_driver(sigmatel_driver); diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index 3de6d3d..485663b 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -99,7 +99,6 @@ struct via_spec { /* HP mode source */ unsigned int dmic_enabled; - unsigned int no_pin_power_ctl; enum VIA_HDA_CODEC codec_type; /* analog low-power control */ @@ -108,9 +107,6 @@ struct via_spec { /* work to check hp jack state */ int hp_work_active; int vt1708_jack_detect; - - void (*set_widgets_power_state)(struct hda_codec *codec); - unsigned int dac_stream_tag[4]; }; static enum VIA_HDA_CODEC get_codec_type(struct hda_codec *codec); @@ -133,11 +129,12 @@ static struct via_spec *via_new_spec(struct hda_codec *codec) /* VT1708BCE & VT1708S are almost same */ if (spec->codec_type == VT1708BCE) spec->codec_type = VT1708S; - spec->no_pin_power_ctl = 1; spec->gen.indep_hp = 1; spec->gen.keep_eapd_on = 1; spec->gen.pcm_playback_hook = via_playback_pcm_hook; spec->gen.add_stereo_mix_input = HDA_HINT_STEREO_MIX_AUTO; + codec->power_save_node = 1; + spec->gen.power_down_unused = 1; return spec; } @@ -222,98 +219,13 @@ static void vt1708_update_hp_work(struct hda_codec *codec) if (!spec->hp_work_active) { codec->jackpoll_interval = msecs_to_jiffies(100); snd_hda_codec_write(codec, 0x1, 0, 0xf81, 0); - queue_delayed_work(codec->bus->workq, - &codec->jackpoll_work, 0); + schedule_delayed_work(&codec->jackpoll_work, 0); spec->hp_work_active = true; } } else if (!hp_detect_with_aa(codec)) vt1708_stop_hp_work(codec); } -static void set_widgets_power_state(struct hda_codec *codec) -{ -#if 0 /* FIXME: the assumed connections don't match always with the - * actual routes by the generic parser, so better to disable - * the control for safety. - */ - struct via_spec *spec = codec->spec; - if (spec->set_widgets_power_state) - spec->set_widgets_power_state(codec); -#endif -} - -static void update_power_state(struct hda_codec *codec, hda_nid_t nid, - unsigned int parm) -{ - if (snd_hda_check_power_state(codec, nid, parm)) - return; - snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_POWER_STATE, parm); -} - -static void update_conv_power_state(struct hda_codec *codec, hda_nid_t nid, - unsigned int parm, unsigned int index) -{ - struct via_spec *spec = codec->spec; - unsigned int format; - - if (snd_hda_check_power_state(codec, nid, parm)) - return; - format = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_CONV, 0); - if (format && (spec->dac_stream_tag[index] != format)) - spec->dac_stream_tag[index] = format; - - snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_POWER_STATE, parm); - if (parm == AC_PWRST_D0) { - format = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_CONV, 0); - if (!format && (spec->dac_stream_tag[index] != format)) - snd_hda_codec_write(codec, nid, 0, - AC_VERB_SET_CHANNEL_STREAMID, - spec->dac_stream_tag[index]); - } -} - -static bool smart51_enabled(struct hda_codec *codec) -{ - struct via_spec *spec = codec->spec; - return spec->gen.ext_channel_count > 2; -} - -static bool is_smart51_pins(struct hda_codec *codec, hda_nid_t pin) -{ - struct via_spec *spec = codec->spec; - int i; - - for (i = 0; i < spec->gen.multi_ios; i++) - if (spec->gen.multi_io[i].pin == pin) - return true; - return false; -} - -static void set_pin_power_state(struct hda_codec *codec, hda_nid_t nid, - unsigned int *affected_parm) -{ - unsigned parm; - unsigned def_conf = snd_hda_codec_get_pincfg(codec, nid); - unsigned no_presence = (def_conf & AC_DEFCFG_MISC) - >> AC_DEFCFG_MISC_SHIFT - & AC_DEFCFG_MISC_NO_PRESENCE; /* do not support pin sense */ - struct via_spec *spec = codec->spec; - unsigned present = 0; - - no_presence |= spec->no_pin_power_ctl; - if (!no_presence) - present = snd_hda_jack_detect(codec, nid); - if ((smart51_enabled(codec) && is_smart51_pins(codec, nid)) - || ((no_presence || present) - && get_defcfg_connect(def_conf) != AC_JACK_PORT_NONE)) { - *affected_parm = AC_PWRST_D0; /* if it's connected */ - parm = AC_PWRST_D0; - } else - parm = AC_PWRST_D3; - - update_power_state(codec, nid, parm); -} - static int via_pin_power_ctl_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { @@ -324,8 +236,7 @@ static int via_pin_power_ctl_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - struct via_spec *spec = codec->spec; - ucontrol->value.enumerated.item[0] = !spec->no_pin_power_ctl; + ucontrol->value.enumerated.item[0] = codec->power_save_node; return 0; } @@ -334,12 +245,12 @@ static int via_pin_power_ctl_put(struct snd_kcontrol *kcontrol, { struct hda_codec *codec = snd_kcontrol_chip(kcontrol); struct via_spec *spec = codec->spec; - unsigned int val = !ucontrol->value.enumerated.item[0]; + bool val = !!ucontrol->value.enumerated.item[0]; - if (val == spec->no_pin_power_ctl) + if (val == codec->power_save_node) return 0; - spec->no_pin_power_ctl = val; - set_widgets_power_state(codec); + codec->power_save_node = val; + spec->gen.power_down_unused = val; analog_low_current_mode(codec); return 1; } @@ -384,7 +295,7 @@ static void __analog_low_current_mode(struct hda_codec *codec, bool force) bool enable; unsigned int verb, parm; - if (spec->no_pin_power_ctl) + if (!codec->power_save_node) enable = false; else enable = is_aa_path_mute(codec) && !spec->gen.active_streams; @@ -441,8 +352,7 @@ static int via_build_controls(struct hda_codec *codec) if (err < 0) return err; - if (spec->set_widgets_power_state) - spec->mixers[spec->num_mixers++] = via_pin_power_ctl_enum; + spec->mixers[spec->num_mixers++] = via_pin_power_ctl_enum; for (i = 0; i < spec->num_mixers; i++) { err = snd_hda_add_new_ctls(codec, spec->mixers[i]); @@ -486,7 +396,6 @@ static int via_suspend(struct hda_codec *codec) static int via_check_power_status(struct hda_codec *codec, hda_nid_t nid) { struct via_spec *spec = codec->spec; - set_widgets_power_state(codec); analog_low_current_mode(codec); vt1708_update_hp_work(codec); return snd_hda_check_amp_list_power(codec, &spec->gen.loopback, nid); @@ -574,34 +483,6 @@ static const struct snd_kcontrol_new vt1708_jack_detect_ctl[] = { {} /* terminator */ }; -static void via_jack_powerstate_event(struct hda_codec *codec, - struct hda_jack_callback *tbl) -{ - set_widgets_power_state(codec); -} - -static void via_set_jack_unsol_events(struct hda_codec *codec) -{ - struct via_spec *spec = codec->spec; - struct auto_pin_cfg *cfg = &spec->gen.autocfg; - hda_nid_t pin; - int i; - - for (i = 0; i < cfg->line_outs; i++) { - pin = cfg->line_out_pins[i]; - if (pin && is_jack_detectable(codec, pin)) - snd_hda_jack_detect_enable_callback(codec, pin, - via_jack_powerstate_event); - } - - for (i = 0; i < cfg->num_inputs; i++) { - pin = cfg->line_out_pins[i]; - if (pin && is_jack_detectable(codec, pin)) - snd_hda_jack_detect_enable_callback(codec, pin, - via_jack_powerstate_event); - } -} - static const struct badness_table via_main_out_badness = { .no_primary_dac = 0x10000, .no_dac = 0x4000, @@ -635,7 +516,9 @@ static int via_parse_auto_config(struct hda_codec *codec) if (err < 0) return err; - via_set_jack_unsol_events(codec); + /* disable widget PM at start for compatibility */ + codec->power_save_node = 0; + spec->gen.power_down_unused = 0; return 0; } @@ -648,7 +531,6 @@ static int via_init(struct hda_codec *codec) snd_hda_sequence_write(codec, spec->init_verbs[i]); /* init power states */ - set_widgets_power_state(codec); __analog_low_current_mode(codec, true); snd_hda_gen_init(codec); @@ -683,8 +565,10 @@ static int vt1708_build_pcms(struct hda_codec *codec) * 24bit samples are used. Until any workaround is found, * disable the 24bit format, so far. */ - for (i = 0; i < codec->num_pcms; i++) { - struct hda_pcm *info = &spec->gen.pcm_rec[i]; + for (i = 0; i < ARRAY_SIZE(spec->gen.pcm_rec); i++) { + struct hda_pcm *info = spec->gen.pcm_rec[i]; + if (!info) + continue; if (!info->stream[SNDRV_PCM_STREAM_PLAYBACK].substreams || info->pcm_type != HDA_PCM_TYPE_AUDIO) continue; @@ -766,78 +650,6 @@ static int patch_vt1709(struct hda_codec *codec) return 0; } -static void set_widgets_power_state_vt1708B(struct hda_codec *codec) -{ - struct via_spec *spec = codec->spec; - int imux_is_smixer; - unsigned int parm; - int is_8ch = 0; - if ((spec->codec_type != VT1708B_4CH) && - (codec->vendor_id != 0x11064397)) - is_8ch = 1; - - /* SW0 (17h) = stereo mixer */ - imux_is_smixer = - (snd_hda_codec_read(codec, 0x17, 0, AC_VERB_GET_CONNECT_SEL, 0x00) - == ((spec->codec_type == VT1708S) ? 5 : 0)); - /* inputs */ - /* PW 1/2/5 (1ah/1bh/1eh) */ - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x1a, &parm); - set_pin_power_state(codec, 0x1b, &parm); - set_pin_power_state(codec, 0x1e, &parm); - if (imux_is_smixer) - parm = AC_PWRST_D0; - /* SW0 (17h), AIW 0/1 (13h/14h) */ - update_power_state(codec, 0x17, parm); - update_power_state(codec, 0x13, parm); - update_power_state(codec, 0x14, parm); - - /* outputs */ - /* PW0 (19h), SW1 (18h), AOW1 (11h) */ - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x19, &parm); - if (smart51_enabled(codec)) - set_pin_power_state(codec, 0x1b, &parm); - update_power_state(codec, 0x18, parm); - update_power_state(codec, 0x11, parm); - - /* PW6 (22h), SW2 (26h), AOW2 (24h) */ - if (is_8ch) { - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x22, &parm); - if (smart51_enabled(codec)) - set_pin_power_state(codec, 0x1a, &parm); - update_power_state(codec, 0x26, parm); - update_power_state(codec, 0x24, parm); - } else if (codec->vendor_id == 0x11064397) { - /* PW7(23h), SW2(27h), AOW2(25h) */ - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x23, &parm); - if (smart51_enabled(codec)) - set_pin_power_state(codec, 0x1a, &parm); - update_power_state(codec, 0x27, parm); - update_power_state(codec, 0x25, parm); - } - - /* PW 3/4/7 (1ch/1dh/23h) */ - parm = AC_PWRST_D3; - /* force to D0 for internal Speaker */ - set_pin_power_state(codec, 0x1c, &parm); - set_pin_power_state(codec, 0x1d, &parm); - if (is_8ch) - set_pin_power_state(codec, 0x23, &parm); - - /* MW0 (16h), Sw3 (27h), AOW 0/3 (10h/25h) */ - update_power_state(codec, 0x16, imux_is_smixer ? AC_PWRST_D0 : parm); - update_power_state(codec, 0x10, parm); - if (is_8ch) { - update_power_state(codec, 0x25, parm); - update_power_state(codec, 0x27, parm); - } else if (codec->vendor_id == 0x11064397 && spec->gen.indep_hp_enabled) - update_power_state(codec, 0x25, parm); -} - static int patch_vt1708S(struct hda_codec *codec); static int patch_vt1708B(struct hda_codec *codec) { @@ -862,9 +674,6 @@ static int patch_vt1708B(struct hda_codec *codec) } codec->patch_ops = via_patch_ops; - - spec->set_widgets_power_state = set_widgets_power_state_vt1708B; - return 0; } @@ -907,16 +716,16 @@ static int patch_vt1708S(struct hda_codec *codec) if (get_codec_type(codec) == VT1708BCE) { kfree(codec->chip_name); codec->chip_name = kstrdup("VT1708BCE", GFP_KERNEL); - snprintf(codec->bus->card->mixername, - sizeof(codec->bus->card->mixername), + snprintf(codec->card->mixername, + sizeof(codec->card->mixername), "%s %s", codec->vendor_name, codec->chip_name); } /* correct names for VT1705 */ if (codec->vendor_id == 0x11064397) { kfree(codec->chip_name); codec->chip_name = kstrdup("VT1705", GFP_KERNEL); - snprintf(codec->bus->card->mixername, - sizeof(codec->bus->card->mixername), + snprintf(codec->card->mixername, + sizeof(codec->card->mixername), "%s %s", codec->vendor_name, codec->chip_name); } @@ -930,8 +739,6 @@ static int patch_vt1708S(struct hda_codec *codec) spec->init_verbs[spec->num_iverbs++] = vt1708S_init_verbs; codec->patch_ops = via_patch_ops; - - spec->set_widgets_power_state = set_widgets_power_state_vt1708B; return 0; } @@ -945,36 +752,6 @@ static const struct hda_verb vt1702_init_verbs[] = { { } }; -static void set_widgets_power_state_vt1702(struct hda_codec *codec) -{ - int imux_is_smixer = - snd_hda_codec_read(codec, 0x13, 0, AC_VERB_GET_CONNECT_SEL, 0x00) == 3; - unsigned int parm; - /* inputs */ - /* PW 1/2/5 (14h/15h/18h) */ - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x14, &parm); - set_pin_power_state(codec, 0x15, &parm); - set_pin_power_state(codec, 0x18, &parm); - if (imux_is_smixer) - parm = AC_PWRST_D0; /* SW0 (13h) = stereo mixer (idx 3) */ - /* SW0 (13h), AIW 0/1/2 (12h/1fh/20h) */ - update_power_state(codec, 0x13, parm); - update_power_state(codec, 0x12, parm); - update_power_state(codec, 0x1f, parm); - update_power_state(codec, 0x20, parm); - - /* outputs */ - /* PW 3/4 (16h/17h) */ - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x17, &parm); - set_pin_power_state(codec, 0x16, &parm); - /* MW0 (1ah), AOW 0/1 (10h/1dh) */ - update_power_state(codec, 0x1a, imux_is_smixer ? AC_PWRST_D0 : parm); - update_power_state(codec, 0x10, parm); - update_power_state(codec, 0x1d, parm); -} - static int patch_vt1702(struct hda_codec *codec) { struct via_spec *spec; @@ -1004,8 +781,6 @@ static int patch_vt1702(struct hda_codec *codec) spec->init_verbs[spec->num_iverbs++] = vt1702_init_verbs; codec->patch_ops = via_patch_ops; - - spec->set_widgets_power_state = set_widgets_power_state_vt1702; return 0; } @@ -1020,71 +795,6 @@ static const struct hda_verb vt1718S_init_verbs[] = { { } }; -static void set_widgets_power_state_vt1718S(struct hda_codec *codec) -{ - struct via_spec *spec = codec->spec; - int imux_is_smixer; - unsigned int parm, parm2; - /* MUX6 (1eh) = stereo mixer */ - imux_is_smixer = - snd_hda_codec_read(codec, 0x1e, 0, AC_VERB_GET_CONNECT_SEL, 0x00) == 5; - /* inputs */ - /* PW 5/6/7 (29h/2ah/2bh) */ - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x29, &parm); - set_pin_power_state(codec, 0x2a, &parm); - set_pin_power_state(codec, 0x2b, &parm); - if (imux_is_smixer) - parm = AC_PWRST_D0; - /* MUX6/7 (1eh/1fh), AIW 0/1 (10h/11h) */ - update_power_state(codec, 0x1e, parm); - update_power_state(codec, 0x1f, parm); - update_power_state(codec, 0x10, parm); - update_power_state(codec, 0x11, parm); - - /* outputs */ - /* PW3 (27h), MW2 (1ah), AOW3 (bh) */ - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x27, &parm); - update_power_state(codec, 0x1a, parm); - parm2 = parm; /* for pin 0x0b */ - - /* PW2 (26h), AOW2 (ah) */ - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x26, &parm); - if (smart51_enabled(codec)) - set_pin_power_state(codec, 0x2b, &parm); - update_power_state(codec, 0xa, parm); - - /* PW0 (24h), AOW0 (8h) */ - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x24, &parm); - if (!spec->gen.indep_hp_enabled) /* check for redirected HP */ - set_pin_power_state(codec, 0x28, &parm); - update_power_state(codec, 0x8, parm); - if (!spec->gen.indep_hp_enabled && parm2 != AC_PWRST_D3) - parm = parm2; - update_power_state(codec, 0xb, parm); - /* MW9 (21h), Mw2 (1ah), AOW0 (8h) */ - update_power_state(codec, 0x21, imux_is_smixer ? AC_PWRST_D0 : parm); - - /* PW1 (25h), AOW1 (9h) */ - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x25, &parm); - if (smart51_enabled(codec)) - set_pin_power_state(codec, 0x2a, &parm); - update_power_state(codec, 0x9, parm); - - if (spec->gen.indep_hp_enabled) { - /* PW4 (28h), MW3 (1bh), MUX1(34h), AOW4 (ch) */ - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x28, &parm); - update_power_state(codec, 0x1b, parm); - update_power_state(codec, 0x34, parm); - update_power_state(codec, 0xc, parm); - } -} - /* Add a connection to the primary DAC from AA-mixer for some codecs * This isn't listed from the raw info, but the chip has a secret connection. */ @@ -1145,9 +855,6 @@ static int patch_vt1718S(struct hda_codec *codec) spec->init_verbs[spec->num_iverbs++] = vt1718S_init_verbs; codec->patch_ops = via_patch_ops; - - spec->set_widgets_power_state = set_widgets_power_state_vt1718S; - return 0; } @@ -1187,7 +894,6 @@ static int vt1716s_dmic_put(struct snd_kcontrol *kcontrol, snd_hda_codec_write(codec, 0x26, 0, AC_VERB_SET_CONNECT_SEL, index); spec->dmic_enabled = index; - set_widgets_power_state(codec); return 1; } @@ -1222,95 +928,6 @@ static const struct hda_verb vt1716S_init_verbs[] = { { } }; -static void set_widgets_power_state_vt1716S(struct hda_codec *codec) -{ - struct via_spec *spec = codec->spec; - int imux_is_smixer; - unsigned int parm; - unsigned int mono_out, present; - /* SW0 (17h) = stereo mixer */ - imux_is_smixer = - (snd_hda_codec_read(codec, 0x17, 0, - AC_VERB_GET_CONNECT_SEL, 0x00) == 5); - /* inputs */ - /* PW 1/2/5 (1ah/1bh/1eh) */ - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x1a, &parm); - set_pin_power_state(codec, 0x1b, &parm); - set_pin_power_state(codec, 0x1e, &parm); - if (imux_is_smixer) - parm = AC_PWRST_D0; - /* SW0 (17h), AIW0(13h) */ - update_power_state(codec, 0x17, parm); - update_power_state(codec, 0x13, parm); - - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x1e, &parm); - /* PW11 (22h) */ - if (spec->dmic_enabled) - set_pin_power_state(codec, 0x22, &parm); - else - update_power_state(codec, 0x22, AC_PWRST_D3); - - /* SW2(26h), AIW1(14h) */ - update_power_state(codec, 0x26, parm); - update_power_state(codec, 0x14, parm); - - /* outputs */ - /* PW0 (19h), SW1 (18h), AOW1 (11h) */ - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x19, &parm); - /* Smart 5.1 PW2(1bh) */ - if (smart51_enabled(codec)) - set_pin_power_state(codec, 0x1b, &parm); - update_power_state(codec, 0x18, parm); - update_power_state(codec, 0x11, parm); - - /* PW7 (23h), SW3 (27h), AOW3 (25h) */ - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x23, &parm); - /* Smart 5.1 PW1(1ah) */ - if (smart51_enabled(codec)) - set_pin_power_state(codec, 0x1a, &parm); - update_power_state(codec, 0x27, parm); - - /* Smart 5.1 PW5(1eh) */ - if (smart51_enabled(codec)) - set_pin_power_state(codec, 0x1e, &parm); - update_power_state(codec, 0x25, parm); - - /* Mono out */ - /* SW4(28h)->MW1(29h)-> PW12 (2ah)*/ - present = snd_hda_jack_detect(codec, 0x1c); - - if (present) - mono_out = 0; - else { - present = snd_hda_jack_detect(codec, 0x1d); - if (!spec->gen.indep_hp_enabled && present) - mono_out = 0; - else - mono_out = 1; - } - parm = mono_out ? AC_PWRST_D0 : AC_PWRST_D3; - update_power_state(codec, 0x28, parm); - update_power_state(codec, 0x29, parm); - update_power_state(codec, 0x2a, parm); - - /* PW 3/4 (1ch/1dh) */ - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x1c, &parm); - set_pin_power_state(codec, 0x1d, &parm); - /* HP Independent Mode, power on AOW3 */ - if (spec->gen.indep_hp_enabled) - update_power_state(codec, 0x25, parm); - - /* force to D0 for internal Speaker */ - /* MW0 (16h), AOW0 (10h) */ - update_power_state(codec, 0x16, imux_is_smixer ? AC_PWRST_D0 : parm); - update_power_state(codec, 0x10, mono_out ? AC_PWRST_D0 : parm); -} - static int patch_vt1716S(struct hda_codec *codec) { struct via_spec *spec; @@ -1338,8 +955,6 @@ static int patch_vt1716S(struct hda_codec *codec) spec->mixers[spec->num_mixers++] = vt1716S_mono_out_mixer; codec->patch_ops = via_patch_ops; - - spec->set_widgets_power_state = set_widgets_power_state_vt1716S; return 0; } @@ -1365,98 +980,6 @@ static const struct hda_verb vt1802_init_verbs[] = { { } }; -static void set_widgets_power_state_vt2002P(struct hda_codec *codec) -{ - struct via_spec *spec = codec->spec; - int imux_is_smixer; - unsigned int parm; - unsigned int present; - /* MUX9 (1eh) = stereo mixer */ - imux_is_smixer = - snd_hda_codec_read(codec, 0x1e, 0, AC_VERB_GET_CONNECT_SEL, 0x00) == 3; - /* inputs */ - /* PW 5/6/7 (29h/2ah/2bh) */ - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x29, &parm); - set_pin_power_state(codec, 0x2a, &parm); - set_pin_power_state(codec, 0x2b, &parm); - parm = AC_PWRST_D0; - /* MUX9/10 (1eh/1fh), AIW 0/1 (10h/11h) */ - update_power_state(codec, 0x1e, parm); - update_power_state(codec, 0x1f, parm); - update_power_state(codec, 0x10, parm); - update_power_state(codec, 0x11, parm); - - /* outputs */ - /* AOW0 (8h)*/ - update_power_state(codec, 0x8, parm); - - if (spec->codec_type == VT1802) { - /* PW4 (28h), MW4 (18h), MUX4(38h) */ - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x28, &parm); - update_power_state(codec, 0x18, parm); - update_power_state(codec, 0x38, parm); - } else { - /* PW4 (26h), MW4 (1ch), MUX4(37h) */ - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x26, &parm); - update_power_state(codec, 0x1c, parm); - update_power_state(codec, 0x37, parm); - } - - if (spec->codec_type == VT1802) { - /* PW1 (25h), MW1 (15h), MUX1(35h), AOW1 (9h) */ - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x25, &parm); - update_power_state(codec, 0x15, parm); - update_power_state(codec, 0x35, parm); - } else { - /* PW1 (25h), MW1 (19h), MUX1(35h), AOW1 (9h) */ - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x25, &parm); - update_power_state(codec, 0x19, parm); - update_power_state(codec, 0x35, parm); - } - - if (spec->gen.indep_hp_enabled) - update_power_state(codec, 0x9, AC_PWRST_D0); - - /* Class-D */ - /* PW0 (24h), MW0(18h/14h), MUX0(34h) */ - present = snd_hda_jack_detect(codec, 0x25); - - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x24, &parm); - parm = present ? AC_PWRST_D3 : AC_PWRST_D0; - if (spec->codec_type == VT1802) - update_power_state(codec, 0x14, parm); - else - update_power_state(codec, 0x18, parm); - update_power_state(codec, 0x34, parm); - - /* Mono Out */ - present = snd_hda_jack_detect(codec, 0x26); - - parm = present ? AC_PWRST_D3 : AC_PWRST_D0; - if (spec->codec_type == VT1802) { - /* PW15 (33h), MW8(1ch), MUX8(3ch) */ - update_power_state(codec, 0x33, parm); - update_power_state(codec, 0x1c, parm); - update_power_state(codec, 0x3c, parm); - } else { - /* PW15 (31h), MW8(17h), MUX8(3bh) */ - update_power_state(codec, 0x31, parm); - update_power_state(codec, 0x17, parm); - update_power_state(codec, 0x3b, parm); - } - /* MW9 (21h) */ - if (imux_is_smixer || !is_aa_path_mute(codec)) - update_power_state(codec, 0x21, AC_PWRST_D0); - else - update_power_state(codec, 0x21, AC_PWRST_D3); -} - /* * pin fix-up */ @@ -1540,8 +1063,6 @@ static int patch_vt2002P(struct hda_codec *codec) spec->init_verbs[spec->num_iverbs++] = vt2002P_init_verbs; codec->patch_ops = via_patch_ops; - - spec->set_widgets_power_state = set_widgets_power_state_vt2002P; return 0; } @@ -1555,81 +1076,6 @@ static const struct hda_verb vt1812_init_verbs[] = { { } }; -static void set_widgets_power_state_vt1812(struct hda_codec *codec) -{ - struct via_spec *spec = codec->spec; - unsigned int parm; - unsigned int present; - /* inputs */ - /* PW 5/6/7 (29h/2ah/2bh) */ - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x29, &parm); - set_pin_power_state(codec, 0x2a, &parm); - set_pin_power_state(codec, 0x2b, &parm); - parm = AC_PWRST_D0; - /* MUX10/11 (1eh/1fh), AIW 0/1 (10h/11h) */ - update_power_state(codec, 0x1e, parm); - update_power_state(codec, 0x1f, parm); - update_power_state(codec, 0x10, parm); - update_power_state(codec, 0x11, parm); - - /* outputs */ - /* AOW0 (8h)*/ - update_power_state(codec, 0x8, AC_PWRST_D0); - - /* PW4 (28h), MW4 (18h), MUX4(38h) */ - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x28, &parm); - update_power_state(codec, 0x18, parm); - update_power_state(codec, 0x38, parm); - - /* PW1 (25h), MW1 (15h), MUX1(35h), AOW1 (9h) */ - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x25, &parm); - update_power_state(codec, 0x15, parm); - update_power_state(codec, 0x35, parm); - if (spec->gen.indep_hp_enabled) - update_power_state(codec, 0x9, AC_PWRST_D0); - - /* Internal Speaker */ - /* PW0 (24h), MW0(14h), MUX0(34h) */ - present = snd_hda_jack_detect(codec, 0x25); - - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x24, &parm); - if (present) { - update_power_state(codec, 0x14, AC_PWRST_D3); - update_power_state(codec, 0x34, AC_PWRST_D3); - } else { - update_power_state(codec, 0x14, AC_PWRST_D0); - update_power_state(codec, 0x34, AC_PWRST_D0); - } - - - /* Mono Out */ - /* PW13 (31h), MW13(1ch), MUX13(3ch), MW14(3eh) */ - present = snd_hda_jack_detect(codec, 0x28); - - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x31, &parm); - if (present) { - update_power_state(codec, 0x1c, AC_PWRST_D3); - update_power_state(codec, 0x3c, AC_PWRST_D3); - update_power_state(codec, 0x3e, AC_PWRST_D3); - } else { - update_power_state(codec, 0x1c, AC_PWRST_D0); - update_power_state(codec, 0x3c, AC_PWRST_D0); - update_power_state(codec, 0x3e, AC_PWRST_D0); - } - - /* PW15 (33h), MW15 (1dh), MUX15(3dh) */ - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x33, &parm); - update_power_state(codec, 0x1d, parm); - update_power_state(codec, 0x3d, parm); - -} - /* patch for vt1812 */ static int patch_vt1812(struct hda_codec *codec) { @@ -1656,8 +1102,6 @@ static int patch_vt1812(struct hda_codec *codec) spec->init_verbs[spec->num_iverbs++] = vt1812_init_verbs; codec->patch_ops = via_patch_ops; - - spec->set_widgets_power_state = set_widgets_power_state_vt1812; return 0; } @@ -1673,84 +1117,6 @@ static const struct hda_verb vt3476_init_verbs[] = { { } }; -static void set_widgets_power_state_vt3476(struct hda_codec *codec) -{ - struct via_spec *spec = codec->spec; - int imux_is_smixer; - unsigned int parm, parm2; - /* MUX10 (1eh) = stereo mixer */ - imux_is_smixer = - snd_hda_codec_read(codec, 0x1e, 0, AC_VERB_GET_CONNECT_SEL, 0x00) == 4; - /* inputs */ - /* PW 5/6/7 (29h/2ah/2bh) */ - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x29, &parm); - set_pin_power_state(codec, 0x2a, &parm); - set_pin_power_state(codec, 0x2b, &parm); - if (imux_is_smixer) - parm = AC_PWRST_D0; - /* MUX10/11 (1eh/1fh), AIW 0/1 (10h/11h) */ - update_power_state(codec, 0x1e, parm); - update_power_state(codec, 0x1f, parm); - update_power_state(codec, 0x10, parm); - update_power_state(codec, 0x11, parm); - - /* outputs */ - /* PW3 (27h), MW3(37h), AOW3 (bh) */ - if (spec->codec_type == VT1705CF) { - parm = AC_PWRST_D3; - update_power_state(codec, 0x27, parm); - update_power_state(codec, 0x37, parm); - } else { - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x27, &parm); - update_power_state(codec, 0x37, parm); - } - - /* PW2 (26h), MW2(36h), AOW2 (ah) */ - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x26, &parm); - update_power_state(codec, 0x36, parm); - if (smart51_enabled(codec)) { - /* PW7(2bh), MW7(3bh), MUX7(1Bh) */ - set_pin_power_state(codec, 0x2b, &parm); - update_power_state(codec, 0x3b, parm); - update_power_state(codec, 0x1b, parm); - } - update_conv_power_state(codec, 0xa, parm, 2); - - /* PW1 (25h), MW1(35h), AOW1 (9h) */ - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x25, &parm); - update_power_state(codec, 0x35, parm); - if (smart51_enabled(codec)) { - /* PW6(2ah), MW6(3ah), MUX6(1ah) */ - set_pin_power_state(codec, 0x2a, &parm); - update_power_state(codec, 0x3a, parm); - update_power_state(codec, 0x1a, parm); - } - update_conv_power_state(codec, 0x9, parm, 1); - - /* PW4 (28h), MW4 (38h), MUX4(18h), AOW3(bh)/AOW0(8h) */ - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x28, &parm); - update_power_state(codec, 0x38, parm); - update_power_state(codec, 0x18, parm); - if (spec->gen.indep_hp_enabled) - update_conv_power_state(codec, 0xb, parm, 3); - parm2 = parm; /* for pin 0x0b */ - - /* PW0 (24h), MW0(34h), MW9(3fh), AOW0 (8h) */ - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x24, &parm); - update_power_state(codec, 0x34, parm); - if (!spec->gen.indep_hp_enabled && parm2 != AC_PWRST_D3) - parm = parm2; - update_conv_power_state(codec, 0x8, parm, 0); - /* MW9 (21h), Mw2 (1ah), AOW0 (8h) */ - update_power_state(codec, 0x3f, imux_is_smixer ? AC_PWRST_D0 : parm); -} - static int patch_vt3476(struct hda_codec *codec) { struct via_spec *spec; @@ -1774,9 +1140,6 @@ static int patch_vt3476(struct hda_codec *codec) spec->init_verbs[spec->num_iverbs++] = vt3476_init_verbs; codec->patch_ops = via_patch_ops; - - spec->set_widgets_power_state = set_widgets_power_state_vt3476; - return 0; } @@ -1884,23 +1247,11 @@ static const struct hda_codec_preset snd_hda_preset_via[] = { MODULE_ALIAS("snd-hda-codec-id:1106*"); -static struct hda_codec_preset_list via_list = { +static struct hda_codec_driver via_driver = { .preset = snd_hda_preset_via, - .owner = THIS_MODULE, }; MODULE_LICENSE("GPL"); MODULE_DESCRIPTION("VIA HD-audio codec"); -static int __init patch_via_init(void) -{ - return snd_hda_add_codec_preset(&via_list); -} - -static void __exit patch_via_exit(void) -{ - snd_hda_delete_codec_preset(&via_list); -} - -module_init(patch_via_init) -module_exit(patch_via_exit) +module_hda_codec_driver(via_driver); diff --git a/sound/pci/ice1712/wtm.c b/sound/pci/ice1712/wtm.c index bcf30a3..9906119 100644 --- a/sound/pci/ice1712/wtm.c +++ b/sound/pci/ice1712/wtm.c @@ -29,12 +29,19 @@ #include <linux/interrupt.h> #include <linux/init.h> #include <sound/core.h> +#include <sound/tlv.h> +#include <linux/slab.h> #include "ice1712.h" #include "envy24ht.h" #include "wtm.h" #include "stac946x.h" +struct wtm_spec { + /* rate change needs atomic mute/unmute of all dacs*/ + struct mutex mute_mutex; +}; + /* * 2*ADC 6*DAC no1 ringbuffer r/w on i2c bus @@ -68,15 +75,65 @@ static inline unsigned char stac9460_2_get(struct snd_ice1712 *ice, int reg) /* * DAC mute control */ +static void stac9460_dac_mute_all(struct snd_ice1712 *ice, unsigned char mute, + unsigned short int *change_mask) +{ + unsigned char new, old; + int id, idx, change; + + /*stac9460 1*/ + for (id = 0; id < 7; id++) { + if (*change_mask & (0x01 << id)) { + if (id == 0) + idx = STAC946X_MASTER_VOLUME; + else + idx = STAC946X_LF_VOLUME - 1 + id; + old = stac9460_get(ice, idx); + new = (~mute << 7 & 0x80) | (old & ~0x80); + change = (new != old); + if (change) { + stac9460_put(ice, idx, new); + *change_mask = *change_mask | (0x01 << id); + } else { + *change_mask = *change_mask & ~(0x01 << id); + } + } + } + + /*stac9460 2*/ + for (id = 0; id < 3; id++) { + if (*change_mask & (0x01 << (id + 7))) { + if (id == 0) + idx = STAC946X_MASTER_VOLUME; + else + idx = STAC946X_LF_VOLUME - 1 + id; + old = stac9460_2_get(ice, idx); + new = (~mute << 7 & 0x80) | (old & ~0x80); + change = (new != old); + if (change) { + stac9460_2_put(ice, idx, new); + *change_mask = *change_mask | (0x01 << id); + } else { + *change_mask = *change_mask & ~(0x01 << id); + } + } + } +} + + + #define stac9460_dac_mute_info snd_ctl_boolean_mono_info static int stac9460_dac_mute_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); + struct wtm_spec *spec = ice->spec; unsigned char val; int idx, id; + mutex_lock(&spec->mute_mutex); + if (kcontrol->private_value) { idx = STAC946X_MASTER_VOLUME; id = 0; @@ -89,6 +146,8 @@ static int stac9460_dac_mute_get(struct snd_kcontrol *kcontrol, else val = stac9460_2_get(ice, idx - 6); ucontrol->value.integer.value[0] = (~val >> 7) & 0x1; + + mutex_unlock(&spec->mute_mutex); return 0; } @@ -338,8 +397,14 @@ static int stac9460_adc_vol_put(struct snd_kcontrol *kcontrol, /* * MIC / LINE switch fonction */ +static int stac9460_mic_sw_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + static const char * const texts[2] = { "Line In", "Mic" }; + + return snd_ctl_enum_info(uinfo, 1, 2, texts); +} -#define stac9460_mic_sw_info snd_ctl_boolean_mono_info static int stac9460_mic_sw_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) @@ -353,7 +418,7 @@ static int stac9460_mic_sw_get(struct snd_kcontrol *kcontrol, val = stac9460_get(ice, STAC946X_GENERAL_PURPOSE); else val = stac9460_2_get(ice, STAC946X_GENERAL_PURPOSE); - ucontrol->value.integer.value[0] = ~val>>7 & 0x1; + ucontrol->value.enumerated.item[0] = (val >> 7) & 0x1; return 0; } @@ -369,7 +434,7 @@ static int stac9460_mic_sw_put(struct snd_kcontrol *kcontrol, old = stac9460_get(ice, STAC946X_GENERAL_PURPOSE); else old = stac9460_2_get(ice, STAC946X_GENERAL_PURPOSE); - new = (~ucontrol->value.integer.value[0] << 7 & 0x80) | (old & ~0x80); + new = (ucontrol->value.enumerated.item[0] << 7 & 0x80) | (old & ~0x80); change = (new != old); if (change) { if (id == 0) @@ -380,17 +445,63 @@ static int stac9460_mic_sw_put(struct snd_kcontrol *kcontrol, return change; } + +/* + * Handler for setting correct codec rate - called when rate change is detected + */ +static void stac9460_set_rate_val(struct snd_ice1712 *ice, unsigned int rate) +{ + unsigned char old, new; + unsigned short int changed; + struct wtm_spec *spec = ice->spec; + + if (rate == 0) /* no hint - S/PDIF input is master, simply return */ + return; + else if (rate <= 48000) + new = 0x08; /* 256x, base rate mode */ + else if (rate <= 96000) + new = 0x11; /* 256x, mid rate mode */ + else + new = 0x12; /* 128x, high rate mode */ + + old = stac9460_get(ice, STAC946X_MASTER_CLOCKING); + if (old == new) + return; + /* change detected, setting master clock, muting first */ + /* due to possible conflicts with mute controls - mutexing */ + mutex_lock(&spec->mute_mutex); + /* we have to remember current mute status for each DAC */ + changed = 0xFFFF; + stac9460_dac_mute_all(ice, 0, &changed); + /*printk(KERN_DEBUG "Rate change: %d, new MC: 0x%02x\n", rate, new);*/ + stac9460_put(ice, STAC946X_MASTER_CLOCKING, new); + stac9460_2_put(ice, STAC946X_MASTER_CLOCKING, new); + udelay(10); + /* unmuting - only originally unmuted dacs - + * i.e. those changed when muting */ + stac9460_dac_mute_all(ice, 1, &changed); + mutex_unlock(&spec->mute_mutex); +} + + +/*Limits value in dB for fader*/ +static const DECLARE_TLV_DB_SCALE(db_scale_dac, -19125, 75, 0); +static const DECLARE_TLV_DB_SCALE(db_scale_adc, 0, 150, 0); + /* * Control tabs */ static struct snd_kcontrol_new stac9640_controls[] = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE | + SNDRV_CTL_ELEM_ACCESS_TLV_READ), .name = "Master Playback Switch", .info = stac9460_dac_mute_info, .get = stac9460_dac_mute_get, .put = stac9460_dac_mute_put, - .private_value = 1 + .private_value = 1, + .tlv = { .p = db_scale_dac } }, { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, @@ -402,7 +513,7 @@ static struct snd_kcontrol_new stac9640_controls[] = { }, { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "MIC/Line switch", + .name = "MIC/Line Input Enum", .count = 2, .info = stac9460_mic_sw_info, .get = stac9460_mic_sw_get, @@ -419,11 +530,15 @@ static struct snd_kcontrol_new stac9640_controls[] = { }, { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE | + SNDRV_CTL_ELEM_ACCESS_TLV_READ), + .name = "DAC Volume", .count = 8, .info = stac9460_dac_vol_info, .get = stac9460_dac_vol_get, .put = stac9460_dac_vol_put, + .tlv = { .p = db_scale_dac } }, { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, @@ -435,12 +550,15 @@ static struct snd_kcontrol_new stac9640_controls[] = { }, { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE | + SNDRV_CTL_ELEM_ACCESS_TLV_READ), + .name = "ADC Volume", .count = 2, .info = stac9460_adc_vol_info, .get = stac9460_adc_vol_get, .put = stac9460_adc_vol_put, - + .tlv = { .p = db_scale_adc } } }; @@ -463,41 +581,53 @@ static int wtm_add_controls(struct snd_ice1712 *ice) static int wtm_init(struct snd_ice1712 *ice) { - static unsigned short stac_inits_prodigy[] = { + static unsigned short stac_inits_wtm[] = { STAC946X_RESET, 0, + STAC946X_MASTER_CLOCKING, 0x11, (unsigned short)-1 }; unsigned short *p; + struct wtm_spec *spec; /*WTM 192M*/ ice->num_total_dacs = 8; ice->num_total_adcs = 4; ice->force_rdma1 = 1; + /*init mutex for dac mute conflict*/ + spec = kzalloc(sizeof(*spec), GFP_KERNEL); + if (!spec) + return -ENOMEM; + ice->spec = spec; + mutex_init(&spec->mute_mutex); + + /*initialize codec*/ - p = stac_inits_prodigy; + p = stac_inits_wtm; for (; *p != (unsigned short)-1; p += 2) { stac9460_put(ice, p[0], p[1]); stac9460_2_put(ice, p[0], p[1]); } + ice->gpio.set_pro_rate = stac9460_set_rate_val; return 0; } static unsigned char wtm_eeprom[] = { - 0x47, /*SYSCONF: clock 192KHz, 4ADC, 8DAC */ - 0x80, /* ACLINK : I2S */ - 0xf8, /* I2S: vol; 96k, 24bit, 192k */ - 0xc1 /*SPDIF: out-en, spidf ext out*/, - 0x9f, /* GPIO_DIR */ - 0xff, /* GPIO_DIR1 */ - 0x7f, /* GPIO_DIR2 */ - 0x9f, /* GPIO_MASK */ - 0xff, /* GPIO_MASK1 */ - 0x7f, /* GPIO_MASK2 */ - 0x16, /* GPIO_STATE */ - 0x80, /* GPIO_STATE1 */ - 0x00, /* GPIO_STATE2 */ + [ICE_EEP2_SYSCONF] = 0x67, /*SYSCONF: clock 192KHz, mpu401, + 4ADC, 8DAC */ + [ICE_EEP2_ACLINK] = 0x80, /* ACLINK : I2S */ + [ICE_EEP2_I2S] = 0xf8, /* I2S: vol; 96k, 24bit, 192k */ + [ICE_EEP2_SPDIF] = 0xc1, /*SPDIF: out-en, spidf ext out*/ + [ICE_EEP2_GPIO_DIR] = 0x9f, + [ICE_EEP2_GPIO_DIR1] = 0xff, + [ICE_EEP2_GPIO_DIR2] = 0x7f, + [ICE_EEP2_GPIO_MASK] = 0x9f, + [ICE_EEP2_GPIO_MASK1] = 0xff, + [ICE_EEP2_GPIO_MASK2] = 0x7f, + [ICE_EEP2_GPIO_STATE] = 0x16, + [ICE_EEP2_GPIO_STATE1] = 0x80, + [ICE_EEP2_GPIO_STATE2] = 0x00, }; diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index ca67f89..cb666c7 100644 --- a/sound/pci/rme9652/hdspm.c +++ b/sound/pci/rme9652/hdspm.c @@ -6043,23 +6043,30 @@ hdspm_hw_constraints_aes32_sample_rates = { .mask = 0 }; -static int snd_hdspm_playback_open(struct snd_pcm_substream *substream) +static int snd_hdspm_open(struct snd_pcm_substream *substream) { struct hdspm *hdspm = snd_pcm_substream_chip(substream); struct snd_pcm_runtime *runtime = substream->runtime; + bool playback = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK); spin_lock_irq(&hdspm->lock); - snd_pcm_set_sync(substream); + runtime->hw = (playback) ? snd_hdspm_playback_subinfo : + snd_hdspm_capture_subinfo; + if (playback) { + if (hdspm->capture_substream == NULL) + hdspm_stop_audio(hdspm); - runtime->hw = snd_hdspm_playback_subinfo; - - if (hdspm->capture_substream == NULL) - hdspm_stop_audio(hdspm); + hdspm->playback_pid = current->pid; + hdspm->playback_substream = substream; + } else { + if (hdspm->playback_substream == NULL) + hdspm_stop_audio(hdspm); - hdspm->playback_pid = current->pid; - hdspm->playback_substream = substream; + hdspm->capture_pid = current->pid; + hdspm->capture_substream = substream; + } spin_unlock_irq(&hdspm->lock); @@ -6094,108 +6101,42 @@ static int snd_hdspm_playback_open(struct snd_pcm_substream *substream) &hdspm_hw_constraints_aes32_sample_rates); } else { snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, - snd_hdspm_hw_rule_rate_out_channels, hdspm, + (playback ? + snd_hdspm_hw_rule_rate_out_channels : + snd_hdspm_hw_rule_rate_in_channels), hdspm, SNDRV_PCM_HW_PARAM_CHANNELS, -1); } snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, - snd_hdspm_hw_rule_out_channels, hdspm, + (playback ? snd_hdspm_hw_rule_out_channels : + snd_hdspm_hw_rule_in_channels), hdspm, SNDRV_PCM_HW_PARAM_CHANNELS, -1); snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, - snd_hdspm_hw_rule_out_channels_rate, hdspm, + (playback ? snd_hdspm_hw_rule_out_channels_rate : + snd_hdspm_hw_rule_in_channels_rate), hdspm, SNDRV_PCM_HW_PARAM_RATE, -1); return 0; } -static int snd_hdspm_playback_release(struct snd_pcm_substream *substream) +static int snd_hdspm_release(struct snd_pcm_substream *substream) { struct hdspm *hdspm = snd_pcm_substream_chip(substream); + bool playback = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK); spin_lock_irq(&hdspm->lock); - hdspm->playback_pid = -1; - hdspm->playback_substream = NULL; - - spin_unlock_irq(&hdspm->lock); - - return 0; -} - - -static int snd_hdspm_capture_open(struct snd_pcm_substream *substream) -{ - struct hdspm *hdspm = snd_pcm_substream_chip(substream); - struct snd_pcm_runtime *runtime = substream->runtime; - - spin_lock_irq(&hdspm->lock); - snd_pcm_set_sync(substream); - runtime->hw = snd_hdspm_capture_subinfo; - - if (hdspm->playback_substream == NULL) - hdspm_stop_audio(hdspm); - - hdspm->capture_pid = current->pid; - hdspm->capture_substream = substream; - - spin_unlock_irq(&hdspm->lock); - - snd_pcm_hw_constraint_msbits(runtime, 0, 32, 24); - snd_pcm_hw_constraint_pow2(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_SIZE); - - switch (hdspm->io_type) { - case AIO: - case RayDAT: - snd_pcm_hw_constraint_minmax(runtime, - SNDRV_PCM_HW_PARAM_PERIOD_SIZE, - 32, 4096); - snd_pcm_hw_constraint_minmax(runtime, - SNDRV_PCM_HW_PARAM_BUFFER_SIZE, - 16384, 16384); - break; - - default: - snd_pcm_hw_constraint_minmax(runtime, - SNDRV_PCM_HW_PARAM_PERIOD_SIZE, - 64, 8192); - snd_pcm_hw_constraint_minmax(runtime, - SNDRV_PCM_HW_PARAM_PERIODS, - 2, 2); - break; - } - - if (AES32 == hdspm->io_type) { - runtime->hw.rates |= SNDRV_PCM_RATE_KNOT; - snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, - &hdspm_hw_constraints_aes32_sample_rates); + if (playback) { + hdspm->playback_pid = -1; + hdspm->playback_substream = NULL; } else { - snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, - snd_hdspm_hw_rule_rate_in_channels, hdspm, - SNDRV_PCM_HW_PARAM_CHANNELS, -1); + hdspm->capture_pid = -1; + hdspm->capture_substream = NULL; } - snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, - snd_hdspm_hw_rule_in_channels, hdspm, - SNDRV_PCM_HW_PARAM_CHANNELS, -1); - - snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, - snd_hdspm_hw_rule_in_channels_rate, hdspm, - SNDRV_PCM_HW_PARAM_RATE, -1); - - return 0; -} - -static int snd_hdspm_capture_release(struct snd_pcm_substream *substream) -{ - struct hdspm *hdspm = snd_pcm_substream_chip(substream); - - spin_lock_irq(&hdspm->lock); - - hdspm->capture_pid = -1; - hdspm->capture_substream = NULL; - spin_unlock_irq(&hdspm->lock); + return 0; } @@ -6413,21 +6354,9 @@ static int snd_hdspm_hwdep_ioctl(struct snd_hwdep *hw, struct file *file, return 0; } -static struct snd_pcm_ops snd_hdspm_playback_ops = { - .open = snd_hdspm_playback_open, - .close = snd_hdspm_playback_release, - .ioctl = snd_hdspm_ioctl, - .hw_params = snd_hdspm_hw_params, - .hw_free = snd_hdspm_hw_free, - .prepare = snd_hdspm_prepare, - .trigger = snd_hdspm_trigger, - .pointer = snd_hdspm_hw_pointer, - .page = snd_pcm_sgbuf_ops_page, -}; - -static struct snd_pcm_ops snd_hdspm_capture_ops = { - .open = snd_hdspm_capture_open, - .close = snd_hdspm_capture_release, +static struct snd_pcm_ops snd_hdspm_ops = { + .open = snd_hdspm_open, + .close = snd_hdspm_release, .ioctl = snd_hdspm_ioctl, .hw_params = snd_hdspm_hw_params, .hw_free = snd_hdspm_hw_free, @@ -6521,9 +6450,9 @@ static int snd_hdspm_create_pcm(struct snd_card *card, strcpy(pcm->name, hdspm->card_name); snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, - &snd_hdspm_playback_ops); + &snd_hdspm_ops); snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, - &snd_hdspm_capture_ops); + &snd_hdspm_ops); pcm->info_flags = SNDRV_PCM_INFO_JOINT_DUPLEX; diff --git a/sound/soc/atmel/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c index fb0b7e8b..841d059 100644 --- a/sound/soc/atmel/atmel_ssc_dai.c +++ b/sound/soc/atmel/atmel_ssc_dai.c @@ -187,6 +187,94 @@ static irqreturn_t atmel_ssc_interrupt(int irq, void *dev_id) return IRQ_HANDLED; } +/* + * When the bit clock is input, limit the maximum rate according to the + * Serial Clock Ratio Considerations section from the SSC documentation: + * + * The Transmitter and the Receiver can be programmed to operate + * with the clock signals provided on either the TK or RK pins. + * This allows the SSC to support many slave-mode data transfers. + * In this case, the maximum clock speed allowed on the RK pin is: + * - Peripheral clock divided by 2 if Receiver Frame Synchro is input + * - Peripheral clock divided by 3 if Receiver Frame Synchro is output + * In addition, the maximum clock speed allowed on the TK pin is: + * - Peripheral clock divided by 6 if Transmit Frame Synchro is input + * - Peripheral clock divided by 2 if Transmit Frame Synchro is output + * + * When the bit clock is output, limit the rate according to the + * SSC divider restrictions. + */ +static int atmel_ssc_hw_rule_rate(struct snd_pcm_hw_params *params, + struct snd_pcm_hw_rule *rule) +{ + struct atmel_ssc_info *ssc_p = rule->private; + struct ssc_device *ssc = ssc_p->ssc; + struct snd_interval *i = hw_param_interval(params, rule->var); + struct snd_interval t; + struct snd_ratnum r = { + .den_min = 1, + .den_max = 4095, + .den_step = 1, + }; + unsigned int num = 0, den = 0; + int frame_size; + int mck_div = 2; + int ret; + + frame_size = snd_soc_params_to_frame_size(params); + if (frame_size < 0) + return frame_size; + + switch (ssc_p->daifmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFS: + if ((ssc_p->dir_mask & SSC_DIR_MASK_CAPTURE) + && ssc->clk_from_rk_pin) + /* Receiver Frame Synchro (i.e. capture) + * is output (format is _CFS) and the RK pin + * is used for input (format is _CBM_). + */ + mck_div = 3; + break; + + case SND_SOC_DAIFMT_CBM_CFM: + if ((ssc_p->dir_mask & SSC_DIR_MASK_PLAYBACK) + && !ssc->clk_from_rk_pin) + /* Transmit Frame Synchro (i.e. playback) + * is input (format is _CFM) and the TK pin + * is used for input (format _CBM_ but not + * using the RK pin). + */ + mck_div = 6; + break; + } + + switch (ssc_p->daifmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + r.num = ssc_p->mck_rate / mck_div / frame_size; + + ret = snd_interval_ratnum(i, 1, &r, &num, &den); + if (ret >= 0 && den && rule->var == SNDRV_PCM_HW_PARAM_RATE) { + params->rate_num = num; + params->rate_den = den; + } + break; + + case SND_SOC_DAIFMT_CBM_CFS: + case SND_SOC_DAIFMT_CBM_CFM: + t.min = 8000; + t.max = ssc_p->mck_rate / mck_div / frame_size; + t.openmin = t.openmax = 0; + t.integer = 0; + ret = snd_interval_refine(i, &t); + break; + + default: + ret = -EINVAL; + break; + } + + return ret; +} /*-------------------------------------------------------------------------*\ * DAI functions @@ -200,6 +288,7 @@ static int atmel_ssc_startup(struct snd_pcm_substream *substream, struct atmel_ssc_info *ssc_p = &ssc_info[dai->id]; struct atmel_pcm_dma_params *dma_params; int dir, dir_mask; + int ret; pr_debug("atmel_ssc_startup: SSC_SR=0x%u\n", ssc_readl(ssc_p->ssc->regs, SR)); @@ -207,6 +296,7 @@ static int atmel_ssc_startup(struct snd_pcm_substream *substream, /* Enable PMC peripheral clock for this SSC */ pr_debug("atmel_ssc_dai: Starting clock\n"); clk_enable(ssc_p->ssc->clk); + ssc_p->mck_rate = clk_get_rate(ssc_p->ssc->clk); /* Reset the SSC to keep it at a clean status */ ssc_writel(ssc_p->ssc->regs, CR, SSC_BIT(CR_SWRST)); @@ -219,6 +309,17 @@ static int atmel_ssc_startup(struct snd_pcm_substream *substream, dir_mask = SSC_DIR_MASK_CAPTURE; } + ret = snd_pcm_hw_rule_add(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, + atmel_ssc_hw_rule_rate, + ssc_p, + SNDRV_PCM_HW_PARAM_FRAME_BITS, + SNDRV_PCM_HW_PARAM_CHANNELS, -1); + if (ret < 0) { + dev_err(dai->dev, "Failed to specify rate rule: %d\n", ret); + return ret; + } + dma_params = &ssc_dma_params[dai->id][dir]; dma_params->ssc = ssc_p->ssc; dma_params->substream = substream; @@ -783,8 +884,6 @@ static int atmel_ssc_resume(struct snd_soc_dai *cpu_dai) # define atmel_ssc_resume NULL #endif /* CONFIG_PM */ -#define ATMEL_SSC_RATES (SNDRV_PCM_RATE_8000_96000) - #define ATMEL_SSC_FORMATS (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE |\ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) @@ -804,12 +903,16 @@ static struct snd_soc_dai_driver atmel_ssc_dai = { .playback = { .channels_min = 1, .channels_max = 2, - .rates = ATMEL_SSC_RATES, + .rates = SNDRV_PCM_RATE_CONTINUOUS, + .rate_min = 8000, + .rate_max = 384000, .formats = ATMEL_SSC_FORMATS,}, .capture = { .channels_min = 1, .channels_max = 2, - .rates = ATMEL_SSC_RATES, + .rates = SNDRV_PCM_RATE_CONTINUOUS, + .rate_min = 8000, + .rate_max = 384000, .formats = ATMEL_SSC_FORMATS,}, .ops = &atmel_ssc_dai_ops, }; diff --git a/sound/soc/atmel/atmel_ssc_dai.h b/sound/soc/atmel/atmel_ssc_dai.h index b1f08d5..80b1538 100644 --- a/sound/soc/atmel/atmel_ssc_dai.h +++ b/sound/soc/atmel/atmel_ssc_dai.h @@ -115,6 +115,7 @@ struct atmel_ssc_info { unsigned short rcmr_period; struct atmel_pcm_dma_params *dma_params[2]; struct atmel_ssc_state ssc_state; + unsigned long mck_rate; }; int atmel_ssc_set_audio(int ssc_id); diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index ea9f0e3..0bddd92 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -141,7 +141,8 @@ config SND_SOC_ALL_CODECS select SND_SOC_WM8770 if SPI_MASTER select SND_SOC_WM8776 if SND_SOC_I2C_AND_SPI select SND_SOC_WM8782 - select SND_SOC_WM8804 if SND_SOC_I2C_AND_SPI + select SND_SOC_WM8804_I2C if I2C + select SND_SOC_WM8804_SPI if SPI_MASTER select SND_SOC_WM8900 if I2C select SND_SOC_WM8903 if I2C select SND_SOC_WM8904 if I2C @@ -744,8 +745,19 @@ config SND_SOC_WM8782 tristate config SND_SOC_WM8804 - tristate "Wolfson Microelectronics WM8804 S/PDIF transceiver" - depends on SND_SOC_I2C_AND_SPI + tristate + +config SND_SOC_WM8804_I2C + tristate "Wolfson Microelectronics WM8804 S/PDIF transceiver I2C" + depends on I2C + select SND_SOC_WM8804 + select REGMAP_I2C + +config SND_SOC_WM8804_SPI + tristate "Wolfson Microelectronics WM8804 S/PDIF transceiver SPI" + depends on SPI_MASTER + select SND_SOC_WM8804 + select REGMAP_SPI config SND_SOC_WM8900 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 69b8666..7acb6c1 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -145,6 +145,8 @@ snd-soc-wm8770-objs := wm8770.o snd-soc-wm8776-objs := wm8776.o snd-soc-wm8782-objs := wm8782.o snd-soc-wm8804-objs := wm8804.o +snd-soc-wm8804-i2c-objs := wm8804-i2c.o +snd-soc-wm8804-spi-objs := wm8804-spi.o snd-soc-wm8900-objs := wm8900.o snd-soc-wm8903-objs := wm8903.o snd-soc-wm8904-objs := wm8904.o @@ -323,6 +325,8 @@ obj-$(CONFIG_SND_SOC_WM8770) += snd-soc-wm8770.o obj-$(CONFIG_SND_SOC_WM8776) += snd-soc-wm8776.o obj-$(CONFIG_SND_SOC_WM8782) += snd-soc-wm8782.o obj-$(CONFIG_SND_SOC_WM8804) += snd-soc-wm8804.o +obj-$(CONFIG_SND_SOC_WM8804_I2C) += snd-soc-wm8804-i2c.o +obj-$(CONFIG_SND_SOC_WM8804_SPI) += snd-soc-wm8804-spi.o obj-$(CONFIG_SND_SOC_WM8900) += snd-soc-wm8900.o obj-$(CONFIG_SND_SOC_WM8903) += snd-soc-wm8903.o obj-$(CONFIG_SND_SOC_WM8904) += snd-soc-wm8904.o diff --git a/sound/soc/codecs/adau1977.c b/sound/soc/codecs/adau1977.c index 70ab357..7ad8e15 100644 --- a/sound/soc/codecs/adau1977.c +++ b/sound/soc/codecs/adau1977.c @@ -938,22 +938,15 @@ int adau1977_probe(struct device *dev, struct regmap *regmap, adau1977->dvdd_reg = NULL; } - adau1977->reset_gpio = devm_gpiod_get(dev, "reset"); - if (IS_ERR(adau1977->reset_gpio)) { - ret = PTR_ERR(adau1977->reset_gpio); - if (ret != -ENOENT && ret != -ENOSYS) - return PTR_ERR(adau1977->reset_gpio); - adau1977->reset_gpio = NULL; - } + adau1977->reset_gpio = devm_gpiod_get_optional(dev, "reset", + GPIOD_OUT_LOW); + if (IS_ERR(adau1977->reset_gpio)) + return PTR_ERR(adau1977->reset_gpio); dev_set_drvdata(dev, adau1977); - if (adau1977->reset_gpio) { - ret = gpiod_direction_output(adau1977->reset_gpio, 0); - if (ret) - return ret; + if (adau1977->reset_gpio) ndelay(100); - } ret = adau1977_power_enable(adau1977); if (ret) diff --git a/sound/soc/codecs/cs35l32.c b/sound/soc/codecs/cs35l32.c index f2b8aad..60598b2 100644 --- a/sound/soc/codecs/cs35l32.c +++ b/sound/soc/codecs/cs35l32.c @@ -437,20 +437,13 @@ static int cs35l32_i2c_probe(struct i2c_client *i2c_client, } /* Reset the Device */ - cs35l32->reset_gpio = devm_gpiod_get(&i2c_client->dev, - "reset-gpios"); - if (IS_ERR(cs35l32->reset_gpio)) { - ret = PTR_ERR(cs35l32->reset_gpio); - if (ret != -ENOENT && ret != -ENOSYS) - return ret; - - cs35l32->reset_gpio = NULL; - } else { - ret = gpiod_direction_output(cs35l32->reset_gpio, 0); - if (ret) - return ret; + cs35l32->reset_gpio = devm_gpiod_get_optional(&i2c_client->dev, + "reset", GPIOD_OUT_LOW); + if (IS_ERR(cs35l32->reset_gpio)) + return PTR_ERR(cs35l32->reset_gpio); + + if (cs35l32->reset_gpio) gpiod_set_value_cansleep(cs35l32->reset_gpio, 1); - } /* initialize codec */ ret = regmap_read(cs35l32->regmap, CS35L32_DEVID_AB, ®); diff --git a/sound/soc/codecs/cs4265.c b/sound/soc/codecs/cs4265.c index ce60868..cac48dd 100644 --- a/sound/soc/codecs/cs4265.c +++ b/sound/soc/codecs/cs4265.c @@ -605,21 +605,14 @@ static int cs4265_i2c_probe(struct i2c_client *i2c_client, return ret; } - cs4265->reset_gpio = devm_gpiod_get(&i2c_client->dev, - "reset-gpios"); - if (IS_ERR(cs4265->reset_gpio)) { - ret = PTR_ERR(cs4265->reset_gpio); - if (ret != -ENOENT && ret != -ENOSYS) - return ret; - - cs4265->reset_gpio = NULL; - } else { - ret = gpiod_direction_output(cs4265->reset_gpio, 0); - if (ret) - return ret; + cs4265->reset_gpio = devm_gpiod_get_optional(&i2c_client->dev, + "reset", GPIOD_OUT_LOW); + if (IS_ERR(cs4265->reset_gpio)) + return PTR_ERR(cs4265->reset_gpio); + + if (cs4265->reset_gpio) { mdelay(1); gpiod_set_value_cansleep(cs4265->reset_gpio, 1); - } i2c_set_clientdata(i2c_client, cs4265); diff --git a/sound/soc/codecs/max98357a.c b/sound/soc/codecs/max98357a.c index e9e6efb..bf3e933 100644 --- a/sound/soc/codecs/max98357a.c +++ b/sound/soc/codecs/max98357a.c @@ -26,8 +26,6 @@ #include <sound/soc-dai.h> #include <sound/soc-dapm.h> -#define DRV_NAME "max98357a" - static int max98357a_daiops_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *dai) { @@ -87,9 +85,9 @@ static struct snd_soc_dai_ops max98357a_dai_ops = { }; static struct snd_soc_dai_driver max98357a_dai_driver = { - .name = DRV_NAME, + .name = "HiFi", .playback = { - .stream_name = DRV_NAME "-playback", + .stream_name = "HiFi Playback", .formats = SNDRV_PCM_FMTBIT_S16 | SNDRV_PCM_FMTBIT_S24 | SNDRV_PCM_FMTBIT_S32, @@ -127,7 +125,7 @@ static int max98357a_platform_remove(struct platform_device *pdev) #ifdef CONFIG_OF static const struct of_device_id max98357a_device_id[] = { - { .compatible = "maxim," DRV_NAME, }, + { .compatible = "maxim,max98357a" }, {} }; MODULE_DEVICE_TABLE(of, max98357a_device_id); @@ -135,7 +133,7 @@ MODULE_DEVICE_TABLE(of, max98357a_device_id); static struct platform_driver max98357a_platform_driver = { .driver = { - .name = DRV_NAME, + .name = "max98357a", .of_match_table = of_match_ptr(max98357a_device_id), }, .probe = max98357a_platform_probe, @@ -145,4 +143,3 @@ module_platform_driver(max98357a_platform_driver); MODULE_DESCRIPTION("Maxim MAX98357A Codec Driver"); MODULE_LICENSE("GPL v2"); -MODULE_ALIAS("platform:" DRV_NAME); diff --git a/sound/soc/codecs/pcm512x.c b/sound/soc/codecs/pcm512x.c index 9974f20..4b5f1fe 100644 --- a/sound/soc/codecs/pcm512x.c +++ b/sound/soc/codecs/pcm512x.c @@ -54,6 +54,9 @@ struct pcm512x_priv { int pll_d; int pll_p; unsigned long real_pll; + unsigned long overclock_pll; + unsigned long overclock_dac; + unsigned long overclock_dsp; }; /* @@ -224,6 +227,90 @@ static bool pcm512x_volatile(struct device *dev, unsigned int reg) } } +static int pcm512x_overclock_pll_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); + struct pcm512x_priv *pcm512x = snd_soc_codec_get_drvdata(codec); + + ucontrol->value.integer.value[0] = pcm512x->overclock_pll; + return 0; +} + +static int pcm512x_overclock_pll_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); + struct pcm512x_priv *pcm512x = snd_soc_codec_get_drvdata(codec); + + switch (codec->dapm.bias_level) { + case SND_SOC_BIAS_OFF: + case SND_SOC_BIAS_STANDBY: + break; + default: + return -EBUSY; + } + + pcm512x->overclock_pll = ucontrol->value.integer.value[0]; + return 0; +} + +static int pcm512x_overclock_dsp_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); + struct pcm512x_priv *pcm512x = snd_soc_codec_get_drvdata(codec); + + ucontrol->value.integer.value[0] = pcm512x->overclock_dsp; + return 0; +} + +static int pcm512x_overclock_dsp_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); + struct pcm512x_priv *pcm512x = snd_soc_codec_get_drvdata(codec); + + switch (codec->dapm.bias_level) { + case SND_SOC_BIAS_OFF: + case SND_SOC_BIAS_STANDBY: + break; + default: + return -EBUSY; + } + + pcm512x->overclock_dsp = ucontrol->value.integer.value[0]; + return 0; +} + +static int pcm512x_overclock_dac_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); + struct pcm512x_priv *pcm512x = snd_soc_codec_get_drvdata(codec); + + ucontrol->value.integer.value[0] = pcm512x->overclock_dac; + return 0; +} + +static int pcm512x_overclock_dac_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); + struct pcm512x_priv *pcm512x = snd_soc_codec_get_drvdata(codec); + + switch (codec->dapm.bias_level) { + case SND_SOC_BIAS_OFF: + case SND_SOC_BIAS_STANDBY: + break; + default: + return -EBUSY; + } + + pcm512x->overclock_dac = ucontrol->value.integer.value[0]; + return 0; +} + static const DECLARE_TLV_DB_SCALE(digital_tlv, -10350, 50, 1); static const DECLARE_TLV_DB_SCALE(analog_tlv, -600, 600, 0); static const DECLARE_TLV_DB_SCALE(boost_tlv, 0, 80, 0); @@ -328,6 +415,13 @@ SOC_ENUM("Volume Ramp Up Rate", pcm512x_vnuf), SOC_ENUM("Volume Ramp Up Step", pcm512x_vnus), SOC_ENUM("Volume Ramp Down Emergency Rate", pcm512x_vedf), SOC_ENUM("Volume Ramp Down Emergency Step", pcm512x_veds), + +SOC_SINGLE_EXT("Max Overclock PLL", SND_SOC_NOPM, 0, 20, 0, + pcm512x_overclock_pll_get, pcm512x_overclock_pll_put), +SOC_SINGLE_EXT("Max Overclock DSP", SND_SOC_NOPM, 0, 40, 0, + pcm512x_overclock_dsp_get, pcm512x_overclock_dsp_put), +SOC_SINGLE_EXT("Max Overclock DAC", SND_SOC_NOPM, 0, 40, 0, + pcm512x_overclock_dac_get, pcm512x_overclock_dac_put), }; static const struct snd_soc_dapm_widget pcm512x_dapm_widgets[] = { @@ -346,6 +440,45 @@ static const struct snd_soc_dapm_route pcm512x_dapm_routes[] = { { "OUTR", NULL, "DACR" }, }; +static unsigned long pcm512x_pll_max(struct pcm512x_priv *pcm512x) +{ + return 25000000 + 25000000 * pcm512x->overclock_pll / 100; +} + +static unsigned long pcm512x_dsp_max(struct pcm512x_priv *pcm512x) +{ + return 50000000 + 50000000 * pcm512x->overclock_dsp / 100; +} + +static unsigned long pcm512x_dac_max(struct pcm512x_priv *pcm512x, + unsigned long rate) +{ + return rate + rate * pcm512x->overclock_dac / 100; +} + +static unsigned long pcm512x_sck_max(struct pcm512x_priv *pcm512x) +{ + if (!pcm512x->pll_out) + return 25000000; + return pcm512x_pll_max(pcm512x); +} + +static unsigned long pcm512x_ncp_target(struct pcm512x_priv *pcm512x, + unsigned long dac_rate) +{ + /* + * If the DAC is not actually overclocked, use the good old + * NCP target rate... + */ + if (dac_rate <= 6144000) + return 1536000; + /* + * ...but if the DAC is in fact overclocked, bump the NCP target + * rate to get the recommended dividers even when overclocking. + */ + return pcm512x_dac_max(pcm512x, 1536000); +} + static const u32 pcm512x_dai_rates[] = { 8000, 11025, 16000, 22050, 32000, 44100, 48000, 64000, 88200, 96000, 176400, 192000, 384000, @@ -359,6 +492,7 @@ static const struct snd_pcm_hw_constraint_list constraints_slave = { static int pcm512x_hw_rule_rate(struct snd_pcm_hw_params *params, struct snd_pcm_hw_rule *rule) { + struct pcm512x_priv *pcm512x = rule->private; struct snd_interval ranges[2]; int frame_size; @@ -377,7 +511,7 @@ static int pcm512x_hw_rule_rate(struct snd_pcm_hw_params *params, */ memset(ranges, 0, sizeof(ranges)); ranges[0].min = 8000; - ranges[0].max = 25000000 / frame_size / 2; + ranges[0].max = pcm512x_sck_max(pcm512x) / frame_size / 2; ranges[1].min = DIV_ROUND_UP(16000000, frame_size); ranges[1].max = 384000; break; @@ -408,7 +542,7 @@ static int pcm512x_dai_startup_master(struct snd_pcm_substream *substream, return snd_pcm_hw_rule_add(substream->runtime, 0, SNDRV_PCM_HW_PARAM_RATE, pcm512x_hw_rule_rate, - NULL, + pcm512x, SNDRV_PCM_HW_PARAM_FRAME_BITS, SNDRV_PCM_HW_PARAM_CHANNELS, -1); @@ -517,6 +651,8 @@ static unsigned long pcm512x_find_sck(struct snd_soc_dai *dai, unsigned long bclk_rate) { struct device *dev = dai->dev; + struct snd_soc_codec *codec = dai->codec; + struct pcm512x_priv *pcm512x = snd_soc_codec_get_drvdata(codec); unsigned long sck_rate; int pow2; @@ -527,9 +663,10 @@ static unsigned long pcm512x_find_sck(struct snd_soc_dai *dai, * as many factors of 2 as possible, as that makes it easier * to find a fast DAC rate */ - pow2 = 1 << fls((25000000 - 16000000) / bclk_rate); + pow2 = 1 << fls((pcm512x_pll_max(pcm512x) - 16000000) / bclk_rate); for (; pow2; pow2 >>= 1) { - sck_rate = rounddown(25000000, bclk_rate * pow2); + sck_rate = rounddown(pcm512x_pll_max(pcm512x), + bclk_rate * pow2); if (sck_rate >= 16000000) break; } @@ -678,7 +815,7 @@ static unsigned long pcm512x_pllin_dac_rate(struct snd_soc_dai *dai, return 0; /* futile, quit early */ /* run DAC no faster than 6144000 Hz */ - for (dac_rate = rounddown(6144000, osr_rate); + for (dac_rate = rounddown(pcm512x_dac_max(pcm512x, 6144000), osr_rate); dac_rate; dac_rate -= osr_rate) { @@ -805,7 +942,7 @@ static int pcm512x_set_dividers(struct snd_soc_dai *dai, osr_rate = 16 * sample_rate; /* run DSP no faster than 50 MHz */ - dsp_div = mck_rate > 50000000 ? 2 : 1; + dsp_div = mck_rate > pcm512x_dsp_max(pcm512x) ? 2 : 1; dac_rate = pcm512x_pllin_dac_rate(dai, osr_rate, pllin_rate); if (dac_rate) { @@ -836,7 +973,8 @@ static int pcm512x_set_dividers(struct snd_soc_dai *dai, dacsrc_rate = pllin_rate; } else { /* run DAC no faster than 6144000 Hz */ - unsigned long dac_mul = 6144000 / osr_rate; + unsigned long dac_mul = pcm512x_dac_max(pcm512x, 6144000) + / osr_rate; unsigned long sck_mul = sck_rate / osr_rate; for (; dac_mul; dac_mul--) { @@ -863,28 +1001,30 @@ static int pcm512x_set_dividers(struct snd_soc_dai *dai, dacsrc_rate = sck_rate; } + osr_div = DIV_ROUND_CLOSEST(dac_rate, osr_rate); + if (osr_div > 128) { + dev_err(dev, "Failed to find OSR divider\n"); + return -EINVAL; + } + dac_div = DIV_ROUND_CLOSEST(dacsrc_rate, dac_rate); if (dac_div > 128) { dev_err(dev, "Failed to find DAC divider\n"); return -EINVAL; } + dac_rate = dacsrc_rate / dac_div; - ncp_div = DIV_ROUND_CLOSEST(dacsrc_rate / dac_div, 1536000); - if (ncp_div > 128 || dacsrc_rate / dac_div / ncp_div > 2048000) { + ncp_div = DIV_ROUND_CLOSEST(dac_rate, + pcm512x_ncp_target(pcm512x, dac_rate)); + if (ncp_div > 128 || dac_rate / ncp_div > 2048000) { /* run NCP no faster than 2048000 Hz, but why? */ - ncp_div = DIV_ROUND_UP(dacsrc_rate / dac_div, 2048000); + ncp_div = DIV_ROUND_UP(dac_rate, 2048000); if (ncp_div > 128) { dev_err(dev, "Failed to find NCP divider\n"); return -EINVAL; } } - osr_div = DIV_ROUND_CLOSEST(dac_rate, osr_rate); - if (osr_div > 128) { - dev_err(dev, "Failed to find OSR divider\n"); - return -EINVAL; - } - idac = mck_rate / (dsp_div * sample_rate); ret = regmap_write(pcm512x->regmap, PCM512x_DSP_CLKDIV, dsp_div - 1); @@ -937,11 +1077,11 @@ static int pcm512x_set_dividers(struct snd_soc_dai *dai, return ret; } - if (sample_rate <= 48000) + if (sample_rate <= pcm512x_dac_max(pcm512x, 48000)) fssp = PCM512x_FSSP_48KHZ; - else if (sample_rate <= 96000) + else if (sample_rate <= pcm512x_dac_max(pcm512x, 96000)) fssp = PCM512x_FSSP_96KHZ; - else if (sample_rate <= 192000) + else if (sample_rate <= pcm512x_dac_max(pcm512x, 192000)) fssp = PCM512x_FSSP_192KHZ; else fssp = PCM512x_FSSP_384KHZ; diff --git a/sound/soc/codecs/rt286.c b/sound/soc/codecs/rt286.c index 9b541e5..8260370 100644 --- a/sound/soc/codecs/rt286.c +++ b/sound/soc/codecs/rt286.c @@ -395,9 +395,20 @@ int rt286_mic_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack) rt286->jack = jack; - /* Send an initial empty report */ - snd_soc_jack_report(rt286->jack, 0, - SND_JACK_MICROPHONE | SND_JACK_HEADPHONE); + if (jack) { + /* enable IRQ */ + if (rt286->jack->status | SND_JACK_HEADPHONE) + snd_soc_dapm_force_enable_pin(&codec->dapm, "LDO1"); + regmap_update_bits(rt286->regmap, RT286_IRQ_CTRL, 0x2, 0x2); + /* Send an initial empty report */ + snd_soc_jack_report(rt286->jack, rt286->jack->status, + SND_JACK_MICROPHONE | SND_JACK_HEADPHONE); + } else { + /* disable IRQ */ + regmap_update_bits(rt286->regmap, RT286_IRQ_CTRL, 0x2, 0x0); + snd_soc_dapm_disable_pin(&codec->dapm, "LDO1"); + } + snd_soc_dapm_sync(&codec->dapm); return 0; } diff --git a/sound/soc/codecs/rt5670.h b/sound/soc/codecs/rt5670.h index 21f8e18..0a67adb 100644 --- a/sound/soc/codecs/rt5670.h +++ b/sound/soc/codecs/rt5670.h @@ -1950,17 +1950,20 @@ enum { }; enum { + RT5670_DMIC1_DISABLED, RT5670_DMIC_DATA_GPIO6, RT5670_DMIC_DATA_IN2P, RT5670_DMIC_DATA_GPIO7, }; enum { + RT5670_DMIC2_DISABLED, RT5670_DMIC_DATA_GPIO8, RT5670_DMIC_DATA_IN3N, }; enum { + RT5670_DMIC3_DISABLED, RT5670_DMIC_DATA_GPIO9, RT5670_DMIC_DATA_GPIO10, RT5670_DMIC_DATA_GPIO5, diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c index fb9c20e..c2a6e40 100644 --- a/sound/soc/codecs/rt5677.c +++ b/sound/soc/codecs/rt5677.c @@ -718,11 +718,24 @@ static int rt5677_set_dsp_vad(struct snd_soc_codec *codec, bool on) RT5677_LDO1_SEL_MASK, 0x0); regmap_update_bits(rt5677->regmap, RT5677_PWR_ANLG2, RT5677_PWR_LDO1, RT5677_PWR_LDO1); - regmap_update_bits(rt5677->regmap, RT5677_GLB_CLK1, - RT5677_MCLK_SRC_MASK, RT5677_MCLK2_SRC); - regmap_update_bits(rt5677->regmap, RT5677_GLB_CLK2, - RT5677_PLL2_PR_SRC_MASK | RT5677_DSP_CLK_SRC_MASK, - RT5677_PLL2_PR_SRC_MCLK2 | RT5677_DSP_CLK_SRC_BYPASS); + switch (rt5677->type) { + case RT5677: + regmap_update_bits(rt5677->regmap, RT5677_GLB_CLK1, + RT5677_MCLK_SRC_MASK, RT5677_MCLK2_SRC); + regmap_update_bits(rt5677->regmap, RT5677_GLB_CLK2, + RT5677_PLL2_PR_SRC_MASK | + RT5677_DSP_CLK_SRC_MASK, + RT5677_PLL2_PR_SRC_MCLK2 | + RT5677_DSP_CLK_SRC_BYPASS); + break; + case RT5676: + regmap_update_bits(rt5677->regmap, RT5677_GLB_CLK2, + RT5677_DSP_CLK_SRC_MASK, + RT5677_DSP_CLK_SRC_BYPASS); + break; + default: + break; + } regmap_write(rt5677->regmap, RT5677_PWR_DSP2, 0x07ff); regmap_write(rt5677->regmap, RT5677_PWR_DSP1, 0x07fd); rt5677_set_dsp_mode(codec, true); @@ -4500,10 +4513,10 @@ static int rt5677_suspend(struct snd_soc_codec *codec) if (!rt5677->dsp_vad_en) { regcache_cache_only(rt5677->regmap, true); regcache_mark_dirty(rt5677->regmap); - } - if (gpio_is_valid(rt5677->pow_ldo2)) - gpio_set_value_cansleep(rt5677->pow_ldo2, 0); + if (gpio_is_valid(rt5677->pow_ldo2)) + gpio_set_value_cansleep(rt5677->pow_ldo2, 0); + } return 0; } @@ -4512,12 +4525,12 @@ static int rt5677_resume(struct snd_soc_codec *codec) { struct rt5677_priv *rt5677 = snd_soc_codec_get_drvdata(codec); - if (gpio_is_valid(rt5677->pow_ldo2)) { - gpio_set_value_cansleep(rt5677->pow_ldo2, 1); - msleep(10); - } - if (!rt5677->dsp_vad_en) { + if (gpio_is_valid(rt5677->pow_ldo2)) { + gpio_set_value_cansleep(rt5677->pow_ldo2, 1); + msleep(10); + } + regcache_cache_only(rt5677->regmap, false); regcache_sync(rt5677->regmap); } @@ -4733,7 +4746,8 @@ static const struct regmap_config rt5677_regmap = { }; static const struct i2c_device_id rt5677_i2c_id[] = { - { "rt5677", 0 }, + { "rt5677", RT5677 }, + { "rt5676", RT5676 }, { } }; MODULE_DEVICE_TABLE(i2c, rt5677_i2c_id); @@ -4850,6 +4864,8 @@ static int rt5677_i2c_probe(struct i2c_client *i2c, i2c_set_clientdata(i2c, rt5677); + rt5677->type = id->driver_data; + if (pdata) rt5677->pdata = *pdata; diff --git a/sound/soc/codecs/rt5677.h b/sound/soc/codecs/rt5677.h index c0a625f..07df96b 100644 --- a/sound/soc/codecs/rt5677.h +++ b/sound/soc/codecs/rt5677.h @@ -1665,6 +1665,11 @@ enum { RT5677_IRQ_JD3, }; +enum rt5677_type { + RT5677, + RT5676, +}; + struct rt5677_priv { struct snd_soc_codec *codec; struct rt5677_platform_data pdata; @@ -1681,6 +1686,7 @@ struct rt5677_priv { int pll_in; int pll_out; int pow_ldo2; /* POW_LDO2 pin */ + enum rt5677_type type; #ifdef CONFIG_GPIOLIB struct gpio_chip gpio_chip; #endif diff --git a/sound/soc/codecs/sn95031.c b/sound/soc/codecs/sn95031.c index 82095d6c..7947c0e 100644 --- a/sound/soc/codecs/sn95031.c +++ b/sound/soc/codecs/sn95031.c @@ -783,19 +783,21 @@ static inline void sn95031_enable_jack_btn(struct snd_soc_codec *codec) snd_soc_write(codec, SN95031_BTNCTRL2, 0x01); } -static int sn95031_get_headset_state(struct snd_soc_jack *mfld_jack) +static int sn95031_get_headset_state(struct snd_soc_codec *codec, + struct snd_soc_jack *mfld_jack) { - int micbias = sn95031_get_mic_bias(mfld_jack->codec); + int micbias = sn95031_get_mic_bias(codec); int jack_type = snd_soc_jack_get_type(mfld_jack, micbias); pr_debug("jack type detected = %d\n", jack_type); if (jack_type == SND_JACK_HEADSET) - sn95031_enable_jack_btn(mfld_jack->codec); + sn95031_enable_jack_btn(codec); return jack_type; } -void sn95031_jack_detection(struct mfld_jack_data *jack_data) +void sn95031_jack_detection(struct snd_soc_codec *codec, + struct mfld_jack_data *jack_data) { unsigned int status; unsigned int mask = SND_JACK_BTN_0 | SND_JACK_BTN_1 | SND_JACK_HEADSET; @@ -809,11 +811,11 @@ void sn95031_jack_detection(struct mfld_jack_data *jack_data) status = SND_JACK_HEADSET | SND_JACK_BTN_1; } else if (jack_data->intr_id & 0x4) { pr_debug("headset or headphones inserted\n"); - status = sn95031_get_headset_state(jack_data->mfld_jack); + status = sn95031_get_headset_state(codec, jack_data->mfld_jack); } else if (jack_data->intr_id & 0x8) { pr_debug("headset or headphones removed\n"); status = 0; - sn95031_disable_jack_btn(jack_data->mfld_jack->codec); + sn95031_disable_jack_btn(codec); } else { pr_err("unidentified interrupt\n"); return; diff --git a/sound/soc/codecs/sn95031.h b/sound/soc/codecs/sn95031.h index 20376d2..7651fe4 100644 --- a/sound/soc/codecs/sn95031.h +++ b/sound/soc/codecs/sn95031.h @@ -127,6 +127,7 @@ struct mfld_jack_data { struct snd_soc_jack *mfld_jack; }; -extern void sn95031_jack_detection(struct mfld_jack_data *jack_data); +extern void sn95031_jack_detection(struct snd_soc_codec *codec, + struct mfld_jack_data *jack_data); #endif diff --git a/sound/soc/codecs/sta350.c b/sound/soc/codecs/sta350.c index bda2ee1..669e322 100644 --- a/sound/soc/codecs/sta350.c +++ b/sound/soc/codecs/sta350.c @@ -1213,27 +1213,15 @@ static int sta350_i2c_probe(struct i2c_client *i2c, #endif /* GPIOs */ - sta350->gpiod_nreset = devm_gpiod_get(dev, "reset"); - if (IS_ERR(sta350->gpiod_nreset)) { - ret = PTR_ERR(sta350->gpiod_nreset); - if (ret != -ENOENT && ret != -ENOSYS) - return ret; - - sta350->gpiod_nreset = NULL; - } else { - gpiod_direction_output(sta350->gpiod_nreset, 0); - } - - sta350->gpiod_power_down = devm_gpiod_get(dev, "power-down"); - if (IS_ERR(sta350->gpiod_power_down)) { - ret = PTR_ERR(sta350->gpiod_power_down); - if (ret != -ENOENT && ret != -ENOSYS) - return ret; - - sta350->gpiod_power_down = NULL; - } else { - gpiod_direction_output(sta350->gpiod_power_down, 0); - } + sta350->gpiod_nreset = devm_gpiod_get_optional(dev, "reset", + GPIOD_OUT_LOW); + if (IS_ERR(sta350->gpiod_nreset)) + return PTR_ERR(sta350->gpiod_nreset); + + sta350->gpiod_power_down = devm_gpiod_get(dev, "power-down", + GPIOD_OUT_LOW); + if (IS_ERR(sta350->gpiod_power_down)) + return PTR_ERR(sta350->gpiod_power_down); /* regulators */ for (i = 0; i < ARRAY_SIZE(sta350->supplies); i++) diff --git a/sound/soc/codecs/tas2552.c b/sound/soc/codecs/tas2552.c index ae23acd..dfb4ff5 100644 --- a/sound/soc/codecs/tas2552.c +++ b/sound/soc/codecs/tas2552.c @@ -485,16 +485,9 @@ static int tas2552_probe(struct i2c_client *client, if (data == NULL) return -ENOMEM; - data->enable_gpio = devm_gpiod_get(dev, "enable"); - if (IS_ERR(data->enable_gpio)) { - ret = PTR_ERR(data->enable_gpio); - if (ret != -ENOENT && ret != -ENOSYS) - return ret; - - data->enable_gpio = NULL; - } else { - gpiod_direction_output(data->enable_gpio, 0); - } + data->enable_gpio = devm_gpiod_get(dev, "enable", GPIOD_OUT_LOW); + if (IS_ERR(data->enable_gpio)) + return PTR_ERR(data->enable_gpio); data->tas2552_client = client; data->regmap = devm_regmap_init_i2c(client, &tas2552_regmap_config); diff --git a/sound/soc/codecs/wm8804-i2c.c b/sound/soc/codecs/wm8804-i2c.c new file mode 100644 index 0000000..5bd4af2 --- /dev/null +++ b/sound/soc/codecs/wm8804-i2c.c @@ -0,0 +1,64 @@ +/* + * wm8804-i2c.c -- WM8804 S/PDIF transceiver driver - I2C + * + * Copyright 2015 Cirrus Logic Inc + * + * Author: Charles Keepax <ckeepax@opensource.wolfsonmicro.com> + * + * 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. + */ + +#include <linux/init.h> +#include <linux/module.h> +#include <linux/i2c.h> + +#include "wm8804.h" + +static int wm8804_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct regmap *regmap; + + regmap = devm_regmap_init_i2c(i2c, &wm8804_regmap_config); + if (IS_ERR(regmap)) + return PTR_ERR(regmap); + + return wm8804_probe(&i2c->dev, regmap); +} + +static int wm8804_i2c_remove(struct i2c_client *i2c) +{ + wm8804_remove(&i2c->dev); + return 0; +} + +static const struct i2c_device_id wm8804_i2c_id[] = { + { "wm8804", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, wm8804_i2c_id); + +static const struct of_device_id wm8804_of_match[] = { + { .compatible = "wlf,wm8804", }, + { } +}; +MODULE_DEVICE_TABLE(of, wm8804_of_match); + +static struct i2c_driver wm8804_i2c_driver = { + .driver = { + .name = "wm8804", + .owner = THIS_MODULE, + .of_match_table = wm8804_of_match, + }, + .probe = wm8804_i2c_probe, + .remove = wm8804_i2c_remove, + .id_table = wm8804_i2c_id +}; + +module_i2c_driver(wm8804_i2c_driver); + +MODULE_DESCRIPTION("ASoC WM8804 driver - I2C"); +MODULE_AUTHOR("Charles Keepax <ckeepax@opensource.wolfsonmicro.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/wm8804-spi.c b/sound/soc/codecs/wm8804-spi.c new file mode 100644 index 0000000..287e11e --- /dev/null +++ b/sound/soc/codecs/wm8804-spi.c @@ -0,0 +1,56 @@ +/* + * wm8804-spi.c -- WM8804 S/PDIF transceiver driver - SPI + * + * Copyright 2015 Cirrus Logic Inc + * + * Author: Charles Keepax <ckeepax@opensource.wolfsonmicro.com> + * + * 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. + */ + +#include <linux/init.h> +#include <linux/module.h> +#include <linux/spi/spi.h> + +#include "wm8804.h" + +static int wm8804_spi_probe(struct spi_device *spi) +{ + struct regmap *regmap; + + regmap = devm_regmap_init_spi(spi, &wm8804_regmap_config); + if (IS_ERR(regmap)) + return PTR_ERR(regmap); + + return wm8804_probe(&spi->dev, regmap); +} + +static int wm8804_spi_remove(struct spi_device *spi) +{ + wm8804_remove(&spi->dev); + return 0; +} + +static const struct of_device_id wm8804_of_match[] = { + { .compatible = "wlf,wm8804", }, + { } +}; +MODULE_DEVICE_TABLE(of, wm8804_of_match); + +static struct spi_driver wm8804_spi_driver = { + .driver = { + .name = "wm8804", + .owner = THIS_MODULE, + .of_match_table = wm8804_of_match, + }, + .probe = wm8804_spi_probe, + .remove = wm8804_spi_remove +}; + +module_spi_driver(wm8804_spi_driver); + +MODULE_DESCRIPTION("ASoC WM8804 driver - SPI"); +MODULE_AUTHOR("Charles Keepax <ckeepax@opensource.wolfsonmicro.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/wm8804.c b/sound/soc/codecs/wm8804.c index b2b0e68..1bd4ace 100644 --- a/sound/soc/codecs/wm8804.c +++ b/sound/soc/codecs/wm8804.c @@ -15,10 +15,7 @@ #include <linux/init.h> #include <linux/delay.h> #include <linux/pm.h> -#include <linux/i2c.h> #include <linux/of_device.h> -#include <linux/spi/spi.h> -#include <linux/regmap.h> #include <linux/regulator/consumer.h> #include <linux/slab.h> #include <sound/core.h> @@ -185,9 +182,9 @@ static bool wm8804_volatile(struct device *dev, unsigned int reg) } } -static int wm8804_reset(struct snd_soc_codec *codec) +static int wm8804_reset(struct wm8804_priv *wm8804) { - return snd_soc_write(codec, WM8804_RST_DEVID1, 0x0); + return regmap_write(wm8804->regmap, WM8804_RST_DEVID1, 0x0); } static int wm8804_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) @@ -518,100 +515,6 @@ static int wm8804_set_bias_level(struct snd_soc_codec *codec, return 0; } -static int wm8804_remove(struct snd_soc_codec *codec) -{ - struct wm8804_priv *wm8804; - int i; - - wm8804 = snd_soc_codec_get_drvdata(codec); - - for (i = 0; i < ARRAY_SIZE(wm8804->supplies); ++i) - regulator_unregister_notifier(wm8804->supplies[i].consumer, - &wm8804->disable_nb[i]); - return 0; -} - -static int wm8804_probe(struct snd_soc_codec *codec) -{ - struct wm8804_priv *wm8804; - int i, id1, id2, ret; - - wm8804 = snd_soc_codec_get_drvdata(codec); - - for (i = 0; i < ARRAY_SIZE(wm8804->supplies); i++) - wm8804->supplies[i].supply = wm8804_supply_names[i]; - - ret = devm_regulator_bulk_get(codec->dev, ARRAY_SIZE(wm8804->supplies), - wm8804->supplies); - if (ret) { - dev_err(codec->dev, "Failed to request supplies: %d\n", ret); - return ret; - } - - wm8804->disable_nb[0].notifier_call = wm8804_regulator_event_0; - wm8804->disable_nb[1].notifier_call = wm8804_regulator_event_1; - - /* This should really be moved into the regulator core */ - for (i = 0; i < ARRAY_SIZE(wm8804->supplies); i++) { - ret = regulator_register_notifier(wm8804->supplies[i].consumer, - &wm8804->disable_nb[i]); - if (ret != 0) { - dev_err(codec->dev, - "Failed to register regulator notifier: %d\n", - ret); - } - } - - ret = regulator_bulk_enable(ARRAY_SIZE(wm8804->supplies), - wm8804->supplies); - if (ret) { - dev_err(codec->dev, "Failed to enable supplies: %d\n", ret); - return ret; - } - - id1 = snd_soc_read(codec, WM8804_RST_DEVID1); - if (id1 < 0) { - dev_err(codec->dev, "Failed to read device ID: %d\n", id1); - ret = id1; - goto err_reg_enable; - } - - id2 = snd_soc_read(codec, WM8804_DEVID2); - if (id2 < 0) { - dev_err(codec->dev, "Failed to read device ID: %d\n", id2); - ret = id2; - goto err_reg_enable; - } - - id2 = (id2 << 8) | id1; - - if (id2 != 0x8805) { - dev_err(codec->dev, "Invalid device ID: %#x\n", id2); - ret = -EINVAL; - goto err_reg_enable; - } - - ret = snd_soc_read(codec, WM8804_DEVREV); - if (ret < 0) { - dev_err(codec->dev, "Failed to read device revision: %d\n", - ret); - goto err_reg_enable; - } - dev_info(codec->dev, "revision %c\n", ret + 'A'); - - ret = wm8804_reset(codec); - if (ret < 0) { - dev_err(codec->dev, "Failed to issue reset: %d\n", ret); - goto err_reg_enable; - } - - return 0; - -err_reg_enable: - regulator_bulk_disable(ARRAY_SIZE(wm8804->supplies), wm8804->supplies); - return ret; -} - static const struct snd_soc_dai_ops wm8804_dai_ops = { .hw_params = wm8804_hw_params, .set_fmt = wm8804_set_fmt, @@ -649,8 +552,6 @@ static struct snd_soc_dai_driver wm8804_dai = { }; static const struct snd_soc_codec_driver soc_codec_dev_wm8804 = { - .probe = wm8804_probe, - .remove = wm8804_remove, .set_bias_level = wm8804_set_bias_level, .idle_bias_off = true, @@ -658,13 +559,7 @@ static const struct snd_soc_codec_driver soc_codec_dev_wm8804 = { .num_controls = ARRAY_SIZE(wm8804_snd_controls), }; -static const struct of_device_id wm8804_of_match[] = { - { .compatible = "wlf,wm8804", }, - { } -}; -MODULE_DEVICE_TABLE(of, wm8804_of_match); - -static const struct regmap_config wm8804_regmap_config = { +const struct regmap_config wm8804_regmap_config = { .reg_bits = 8, .val_bits = 8, @@ -675,128 +570,110 @@ static const struct regmap_config wm8804_regmap_config = { .reg_defaults = wm8804_reg_defaults, .num_reg_defaults = ARRAY_SIZE(wm8804_reg_defaults), }; +EXPORT_SYMBOL_GPL(wm8804_regmap_config); -#if defined(CONFIG_SPI_MASTER) -static int wm8804_spi_probe(struct spi_device *spi) +int wm8804_probe(struct device *dev, struct regmap *regmap) { struct wm8804_priv *wm8804; - int ret; + unsigned int id1, id2; + int i, ret; - wm8804 = devm_kzalloc(&spi->dev, sizeof *wm8804, GFP_KERNEL); + wm8804 = devm_kzalloc(dev, sizeof(*wm8804), GFP_KERNEL); if (!wm8804) return -ENOMEM; - wm8804->regmap = devm_regmap_init_spi(spi, &wm8804_regmap_config); - if (IS_ERR(wm8804->regmap)) { - ret = PTR_ERR(wm8804->regmap); + dev_set_drvdata(dev, wm8804); + + wm8804->regmap = regmap; + + for (i = 0; i < ARRAY_SIZE(wm8804->supplies); i++) + wm8804->supplies[i].supply = wm8804_supply_names[i]; + + ret = devm_regulator_bulk_get(dev, ARRAY_SIZE(wm8804->supplies), + wm8804->supplies); + if (ret) { + dev_err(dev, "Failed to request supplies: %d\n", ret); return ret; } - spi_set_drvdata(spi, wm8804); + wm8804->disable_nb[0].notifier_call = wm8804_regulator_event_0; + wm8804->disable_nb[1].notifier_call = wm8804_regulator_event_1; - ret = snd_soc_register_codec(&spi->dev, - &soc_codec_dev_wm8804, &wm8804_dai, 1); + /* This should really be moved into the regulator core */ + for (i = 0; i < ARRAY_SIZE(wm8804->supplies); i++) { + ret = regulator_register_notifier(wm8804->supplies[i].consumer, + &wm8804->disable_nb[i]); + if (ret != 0) { + dev_err(dev, + "Failed to register regulator notifier: %d\n", + ret); + } + } - return ret; -} + ret = regulator_bulk_enable(ARRAY_SIZE(wm8804->supplies), + wm8804->supplies); + if (ret) { + dev_err(dev, "Failed to enable supplies: %d\n", ret); + goto err_reg_enable; + } -static int wm8804_spi_remove(struct spi_device *spi) -{ - snd_soc_unregister_codec(&spi->dev); - return 0; -} + ret = regmap_read(regmap, WM8804_RST_DEVID1, &id1); + if (ret < 0) { + dev_err(dev, "Failed to read device ID: %d\n", ret); + goto err_reg_enable; + } -static struct spi_driver wm8804_spi_driver = { - .driver = { - .name = "wm8804", - .owner = THIS_MODULE, - .of_match_table = wm8804_of_match, - }, - .probe = wm8804_spi_probe, - .remove = wm8804_spi_remove -}; -#endif + ret = regmap_read(regmap, WM8804_DEVID2, &id2); + if (ret < 0) { + dev_err(dev, "Failed to read device ID: %d\n", ret); + goto err_reg_enable; + } -#if IS_ENABLED(CONFIG_I2C) -static int wm8804_i2c_probe(struct i2c_client *i2c, - const struct i2c_device_id *id) -{ - struct wm8804_priv *wm8804; - int ret; + id2 = (id2 << 8) | id1; - wm8804 = devm_kzalloc(&i2c->dev, sizeof *wm8804, GFP_KERNEL); - if (!wm8804) - return -ENOMEM; + if (id2 != 0x8805) { + dev_err(dev, "Invalid device ID: %#x\n", id2); + ret = -EINVAL; + goto err_reg_enable; + } - wm8804->regmap = devm_regmap_init_i2c(i2c, &wm8804_regmap_config); - if (IS_ERR(wm8804->regmap)) { - ret = PTR_ERR(wm8804->regmap); - return ret; + ret = regmap_read(regmap, WM8804_DEVREV, &id1); + if (ret < 0) { + dev_err(dev, "Failed to read device revision: %d\n", + ret); + goto err_reg_enable; } + dev_info(dev, "revision %c\n", id1 + 'A'); - i2c_set_clientdata(i2c, wm8804); + ret = wm8804_reset(wm8804); + if (ret < 0) { + dev_err(dev, "Failed to issue reset: %d\n", ret); + goto err_reg_enable; + } - ret = snd_soc_register_codec(&i2c->dev, - &soc_codec_dev_wm8804, &wm8804_dai, 1); + return snd_soc_register_codec(dev, &soc_codec_dev_wm8804, + &wm8804_dai, 1); + +err_reg_enable: + regulator_bulk_disable(ARRAY_SIZE(wm8804->supplies), wm8804->supplies); return ret; } +EXPORT_SYMBOL_GPL(wm8804_probe); -static int wm8804_i2c_remove(struct i2c_client *i2c) +void wm8804_remove(struct device *dev) { - snd_soc_unregister_codec(&i2c->dev); - return 0; -} - -static const struct i2c_device_id wm8804_i2c_id[] = { - { "wm8804", 0 }, - { } -}; -MODULE_DEVICE_TABLE(i2c, wm8804_i2c_id); - -static struct i2c_driver wm8804_i2c_driver = { - .driver = { - .name = "wm8804", - .owner = THIS_MODULE, - .of_match_table = wm8804_of_match, - }, - .probe = wm8804_i2c_probe, - .remove = wm8804_i2c_remove, - .id_table = wm8804_i2c_id -}; -#endif + struct wm8804_priv *wm8804; + int i; -static int __init wm8804_modinit(void) -{ - int ret = 0; + wm8804 = dev_get_drvdata(dev); -#if IS_ENABLED(CONFIG_I2C) - ret = i2c_add_driver(&wm8804_i2c_driver); - if (ret) { - printk(KERN_ERR "Failed to register wm8804 I2C driver: %d\n", - ret); - } -#endif -#if defined(CONFIG_SPI_MASTER) - ret = spi_register_driver(&wm8804_spi_driver); - if (ret != 0) { - printk(KERN_ERR "Failed to register wm8804 SPI driver: %d\n", - ret); - } -#endif - return ret; -} -module_init(wm8804_modinit); + for (i = 0; i < ARRAY_SIZE(wm8804->supplies); ++i) + regulator_unregister_notifier(wm8804->supplies[i].consumer, + &wm8804->disable_nb[i]); -static void __exit wm8804_exit(void) -{ -#if IS_ENABLED(CONFIG_I2C) - i2c_del_driver(&wm8804_i2c_driver); -#endif -#if defined(CONFIG_SPI_MASTER) - spi_unregister_driver(&wm8804_spi_driver); -#endif + snd_soc_unregister_codec(dev); } -module_exit(wm8804_exit); +EXPORT_SYMBOL_GPL(wm8804_remove); MODULE_DESCRIPTION("ASoC WM8804 driver"); MODULE_AUTHOR("Dimitris Papastamos <dp@opensource.wolfsonmicro.com>"); diff --git a/sound/soc/codecs/wm8804.h b/sound/soc/codecs/wm8804.h index e72d4f4..a39a256 100644 --- a/sound/soc/codecs/wm8804.h +++ b/sound/soc/codecs/wm8804.h @@ -13,6 +13,8 @@ #ifndef _WM8804_H #define _WM8804_H +#include <linux/regmap.h> + /* * Register values. */ @@ -62,4 +64,9 @@ #define WM8804_MCLKDIV_256FS 0 #define WM8804_MCLKDIV_128FS 1 +extern const struct regmap_config wm8804_regmap_config; + +int wm8804_probe(struct device *dev, struct regmap *regmap); +void wm8804_remove(struct device *dev); + #endif /* _WM8804_H */ diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index ff67b33..d01c209 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -420,10 +420,9 @@ static int wm_coeff_put(struct snd_kcontrol *kcontrol, memcpy(ctl->cache, p, ctl->len); - if (!ctl->enabled) { - ctl->set = 1; + ctl->set = 1; + if (!ctl->enabled) return 0; - } return wm_coeff_write_control(kcontrol, p, ctl->len); } @@ -1185,7 +1184,6 @@ static int wm_adsp_load_coeff(struct wm_adsp *dsp) int ret, pos, blocks, type, offset, reg; char *file; struct wm_adsp_buf *buf; - int tmp; file = kzalloc(PAGE_SIZE, GFP_KERNEL); if (file == NULL) @@ -1335,12 +1333,7 @@ static int wm_adsp_load_coeff(struct wm_adsp *dsp) } } - tmp = le32_to_cpu(blk->len) % 4; - if (tmp) - pos += le32_to_cpu(blk->len) + (4 - tmp) + sizeof(*blk); - else - pos += le32_to_cpu(blk->len) + sizeof(*blk); - + pos += (le32_to_cpu(blk->len) + sizeof(*blk) + 3) & ~0x03; blocks++; } diff --git a/sound/soc/davinci/Kconfig b/sound/soc/davinci/Kconfig index 2b81ca4..3736d9a 100644 --- a/sound/soc/davinci/Kconfig +++ b/sound/soc/davinci/Kconfig @@ -1,14 +1,16 @@ config SND_DAVINCI_SOC - tristate "SoC Audio for TI DAVINCI" + tristate depends on ARCH_DAVINCI + select SND_EDMA_SOC config SND_EDMA_SOC - tristate "SoC Audio for Texas Instruments chips using eDMA (AM33XX/43XX)" - depends on SOC_AM33XX || SOC_AM43XX + tristate "SoC Audio for Texas Instruments chips using eDMA" + depends on SOC_AM33XX || SOC_AM43XX || ARCH_DAVINCI select SND_SOC_GENERIC_DMAENGINE_PCM help Say Y or M here if you want audio support for TI SoC which uses eDMA. The following line of SoCs are supported by this platform driver: + - daVinci devices - AM335x - AM437x/AM438x @@ -17,7 +19,7 @@ config SND_DAVINCI_SOC_I2S config SND_DAVINCI_SOC_MCASP tristate "Multichannel Audio Serial Port (McASP) support" - depends on SND_DAVINCI_SOC || SND_OMAP_SOC || SND_EDMA_SOC + depends on SND_OMAP_SOC || SND_EDMA_SOC help Say Y or M here if you want to have support for McASP IP found in various Texas Instruments SoCs like: @@ -45,7 +47,7 @@ config SND_AM33XX_SOC_EVM config SND_DAVINCI_SOC_EVM tristate "SoC Audio support for DaVinci DM6446, DM355 or DM365 EVM" - depends on SND_DAVINCI_SOC && I2C + depends on SND_EDMA_SOC && I2C depends on MACH_DAVINCI_EVM || MACH_DAVINCI_DM355_EVM || MACH_DAVINCI_DM365_EVM select SND_DAVINCI_SOC_GENERIC_EVM help @@ -73,7 +75,7 @@ endchoice config SND_DM6467_SOC_EVM tristate "SoC Audio support for DaVinci DM6467 EVM" - depends on SND_DAVINCI_SOC && MACH_DAVINCI_DM6467_EVM && I2C + depends on SND_EDMA_SOC && MACH_DAVINCI_DM6467_EVM && I2C select SND_DAVINCI_SOC_GENERIC_EVM select SND_SOC_SPDIF @@ -82,7 +84,7 @@ config SND_DM6467_SOC_EVM config SND_DA830_SOC_EVM tristate "SoC Audio support for DA830/OMAP-L137 EVM" - depends on SND_DAVINCI_SOC && MACH_DAVINCI_DA830_EVM && I2C + depends on SND_EDMA_SOC && MACH_DAVINCI_DA830_EVM && I2C select SND_DAVINCI_SOC_GENERIC_EVM help @@ -91,7 +93,7 @@ config SND_DA830_SOC_EVM config SND_DA850_SOC_EVM tristate "SoC Audio support for DA850/OMAP-L138 EVM" - depends on SND_DAVINCI_SOC && MACH_DAVINCI_DA850_EVM && I2C + depends on SND_EDMA_SOC && MACH_DAVINCI_DA850_EVM && I2C select SND_DAVINCI_SOC_GENERIC_EVM help Say Y if you want to add support for SoC audio on TI diff --git a/sound/soc/davinci/Makefile b/sound/soc/davinci/Makefile index 09bf2ba..f883933 100644 --- a/sound/soc/davinci/Makefile +++ b/sound/soc/davinci/Makefile @@ -1,11 +1,9 @@ # DAVINCI Platform Support -snd-soc-davinci-objs := davinci-pcm.o snd-soc-edma-objs := edma-pcm.o snd-soc-davinci-i2s-objs := davinci-i2s.o snd-soc-davinci-mcasp-objs:= davinci-mcasp.o snd-soc-davinci-vcif-objs:= davinci-vcif.o -obj-$(CONFIG_SND_DAVINCI_SOC) += snd-soc-davinci.o obj-$(CONFIG_SND_EDMA_SOC) += snd-soc-edma.o obj-$(CONFIG_SND_DAVINCI_SOC_I2S) += snd-soc-davinci-i2s.o obj-$(CONFIG_SND_DAVINCI_SOC_MCASP) += snd-soc-davinci-mcasp.o diff --git a/sound/soc/davinci/davinci-i2s.c b/sound/soc/davinci/davinci-i2s.c index 15fb28f..56cb4d9 100644 --- a/sound/soc/davinci/davinci-i2s.c +++ b/sound/soc/davinci/davinci-i2s.c @@ -23,8 +23,9 @@ #include <sound/pcm_params.h> #include <sound/initval.h> #include <sound/soc.h> +#include <sound/dmaengine_pcm.h> -#include "davinci-pcm.h" +#include "edma-pcm.h" #include "davinci-i2s.h" @@ -122,7 +123,8 @@ static const unsigned char double_fmt[SNDRV_PCM_FORMAT_S32_LE + 1] = { struct davinci_mcbsp_dev { struct device *dev; - struct davinci_pcm_dma_params dma_params[2]; + struct snd_dmaengine_dai_dma_data dma_data[2]; + int dma_request[2]; void __iomem *base; #define MOD_DSP_A 0 #define MOD_DSP_B 1 @@ -419,8 +421,6 @@ static int davinci_i2s_hw_params(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { struct davinci_mcbsp_dev *dev = snd_soc_dai_get_drvdata(dai); - struct davinci_pcm_dma_params *dma_params = - &dev->dma_params[substream->stream]; struct snd_interval *i = NULL; int mcbsp_word_length, master; unsigned int rcr, xcr, srgr, clk_div, freq, framesize; @@ -532,8 +532,6 @@ static int davinci_i2s_hw_params(struct snd_pcm_substream *substream, return -EINVAL; } } - dma_params->acnt = dma_params->data_type = data_type[fmt]; - dma_params->fifo_level = 0; mcbsp_word_length = asp_word_length[fmt]; switch (master) { @@ -600,15 +598,6 @@ static int davinci_i2s_trigger(struct snd_pcm_substream *substream, int cmd, return ret; } -static int davinci_i2s_startup(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai) -{ - struct davinci_mcbsp_dev *dev = snd_soc_dai_get_drvdata(dai); - - snd_soc_dai_set_dma_data(dai, substream, dev->dma_params); - return 0; -} - static void davinci_i2s_shutdown(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { @@ -620,7 +609,6 @@ static void davinci_i2s_shutdown(struct snd_pcm_substream *substream, #define DAVINCI_I2S_RATES SNDRV_PCM_RATE_8000_96000 static const struct snd_soc_dai_ops davinci_i2s_dai_ops = { - .startup = davinci_i2s_startup, .shutdown = davinci_i2s_shutdown, .prepare = davinci_i2s_prepare, .trigger = davinci_i2s_trigger, @@ -630,7 +618,18 @@ static const struct snd_soc_dai_ops davinci_i2s_dai_ops = { }; +static int davinci_i2s_dai_probe(struct snd_soc_dai *dai) +{ + struct davinci_mcbsp_dev *dev = snd_soc_dai_get_drvdata(dai); + + dai->playback_dma_data = &dev->dma_data[SNDRV_PCM_STREAM_PLAYBACK]; + dai->capture_dma_data = &dev->dma_data[SNDRV_PCM_STREAM_CAPTURE]; + + return 0; +} + static struct snd_soc_dai_driver davinci_i2s_dai = { + .probe = davinci_i2s_dai_probe, .playback = { .channels_min = 2, .channels_max = 2, @@ -651,11 +650,9 @@ static const struct snd_soc_component_driver davinci_i2s_component = { static int davinci_i2s_probe(struct platform_device *pdev) { - struct snd_platform_data *pdata = pdev->dev.platform_data; struct davinci_mcbsp_dev *dev; struct resource *mem, *ioarea, *res; - enum dma_event_q asp_chan_q = EVENTQ_0; - enum dma_event_q ram_chan_q = EVENTQ_1; + int *dma; int ret; mem = platform_get_resource(pdev, IORESOURCE_MEM, 0); @@ -676,22 +673,6 @@ static int davinci_i2s_probe(struct platform_device *pdev) GFP_KERNEL); if (!dev) return -ENOMEM; - if (pdata) { - dev->enable_channel_combine = pdata->enable_channel_combine; - dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK].sram_size = - pdata->sram_size_playback; - dev->dma_params[SNDRV_PCM_STREAM_CAPTURE].sram_size = - pdata->sram_size_capture; - dev->clk_input_pin = pdata->clk_input_pin; - dev->i2s_accurate_sck = pdata->i2s_accurate_sck; - asp_chan_q = pdata->asp_chan_q; - ram_chan_q = pdata->ram_chan_q; - } - - dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK].asp_chan_q = asp_chan_q; - dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK].ram_chan_q = ram_chan_q; - dev->dma_params[SNDRV_PCM_STREAM_CAPTURE].asp_chan_q = asp_chan_q; - dev->dma_params[SNDRV_PCM_STREAM_CAPTURE].ram_chan_q = ram_chan_q; dev->clk = clk_get(&pdev->dev, NULL); if (IS_ERR(dev->clk)) @@ -705,10 +686,10 @@ static int davinci_i2s_probe(struct platform_device *pdev) goto err_release_clk; } - dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK].dma_addr = + dev->dma_data[SNDRV_PCM_STREAM_PLAYBACK].addr = (dma_addr_t)(mem->start + DAVINCI_MCBSP_DXR_REG); - dev->dma_params[SNDRV_PCM_STREAM_CAPTURE].dma_addr = + dev->dma_data[SNDRV_PCM_STREAM_CAPTURE].addr = (dma_addr_t)(mem->start + DAVINCI_MCBSP_DRR_REG); /* first TX, then RX */ @@ -718,7 +699,9 @@ static int davinci_i2s_probe(struct platform_device *pdev) ret = -ENXIO; goto err_release_clk; } - dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK].channel = res->start; + dma = &dev->dma_request[SNDRV_PCM_STREAM_PLAYBACK]; + *dma = res->start; + dev->dma_data[SNDRV_PCM_STREAM_PLAYBACK].filter_data = dma; res = platform_get_resource(pdev, IORESOURCE_DMA, 1); if (!res) { @@ -726,9 +709,11 @@ static int davinci_i2s_probe(struct platform_device *pdev) ret = -ENXIO; goto err_release_clk; } - dev->dma_params[SNDRV_PCM_STREAM_CAPTURE].channel = res->start; - dev->dev = &pdev->dev; + dma = &dev->dma_request[SNDRV_PCM_STREAM_CAPTURE]; + *dma = res->start; + dev->dma_data[SNDRV_PCM_STREAM_CAPTURE].filter_data = dma; + dev->dev = &pdev->dev; dev_set_drvdata(&pdev->dev, dev); ret = snd_soc_register_component(&pdev->dev, &davinci_i2s_component, @@ -736,7 +721,7 @@ static int davinci_i2s_probe(struct platform_device *pdev) if (ret != 0) goto err_release_clk; - ret = davinci_soc_platform_register(&pdev->dev); + ret = edma_pcm_platform_register(&pdev->dev); if (ret) { dev_err(&pdev->dev, "register PCM failed: %d\n", ret); goto err_unregister_component; diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index de3b155..0c88299 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -26,6 +26,7 @@ #include <linux/of.h> #include <linux/of_platform.h> #include <linux/of_device.h> +#include <linux/platform_data/davinci_asp.h> #include <sound/asoundef.h> #include <sound/core.h> @@ -36,7 +37,6 @@ #include <sound/dmaengine_pcm.h> #include <sound/omap-pcm.h> -#include "davinci-pcm.h" #include "edma-pcm.h" #include "davinci-mcasp.h" @@ -65,7 +65,6 @@ struct davinci_mcasp_context { }; struct davinci_mcasp { - struct davinci_pcm_dma_params dma_params[2]; struct snd_dmaengine_dai_dma_data dma_data[2]; void __iomem *base; u32 fifo_base; @@ -82,6 +81,7 @@ struct davinci_mcasp { u16 bclk_lrclk_ratio; int streams; u32 irq_request[2]; + int dma_request[2]; int sysclk_freq; bool bclk_master; @@ -441,6 +441,18 @@ static int davinci_mcasp_set_dai_fmt(struct snd_soc_dai *cpu_dai, mcasp_set_bits(mcasp, DAVINCI_MCASP_PDIR_REG, AFSX | AFSR); mcasp->bclk_master = 1; break; + case SND_SOC_DAIFMT_CBS_CFM: + /* codec is clock slave and frame master */ + mcasp_set_bits(mcasp, DAVINCI_MCASP_ACLKXCTL_REG, ACLKXE); + mcasp_clr_bits(mcasp, DAVINCI_MCASP_TXFMCTL_REG, AFSXE); + + mcasp_set_bits(mcasp, DAVINCI_MCASP_ACLKRCTL_REG, ACLKRE); + mcasp_clr_bits(mcasp, DAVINCI_MCASP_RXFMCTL_REG, AFSRE); + + mcasp_set_bits(mcasp, DAVINCI_MCASP_PDIR_REG, ACLKX | ACLKR); + mcasp_clr_bits(mcasp, DAVINCI_MCASP_PDIR_REG, AFSX | AFSR); + mcasp->bclk_master = 1; + break; case SND_SOC_DAIFMT_CBM_CFS: /* codec is clock master and frame slave */ mcasp_clr_bits(mcasp, DAVINCI_MCASP_ACLKXCTL_REG, ACLKXE); @@ -631,7 +643,6 @@ static int davinci_config_channel_size(struct davinci_mcasp *mcasp, static int mcasp_common_hw_param(struct davinci_mcasp *mcasp, int stream, int period_words, int channels) { - struct davinci_pcm_dma_params *dma_params = &mcasp->dma_params[stream]; struct snd_dmaengine_dai_dma_data *dma_data = &mcasp->dma_data[stream]; int i; u8 tx_ser = 0; @@ -699,10 +710,8 @@ static int mcasp_common_hw_param(struct davinci_mcasp *mcasp, int stream, * For example if three serializers are enabled the DMA * need to transfer three words per DMA request. */ - dma_params->fifo_level = active_serializers; dma_data->maxburst = active_serializers; } else { - dma_params->fifo_level = 0; dma_data->maxburst = 0; } return 0; @@ -734,7 +743,6 @@ static int mcasp_common_hw_param(struct davinci_mcasp *mcasp, int stream, /* Configure the burst size for platform drivers */ if (numevt == 1) numevt = 0; - dma_params->fifo_level = numevt; dma_data->maxburst = numevt; return 0; @@ -860,8 +868,6 @@ static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream, struct snd_soc_dai *cpu_dai) { struct davinci_mcasp *mcasp = snd_soc_dai_get_drvdata(cpu_dai); - struct davinci_pcm_dma_params *dma_params = - &mcasp->dma_params[substream->stream]; int word_length; int channels = params_channels(params); int period_size = params_period_size(params); @@ -902,31 +908,26 @@ static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream, switch (params_format(params)) { case SNDRV_PCM_FORMAT_U8: case SNDRV_PCM_FORMAT_S8: - dma_params->data_type = 1; word_length = 8; break; case SNDRV_PCM_FORMAT_U16_LE: case SNDRV_PCM_FORMAT_S16_LE: - dma_params->data_type = 2; word_length = 16; break; case SNDRV_PCM_FORMAT_U24_3LE: case SNDRV_PCM_FORMAT_S24_3LE: - dma_params->data_type = 3; word_length = 24; break; case SNDRV_PCM_FORMAT_U24_LE: case SNDRV_PCM_FORMAT_S24_LE: - dma_params->data_type = 4; word_length = 24; break; case SNDRV_PCM_FORMAT_U32_LE: case SNDRV_PCM_FORMAT_S32_LE: - dma_params->data_type = 4; word_length = 32; break; @@ -935,11 +936,6 @@ static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream, return -EINVAL; } - if (mcasp->version == MCASP_VERSION_2 && !dma_params->fifo_level) - dma_params->acnt = 4; - else - dma_params->acnt = dma_params->data_type; - davinci_config_channel_size(mcasp, word_length); if (mcasp->op_mode == DAVINCI_MCASP_IIS_MODE) @@ -1043,17 +1039,8 @@ static int davinci_mcasp_dai_probe(struct snd_soc_dai *dai) { struct davinci_mcasp *mcasp = snd_soc_dai_get_drvdata(dai); - if (mcasp->version >= MCASP_VERSION_3) { - /* Using dmaengine PCM */ - dai->playback_dma_data = - &mcasp->dma_data[SNDRV_PCM_STREAM_PLAYBACK]; - dai->capture_dma_data = - &mcasp->dma_data[SNDRV_PCM_STREAM_CAPTURE]; - } else { - /* Using davinci-pcm */ - dai->playback_dma_data = mcasp->dma_params; - dai->capture_dma_data = mcasp->dma_params; - } + dai->playback_dma_data = &mcasp->dma_data[SNDRV_PCM_STREAM_PLAYBACK]; + dai->capture_dma_data = &mcasp->dma_data[SNDRV_PCM_STREAM_CAPTURE]; return 0; } @@ -1172,28 +1159,24 @@ static const struct snd_soc_component_driver davinci_mcasp_component = { static struct davinci_mcasp_pdata dm646x_mcasp_pdata = { .tx_dma_offset = 0x400, .rx_dma_offset = 0x400, - .asp_chan_q = EVENTQ_0, .version = MCASP_VERSION_1, }; static struct davinci_mcasp_pdata da830_mcasp_pdata = { .tx_dma_offset = 0x2000, .rx_dma_offset = 0x2000, - .asp_chan_q = EVENTQ_0, .version = MCASP_VERSION_2, }; static struct davinci_mcasp_pdata am33xx_mcasp_pdata = { .tx_dma_offset = 0, .rx_dma_offset = 0, - .asp_chan_q = EVENTQ_0, .version = MCASP_VERSION_3, }; static struct davinci_mcasp_pdata dra7_mcasp_pdata = { .tx_dma_offset = 0x200, .rx_dma_offset = 0x284, - .asp_chan_q = EVENTQ_0, .version = MCASP_VERSION_4, }; @@ -1370,12 +1353,12 @@ nodata: static int davinci_mcasp_probe(struct platform_device *pdev) { - struct davinci_pcm_dma_params *dma_params; struct snd_dmaengine_dai_dma_data *dma_data; struct resource *mem, *ioarea, *res, *dat; struct davinci_mcasp_pdata *pdata; struct davinci_mcasp *mcasp; char *irq_name; + int *dma; int irq; int ret; @@ -1509,59 +1492,45 @@ static int davinci_mcasp_probe(struct platform_device *pdev) if (dat) mcasp->dat_port = true; - dma_params = &mcasp->dma_params[SNDRV_PCM_STREAM_PLAYBACK]; dma_data = &mcasp->dma_data[SNDRV_PCM_STREAM_PLAYBACK]; - dma_params->asp_chan_q = pdata->asp_chan_q; - dma_params->ram_chan_q = pdata->ram_chan_q; - dma_params->sram_pool = pdata->sram_pool; - dma_params->sram_size = pdata->sram_size_playback; if (dat) - dma_params->dma_addr = dat->start; + dma_data->addr = dat->start; else - dma_params->dma_addr = mem->start + pdata->tx_dma_offset; - - /* Unconditional dmaengine stuff */ - dma_data->addr = dma_params->dma_addr; + dma_data->addr = mem->start + pdata->tx_dma_offset; + dma = &mcasp->dma_request[SNDRV_PCM_STREAM_PLAYBACK]; res = platform_get_resource(pdev, IORESOURCE_DMA, 0); if (res) - dma_params->channel = res->start; + *dma = res->start; else - dma_params->channel = pdata->tx_dma_channel; + *dma = pdata->tx_dma_channel; /* dmaengine filter data for DT and non-DT boot */ if (pdev->dev.of_node) dma_data->filter_data = "tx"; else - dma_data->filter_data = &dma_params->channel; + dma_data->filter_data = dma; /* RX is not valid in DIT mode */ if (mcasp->op_mode != DAVINCI_MCASP_DIT_MODE) { - dma_params = &mcasp->dma_params[SNDRV_PCM_STREAM_CAPTURE]; dma_data = &mcasp->dma_data[SNDRV_PCM_STREAM_CAPTURE]; - dma_params->asp_chan_q = pdata->asp_chan_q; - dma_params->ram_chan_q = pdata->ram_chan_q; - dma_params->sram_pool = pdata->sram_pool; - dma_params->sram_size = pdata->sram_size_capture; if (dat) - dma_params->dma_addr = dat->start; + dma_data->addr = dat->start; else - dma_params->dma_addr = mem->start + pdata->rx_dma_offset; - - /* Unconditional dmaengine stuff */ - dma_data->addr = dma_params->dma_addr; + dma_data->addr = mem->start + pdata->rx_dma_offset; + dma = &mcasp->dma_request[SNDRV_PCM_STREAM_CAPTURE]; res = platform_get_resource(pdev, IORESOURCE_DMA, 1); if (res) - dma_params->channel = res->start; + *dma = res->start; else - dma_params->channel = pdata->rx_dma_channel; + *dma = pdata->rx_dma_channel; /* dmaengine filter data for DT and non-DT boot */ if (pdev->dev.of_node) dma_data->filter_data = "rx"; else - dma_data->filter_data = &dma_params->channel; + dma_data->filter_data = dma; } if (mcasp->version < MCASP_VERSION_3) { @@ -1584,17 +1553,11 @@ static int davinci_mcasp_probe(struct platform_device *pdev) goto err; switch (mcasp->version) { -#if IS_BUILTIN(CONFIG_SND_DAVINCI_SOC) || \ - (IS_MODULE(CONFIG_SND_DAVINCI_SOC_MCASP) && \ - IS_MODULE(CONFIG_SND_DAVINCI_SOC)) - case MCASP_VERSION_1: - case MCASP_VERSION_2: - ret = davinci_soc_platform_register(&pdev->dev); - break; -#endif #if IS_BUILTIN(CONFIG_SND_EDMA_SOC) || \ (IS_MODULE(CONFIG_SND_DAVINCI_SOC_MCASP) && \ IS_MODULE(CONFIG_SND_EDMA_SOC)) + case MCASP_VERSION_1: + case MCASP_VERSION_2: case MCASP_VERSION_3: ret = edma_pcm_platform_register(&pdev->dev); break; diff --git a/sound/soc/davinci/davinci-pcm.c b/sound/soc/davinci/davinci-pcm.c deleted file mode 100644 index 7809e9d..0000000 --- a/sound/soc/davinci/davinci-pcm.c +++ /dev/null @@ -1,861 +0,0 @@ -/* - * ALSA PCM interface for the TI DAVINCI processor - * - * Author: Vladimir Barinov, <vbarinov@embeddedalley.com> - * Copyright: (C) 2007 MontaVista Software, Inc., <source@mvista.com> - * added SRAM ping/pong (C) 2008 Troy Kisky <troy.kisky@boundarydevices.com> - * - * 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. - */ - -#include <linux/module.h> -#include <linux/init.h> -#include <linux/platform_device.h> -#include <linux/slab.h> -#include <linux/dma-mapping.h> -#include <linux/kernel.h> -#include <linux/genalloc.h> -#include <linux/platform_data/edma.h> - -#include <sound/core.h> -#include <sound/pcm.h> -#include <sound/pcm_params.h> -#include <sound/soc.h> - -#include <asm/dma.h> - -#include "davinci-pcm.h" - -#ifdef DEBUG -static void print_buf_info(int slot, char *name) -{ - struct edmacc_param p; - if (slot < 0) - return; - edma_read_slot(slot, &p); - printk(KERN_DEBUG "%s: 0x%x, opt=%x, src=%x, a_b_cnt=%x dst=%x\n", - name, slot, p.opt, p.src, p.a_b_cnt, p.dst); - printk(KERN_DEBUG " src_dst_bidx=%x link_bcntrld=%x src_dst_cidx=%x ccnt=%x\n", - p.src_dst_bidx, p.link_bcntrld, p.src_dst_cidx, p.ccnt); -} -#else -static void print_buf_info(int slot, char *name) -{ -} -#endif - -static struct snd_pcm_hardware pcm_hardware_playback = { - .info = (SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER | - SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID | - SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME| - SNDRV_PCM_INFO_BATCH), - .buffer_bytes_max = 128 * 1024, - .period_bytes_min = 32, - .period_bytes_max = 8 * 1024, - .periods_min = 16, - .periods_max = 255, - .fifo_size = 0, -}; - -static struct snd_pcm_hardware pcm_hardware_capture = { - .info = (SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER | - SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID | - SNDRV_PCM_INFO_PAUSE | - SNDRV_PCM_INFO_BATCH), - .buffer_bytes_max = 128 * 1024, - .period_bytes_min = 32, - .period_bytes_max = 8 * 1024, - .periods_min = 16, - .periods_max = 255, - .fifo_size = 0, -}; - -/* - * How ping/pong works.... - * - * Playback: - * ram_params - copys 2*ping_size from start of SDRAM to iram, - * links to ram_link2 - * ram_link2 - copys rest of SDRAM to iram in ping_size units, - * links to ram_link - * ram_link - copys entire SDRAM to iram in ping_size uints, - * links to self - * - * asp_params - same as asp_link[0] - * asp_link[0] - copys from lower half of iram to asp port - * links to asp_link[1], triggers iram copy event on completion - * asp_link[1] - copys from upper half of iram to asp port - * links to asp_link[0], triggers iram copy event on completion - * triggers interrupt only needed to let upper SOC levels update position - * in stream on completion - * - * When playback is started: - * ram_params started - * asp_params started - * - * Capture: - * ram_params - same as ram_link, - * links to ram_link - * ram_link - same as playback - * links to self - * - * asp_params - same as playback - * asp_link[0] - same as playback - * asp_link[1] - same as playback - * - * When capture is started: - * asp_params started - */ -struct davinci_runtime_data { - spinlock_t lock; - int period; /* current DMA period */ - int asp_channel; /* Master DMA channel */ - int asp_link[2]; /* asp parameter link channel, ping/pong */ - struct davinci_pcm_dma_params *params; /* DMA params */ - int ram_channel; - int ram_link; - int ram_link2; - struct edmacc_param asp_params; - struct edmacc_param ram_params; -}; - -static void davinci_pcm_period_elapsed(struct snd_pcm_substream *substream) -{ - struct davinci_runtime_data *prtd = substream->runtime->private_data; - struct snd_pcm_runtime *runtime = substream->runtime; - - prtd->period++; - if (unlikely(prtd->period >= runtime->periods)) - prtd->period = 0; -} - -static void davinci_pcm_period_reset(struct snd_pcm_substream *substream) -{ - struct davinci_runtime_data *prtd = substream->runtime->private_data; - - prtd->period = 0; -} -/* - * Not used with ping/pong - */ -static void davinci_pcm_enqueue_dma(struct snd_pcm_substream *substream) -{ - struct davinci_runtime_data *prtd = substream->runtime->private_data; - struct snd_pcm_runtime *runtime = substream->runtime; - unsigned int period_size; - unsigned int dma_offset; - dma_addr_t dma_pos; - dma_addr_t src, dst; - unsigned short src_bidx, dst_bidx; - unsigned short src_cidx, dst_cidx; - unsigned int data_type; - unsigned short acnt; - unsigned int count; - unsigned int fifo_level; - - period_size = snd_pcm_lib_period_bytes(substream); - dma_offset = prtd->period * period_size; - dma_pos = runtime->dma_addr + dma_offset; - fifo_level = prtd->params->fifo_level; - - pr_debug("davinci_pcm: audio_set_dma_params_play channel = %d " - "dma_ptr = %x period_size=%x\n", prtd->asp_link[0], dma_pos, - period_size); - - data_type = prtd->params->data_type; - count = period_size / data_type; - if (fifo_level) - count /= fifo_level; - - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - src = dma_pos; - dst = prtd->params->dma_addr; - src_bidx = data_type; - dst_bidx = 4; - src_cidx = data_type * fifo_level; - dst_cidx = 0; - } else { - src = prtd->params->dma_addr; - dst = dma_pos; - src_bidx = 0; - dst_bidx = data_type; - src_cidx = 0; - dst_cidx = data_type * fifo_level; - } - - acnt = prtd->params->acnt; - edma_set_src(prtd->asp_link[0], src, INCR, W8BIT); - edma_set_dest(prtd->asp_link[0], dst, INCR, W8BIT); - - edma_set_src_index(prtd->asp_link[0], src_bidx, src_cidx); - edma_set_dest_index(prtd->asp_link[0], dst_bidx, dst_cidx); - - if (!fifo_level) - edma_set_transfer_params(prtd->asp_link[0], acnt, count, 1, 0, - ASYNC); - else - edma_set_transfer_params(prtd->asp_link[0], acnt, - fifo_level, - count, fifo_level, - ABSYNC); -} - -static void davinci_pcm_dma_irq(unsigned link, u16 ch_status, void *data) -{ - struct snd_pcm_substream *substream = data; - struct davinci_runtime_data *prtd = substream->runtime->private_data; - - print_buf_info(prtd->ram_channel, "i ram_channel"); - pr_debug("davinci_pcm: link=%d, status=0x%x\n", link, ch_status); - - if (unlikely(ch_status != EDMA_DMA_COMPLETE)) - return; - - if (snd_pcm_running(substream)) { - spin_lock(&prtd->lock); - if (prtd->ram_channel < 0) { - /* No ping/pong must fix up link dma data*/ - davinci_pcm_enqueue_dma(substream); - } - davinci_pcm_period_elapsed(substream); - spin_unlock(&prtd->lock); - snd_pcm_period_elapsed(substream); - } -} - -#ifdef CONFIG_GENERIC_ALLOCATOR -static int allocate_sram(struct snd_pcm_substream *substream, - struct gen_pool *sram_pool, unsigned size, - struct snd_pcm_hardware *ppcm) -{ - struct snd_dma_buffer *buf = &substream->dma_buffer; - struct snd_dma_buffer *iram_dma = NULL; - dma_addr_t iram_phys = 0; - void *iram_virt = NULL; - - if (buf->private_data || !size) - return 0; - - ppcm->period_bytes_max = size; - iram_virt = gen_pool_dma_alloc(sram_pool, size, &iram_phys); - if (!iram_virt) - goto exit1; - iram_dma = kzalloc(sizeof(*iram_dma), GFP_KERNEL); - if (!iram_dma) - goto exit2; - iram_dma->area = iram_virt; - iram_dma->addr = iram_phys; - memset(iram_dma->area, 0, size); - iram_dma->bytes = size; - buf->private_data = iram_dma; - return 0; -exit2: - if (iram_virt) - gen_pool_free(sram_pool, (unsigned)iram_virt, size); -exit1: - return -ENOMEM; -} - -static void davinci_free_sram(struct snd_pcm_substream *substream, - struct snd_dma_buffer *iram_dma) -{ - struct davinci_runtime_data *prtd = substream->runtime->private_data; - struct gen_pool *sram_pool = prtd->params->sram_pool; - - gen_pool_free(sram_pool, (unsigned) iram_dma->area, iram_dma->bytes); -} -#else -static int allocate_sram(struct snd_pcm_substream *substream, - struct gen_pool *sram_pool, unsigned size, - struct snd_pcm_hardware *ppcm) -{ - return 0; -} - -static void davinci_free_sram(struct snd_pcm_substream *substream, - struct snd_dma_buffer *iram_dma) -{ -} -#endif - -/* - * Only used with ping/pong. - * This is called after runtime->dma_addr, period_bytes and data_type are valid - */ -static int ping_pong_dma_setup(struct snd_pcm_substream *substream) -{ - unsigned short ram_src_cidx, ram_dst_cidx; - struct snd_pcm_runtime *runtime = substream->runtime; - struct davinci_runtime_data *prtd = runtime->private_data; - struct snd_dma_buffer *iram_dma = - (struct snd_dma_buffer *)substream->dma_buffer.private_data; - struct davinci_pcm_dma_params *params = prtd->params; - unsigned int data_type = params->data_type; - unsigned int acnt = params->acnt; - /* divide by 2 for ping/pong */ - unsigned int ping_size = snd_pcm_lib_period_bytes(substream) >> 1; - unsigned int fifo_level = prtd->params->fifo_level; - unsigned int count; - if ((data_type == 0) || (data_type > 4)) { - printk(KERN_ERR "%s: data_type=%i\n", __func__, data_type); - return -EINVAL; - } - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - dma_addr_t asp_src_pong = iram_dma->addr + ping_size; - ram_src_cidx = ping_size; - ram_dst_cidx = -ping_size; - edma_set_src(prtd->asp_link[1], asp_src_pong, INCR, W8BIT); - - edma_set_src_index(prtd->asp_link[0], data_type, - data_type * fifo_level); - edma_set_src_index(prtd->asp_link[1], data_type, - data_type * fifo_level); - - edma_set_src(prtd->ram_link, runtime->dma_addr, INCR, W32BIT); - } else { - dma_addr_t asp_dst_pong = iram_dma->addr + ping_size; - ram_src_cidx = -ping_size; - ram_dst_cidx = ping_size; - edma_set_dest(prtd->asp_link[1], asp_dst_pong, INCR, W8BIT); - - edma_set_dest_index(prtd->asp_link[0], data_type, - data_type * fifo_level); - edma_set_dest_index(prtd->asp_link[1], data_type, - data_type * fifo_level); - - edma_set_dest(prtd->ram_link, runtime->dma_addr, INCR, W32BIT); - } - - if (!fifo_level) { - count = ping_size / data_type; - edma_set_transfer_params(prtd->asp_link[0], acnt, count, - 1, 0, ASYNC); - edma_set_transfer_params(prtd->asp_link[1], acnt, count, - 1, 0, ASYNC); - } else { - count = ping_size / (data_type * fifo_level); - edma_set_transfer_params(prtd->asp_link[0], acnt, fifo_level, - count, fifo_level, ABSYNC); - edma_set_transfer_params(prtd->asp_link[1], acnt, fifo_level, - count, fifo_level, ABSYNC); - } - - edma_set_src_index(prtd->ram_link, ping_size, ram_src_cidx); - edma_set_dest_index(prtd->ram_link, ping_size, ram_dst_cidx); - edma_set_transfer_params(prtd->ram_link, ping_size, 2, - runtime->periods, 2, ASYNC); - - /* init master params */ - edma_read_slot(prtd->asp_link[0], &prtd->asp_params); - edma_read_slot(prtd->ram_link, &prtd->ram_params); - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - struct edmacc_param p_ram; - /* Copy entire iram buffer before playback started */ - prtd->ram_params.a_b_cnt = (1 << 16) | (ping_size << 1); - /* 0 dst_bidx */ - prtd->ram_params.src_dst_bidx = (ping_size << 1); - /* 0 dst_cidx */ - prtd->ram_params.src_dst_cidx = (ping_size << 1); - prtd->ram_params.ccnt = 1; - - /* Skip 1st period */ - edma_read_slot(prtd->ram_link, &p_ram); - p_ram.src += (ping_size << 1); - p_ram.ccnt -= 1; - edma_write_slot(prtd->ram_link2, &p_ram); - /* - * When 1st started, ram -> iram dma channel will fill the - * entire iram. Then, whenever a ping/pong asp buffer finishes, - * 1/2 iram will be filled. - */ - prtd->ram_params.link_bcntrld = - EDMA_CHAN_SLOT(prtd->ram_link2) << 5; - } - return 0; -} - -/* 1 asp tx or rx channel using 2 parameter channels - * 1 ram to/from iram channel using 1 parameter channel - * - * Playback - * ram copy channel kicks off first, - * 1st ram copy of entire iram buffer completion kicks off asp channel - * asp tcc always kicks off ram copy of 1/2 iram buffer - * - * Record - * asp channel starts, tcc kicks off ram copy - */ -static int request_ping_pong(struct snd_pcm_substream *substream, - struct davinci_runtime_data *prtd, - struct snd_dma_buffer *iram_dma) -{ - dma_addr_t asp_src_ping; - dma_addr_t asp_dst_ping; - int ret; - struct davinci_pcm_dma_params *params = prtd->params; - - /* Request ram master channel */ - ret = prtd->ram_channel = edma_alloc_channel(EDMA_CHANNEL_ANY, - davinci_pcm_dma_irq, substream, - prtd->params->ram_chan_q); - if (ret < 0) - goto exit1; - - /* Request ram link channel */ - ret = prtd->ram_link = edma_alloc_slot( - EDMA_CTLR(prtd->ram_channel), EDMA_SLOT_ANY); - if (ret < 0) - goto exit2; - - ret = prtd->asp_link[1] = edma_alloc_slot( - EDMA_CTLR(prtd->asp_channel), EDMA_SLOT_ANY); - if (ret < 0) - goto exit3; - - prtd->ram_link2 = -1; - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - ret = prtd->ram_link2 = edma_alloc_slot( - EDMA_CTLR(prtd->ram_channel), EDMA_SLOT_ANY); - if (ret < 0) - goto exit4; - } - /* circle ping-pong buffers */ - edma_link(prtd->asp_link[0], prtd->asp_link[1]); - edma_link(prtd->asp_link[1], prtd->asp_link[0]); - /* circle ram buffers */ - edma_link(prtd->ram_link, prtd->ram_link); - - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - asp_src_ping = iram_dma->addr; - asp_dst_ping = params->dma_addr; /* fifo */ - } else { - asp_src_ping = params->dma_addr; /* fifo */ - asp_dst_ping = iram_dma->addr; - } - /* ping */ - edma_set_src(prtd->asp_link[0], asp_src_ping, INCR, W16BIT); - edma_set_dest(prtd->asp_link[0], asp_dst_ping, INCR, W16BIT); - edma_set_src_index(prtd->asp_link[0], 0, 0); - edma_set_dest_index(prtd->asp_link[0], 0, 0); - - edma_read_slot(prtd->asp_link[0], &prtd->asp_params); - prtd->asp_params.opt &= ~(TCCMODE | EDMA_TCC(0x3f) | TCINTEN); - prtd->asp_params.opt |= TCCHEN | - EDMA_TCC(prtd->ram_channel & 0x3f); - edma_write_slot(prtd->asp_link[0], &prtd->asp_params); - - /* pong */ - edma_set_src(prtd->asp_link[1], asp_src_ping, INCR, W16BIT); - edma_set_dest(prtd->asp_link[1], asp_dst_ping, INCR, W16BIT); - edma_set_src_index(prtd->asp_link[1], 0, 0); - edma_set_dest_index(prtd->asp_link[1], 0, 0); - - edma_read_slot(prtd->asp_link[1], &prtd->asp_params); - prtd->asp_params.opt &= ~(TCCMODE | EDMA_TCC(0x3f)); - /* interrupt after every pong completion */ - prtd->asp_params.opt |= TCINTEN | TCCHEN | - EDMA_TCC(prtd->ram_channel & 0x3f); - edma_write_slot(prtd->asp_link[1], &prtd->asp_params); - - /* ram */ - edma_set_src(prtd->ram_link, iram_dma->addr, INCR, W32BIT); - edma_set_dest(prtd->ram_link, iram_dma->addr, INCR, W32BIT); - pr_debug("%s: audio dma channels/slots in use for ram:%u %u %u," - "for asp:%u %u %u\n", __func__, - prtd->ram_channel, prtd->ram_link, prtd->ram_link2, - prtd->asp_channel, prtd->asp_link[0], - prtd->asp_link[1]); - return 0; -exit4: - edma_free_channel(prtd->asp_link[1]); - prtd->asp_link[1] = -1; -exit3: - edma_free_channel(prtd->ram_link); - prtd->ram_link = -1; -exit2: - edma_free_channel(prtd->ram_channel); - prtd->ram_channel = -1; -exit1: - return ret; -} - -static int davinci_pcm_dma_request(struct snd_pcm_substream *substream) -{ - struct snd_dma_buffer *iram_dma; - struct davinci_runtime_data *prtd = substream->runtime->private_data; - struct davinci_pcm_dma_params *params = prtd->params; - int ret; - - if (!params) - return -ENODEV; - - /* Request asp master DMA channel */ - ret = prtd->asp_channel = edma_alloc_channel(params->channel, - davinci_pcm_dma_irq, substream, - prtd->params->asp_chan_q); - if (ret < 0) - goto exit1; - - /* Request asp link channels */ - ret = prtd->asp_link[0] = edma_alloc_slot( - EDMA_CTLR(prtd->asp_channel), EDMA_SLOT_ANY); - if (ret < 0) - goto exit2; - - iram_dma = (struct snd_dma_buffer *)substream->dma_buffer.private_data; - if (iram_dma) { - if (request_ping_pong(substream, prtd, iram_dma) == 0) - return 0; - printk(KERN_WARNING "%s: dma channel allocation failed," - "not using sram\n", __func__); - } - - /* Issue transfer completion IRQ when the channel completes a - * transfer, then always reload from the same slot (by a kind - * of loopback link). The completion IRQ handler will update - * the reload slot with a new buffer. - * - * REVISIT save p_ram here after setting up everything except - * the buffer and its length (ccnt) ... use it as a template - * so davinci_pcm_enqueue_dma() takes less time in IRQ. - */ - edma_read_slot(prtd->asp_link[0], &prtd->asp_params); - prtd->asp_params.opt |= TCINTEN | - EDMA_TCC(EDMA_CHAN_SLOT(prtd->asp_channel)); - prtd->asp_params.link_bcntrld = EDMA_CHAN_SLOT(prtd->asp_link[0]) << 5; - edma_write_slot(prtd->asp_link[0], &prtd->asp_params); - return 0; -exit2: - edma_free_channel(prtd->asp_channel); - prtd->asp_channel = -1; -exit1: - return ret; -} - -static int davinci_pcm_trigger(struct snd_pcm_substream *substream, int cmd) -{ - struct davinci_runtime_data *prtd = substream->runtime->private_data; - int ret = 0; - - spin_lock(&prtd->lock); - - switch (cmd) { - case SNDRV_PCM_TRIGGER_START: - edma_start(prtd->asp_channel); - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK && - prtd->ram_channel >= 0) { - /* copy 1st iram buffer */ - edma_start(prtd->ram_channel); - } - break; - case SNDRV_PCM_TRIGGER_RESUME: - case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - edma_resume(prtd->asp_channel); - break; - case SNDRV_PCM_TRIGGER_STOP: - case SNDRV_PCM_TRIGGER_SUSPEND: - case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - edma_pause(prtd->asp_channel); - break; - default: - ret = -EINVAL; - break; - } - - spin_unlock(&prtd->lock); - - return ret; -} - -static int davinci_pcm_prepare(struct snd_pcm_substream *substream) -{ - struct davinci_runtime_data *prtd = substream->runtime->private_data; - - davinci_pcm_period_reset(substream); - if (prtd->ram_channel >= 0) { - int ret = ping_pong_dma_setup(substream); - if (ret < 0) - return ret; - - edma_write_slot(prtd->ram_channel, &prtd->ram_params); - edma_write_slot(prtd->asp_channel, &prtd->asp_params); - - print_buf_info(prtd->ram_channel, "ram_channel"); - print_buf_info(prtd->ram_link, "ram_link"); - print_buf_info(prtd->ram_link2, "ram_link2"); - print_buf_info(prtd->asp_channel, "asp_channel"); - print_buf_info(prtd->asp_link[0], "asp_link[0]"); - print_buf_info(prtd->asp_link[1], "asp_link[1]"); - - /* - * There is a phase offset of 2 periods between the position - * used by dma setup and the position reported in the pointer - * function. - * - * The phase offset, when not using ping-pong buffers, is due to - * the two consecutive calls to davinci_pcm_enqueue_dma() below. - * - * Whereas here, with ping-pong buffers, the phase is due to - * there being an entire buffer transfer complete before the - * first dma completion event triggers davinci_pcm_dma_irq(). - */ - davinci_pcm_period_elapsed(substream); - davinci_pcm_period_elapsed(substream); - - return 0; - } - davinci_pcm_enqueue_dma(substream); - davinci_pcm_period_elapsed(substream); - - /* Copy self-linked parameter RAM entry into master channel */ - edma_read_slot(prtd->asp_link[0], &prtd->asp_params); - edma_write_slot(prtd->asp_channel, &prtd->asp_params); - davinci_pcm_enqueue_dma(substream); - davinci_pcm_period_elapsed(substream); - - return 0; -} - -static snd_pcm_uframes_t -davinci_pcm_pointer(struct snd_pcm_substream *substream) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - struct davinci_runtime_data *prtd = runtime->private_data; - unsigned int offset; - int asp_count; - unsigned int period_size = snd_pcm_lib_period_bytes(substream); - - /* - * There is a phase offset of 2 periods between the position used by dma - * setup and the position reported in the pointer function. Either +2 in - * the dma setup or -2 here in the pointer function (with wrapping, - * both) accounts for this offset -- choose the latter since it makes - * the first-time setup clearer. - */ - spin_lock(&prtd->lock); - asp_count = prtd->period - 2; - spin_unlock(&prtd->lock); - - if (asp_count < 0) - asp_count += runtime->periods; - asp_count *= period_size; - - offset = bytes_to_frames(runtime, asp_count); - if (offset >= runtime->buffer_size) - offset = 0; - - return offset; -} - -static int davinci_pcm_open(struct snd_pcm_substream *substream) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - struct davinci_runtime_data *prtd; - struct snd_pcm_hardware *ppcm; - int ret = 0; - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct davinci_pcm_dma_params *pa; - struct davinci_pcm_dma_params *params; - - pa = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); - if (!pa) - return -ENODEV; - params = &pa[substream->stream]; - - ppcm = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ? - &pcm_hardware_playback : &pcm_hardware_capture; - allocate_sram(substream, params->sram_pool, params->sram_size, ppcm); - snd_soc_set_runtime_hwparams(substream, ppcm); - /* ensure that buffer size is a multiple of period size */ - ret = snd_pcm_hw_constraint_integer(runtime, - SNDRV_PCM_HW_PARAM_PERIODS); - if (ret < 0) - return ret; - - prtd = kzalloc(sizeof(struct davinci_runtime_data), GFP_KERNEL); - if (prtd == NULL) - return -ENOMEM; - - spin_lock_init(&prtd->lock); - prtd->params = params; - prtd->asp_channel = -1; - prtd->asp_link[0] = prtd->asp_link[1] = -1; - prtd->ram_channel = -1; - prtd->ram_link = -1; - prtd->ram_link2 = -1; - - runtime->private_data = prtd; - - ret = davinci_pcm_dma_request(substream); - if (ret) { - printk(KERN_ERR "davinci_pcm: Failed to get dma channels\n"); - kfree(prtd); - } - - return ret; -} - -static int davinci_pcm_close(struct snd_pcm_substream *substream) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - struct davinci_runtime_data *prtd = runtime->private_data; - - if (prtd->ram_channel >= 0) - edma_stop(prtd->ram_channel); - if (prtd->asp_channel >= 0) - edma_stop(prtd->asp_channel); - if (prtd->asp_link[0] >= 0) - edma_unlink(prtd->asp_link[0]); - if (prtd->asp_link[1] >= 0) - edma_unlink(prtd->asp_link[1]); - if (prtd->ram_link >= 0) - edma_unlink(prtd->ram_link); - - if (prtd->asp_link[0] >= 0) - edma_free_slot(prtd->asp_link[0]); - if (prtd->asp_link[1] >= 0) - edma_free_slot(prtd->asp_link[1]); - if (prtd->asp_channel >= 0) - edma_free_channel(prtd->asp_channel); - if (prtd->ram_link >= 0) - edma_free_slot(prtd->ram_link); - if (prtd->ram_link2 >= 0) - edma_free_slot(prtd->ram_link2); - if (prtd->ram_channel >= 0) - edma_free_channel(prtd->ram_channel); - - kfree(prtd); - - return 0; -} - -static int davinci_pcm_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *hw_params) -{ - return snd_pcm_lib_malloc_pages(substream, - params_buffer_bytes(hw_params)); -} - -static int davinci_pcm_hw_free(struct snd_pcm_substream *substream) -{ - return snd_pcm_lib_free_pages(substream); -} - -static int davinci_pcm_mmap(struct snd_pcm_substream *substream, - struct vm_area_struct *vma) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - - return dma_mmap_writecombine(substream->pcm->card->dev, vma, - runtime->dma_area, - runtime->dma_addr, - runtime->dma_bytes); -} - -static struct snd_pcm_ops davinci_pcm_ops = { - .open = davinci_pcm_open, - .close = davinci_pcm_close, - .ioctl = snd_pcm_lib_ioctl, - .hw_params = davinci_pcm_hw_params, - .hw_free = davinci_pcm_hw_free, - .prepare = davinci_pcm_prepare, - .trigger = davinci_pcm_trigger, - .pointer = davinci_pcm_pointer, - .mmap = davinci_pcm_mmap, -}; - -static int davinci_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, int stream, - size_t size) -{ - struct snd_pcm_substream *substream = pcm->streams[stream].substream; - struct snd_dma_buffer *buf = &substream->dma_buffer; - - buf->dev.type = SNDRV_DMA_TYPE_DEV; - buf->dev.dev = pcm->card->dev; - buf->private_data = NULL; - buf->area = dma_alloc_writecombine(pcm->card->dev, size, - &buf->addr, GFP_KERNEL); - - pr_debug("davinci_pcm: preallocate_dma_buffer: area=%p, addr=%p, " - "size=%d\n", (void *) buf->area, (void *) buf->addr, size); - - if (!buf->area) - return -ENOMEM; - - buf->bytes = size; - return 0; -} - -static void davinci_pcm_free(struct snd_pcm *pcm) -{ - struct snd_pcm_substream *substream; - struct snd_dma_buffer *buf; - int stream; - - for (stream = 0; stream < 2; stream++) { - struct snd_dma_buffer *iram_dma; - substream = pcm->streams[stream].substream; - if (!substream) - continue; - - buf = &substream->dma_buffer; - if (!buf->area) - continue; - - dma_free_writecombine(pcm->card->dev, buf->bytes, - buf->area, buf->addr); - buf->area = NULL; - iram_dma = buf->private_data; - if (iram_dma) { - davinci_free_sram(substream, iram_dma); - kfree(iram_dma); - } - } -} - -static int davinci_pcm_new(struct snd_soc_pcm_runtime *rtd) -{ - struct snd_card *card = rtd->card->snd_card; - struct snd_pcm *pcm = rtd->pcm; - int ret; - - ret = dma_coerce_mask_and_coherent(card->dev, DMA_BIT_MASK(32)); - if (ret) - return ret; - - if (pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream) { - ret = davinci_pcm_preallocate_dma_buffer(pcm, - SNDRV_PCM_STREAM_PLAYBACK, - pcm_hardware_playback.buffer_bytes_max); - if (ret) - return ret; - } - - if (pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream) { - ret = davinci_pcm_preallocate_dma_buffer(pcm, - SNDRV_PCM_STREAM_CAPTURE, - pcm_hardware_capture.buffer_bytes_max); - if (ret) - return ret; - } - - return 0; -} - -static struct snd_soc_platform_driver davinci_soc_platform = { - .ops = &davinci_pcm_ops, - .pcm_new = davinci_pcm_new, - .pcm_free = davinci_pcm_free, -}; - -int davinci_soc_platform_register(struct device *dev) -{ - return devm_snd_soc_register_platform(dev, &davinci_soc_platform); -} -EXPORT_SYMBOL_GPL(davinci_soc_platform_register); - -MODULE_AUTHOR("Vladimir Barinov"); -MODULE_DESCRIPTION("TI DAVINCI PCM DMA module"); -MODULE_LICENSE("GPL"); diff --git a/sound/soc/davinci/davinci-pcm.h b/sound/soc/davinci/davinci-pcm.h deleted file mode 100644 index 0fe2346..0000000 --- a/sound/soc/davinci/davinci-pcm.h +++ /dev/null @@ -1,41 +0,0 @@ -/* - * ALSA PCM interface for the TI DAVINCI processor - * - * Author: Vladimir Barinov, <vbarinov@embeddedalley.com> - * Copyright: (C) 2007 MontaVista Software, Inc., <source@mvista.com> - * - * 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 _DAVINCI_PCM_H -#define _DAVINCI_PCM_H - -#include <linux/genalloc.h> -#include <linux/platform_data/davinci_asp.h> -#include <linux/platform_data/edma.h> - -struct davinci_pcm_dma_params { - int channel; /* sync dma channel ID */ - unsigned short acnt; - dma_addr_t dma_addr; /* device physical address for DMA */ - unsigned sram_size; - struct gen_pool *sram_pool; /* SRAM gen_pool for ping pong */ - enum dma_event_q asp_chan_q; /* event queue number for ASP channel */ - enum dma_event_q ram_chan_q; /* event queue number for RAM channel */ - unsigned char data_type; /* xfer data type */ - unsigned char convert_mono_stereo; - unsigned int fifo_level; -}; - -#if IS_ENABLED(CONFIG_SND_DAVINCI_SOC) -int davinci_soc_platform_register(struct device *dev); -#else -static inline int davinci_soc_platform_register(struct device *dev) -{ - return 0; -} -#endif /* CONFIG_SND_DAVINCI_SOC */ - -#endif diff --git a/sound/soc/davinci/davinci-vcif.c b/sound/soc/davinci/davinci-vcif.c index 5bee0427..fabd05f 100644 --- a/sound/soc/davinci/davinci-vcif.c +++ b/sound/soc/davinci/davinci-vcif.c @@ -33,8 +33,9 @@ #include <sound/pcm_params.h> #include <sound/initval.h> #include <sound/soc.h> +#include <sound/dmaengine_pcm.h> -#include "davinci-pcm.h" +#include "edma-pcm.h" #include "davinci-i2s.h" #define MOD_REG_BIT(val, mask, set) do { \ @@ -47,7 +48,8 @@ struct davinci_vcif_dev { struct davinci_vc *davinci_vc; - struct davinci_pcm_dma_params dma_params[2]; + struct snd_dmaengine_dai_dma_data dma_data[2]; + int dma_request[2]; }; static void davinci_vcif_start(struct snd_pcm_substream *substream) @@ -93,8 +95,6 @@ static int davinci_vcif_hw_params(struct snd_pcm_substream *substream, { struct davinci_vcif_dev *davinci_vcif_dev = snd_soc_dai_get_drvdata(dai); struct davinci_vc *davinci_vc = davinci_vcif_dev->davinci_vc; - struct davinci_pcm_dma_params *dma_params = - &davinci_vcif_dev->dma_params[substream->stream]; u32 w; /* Restart the codec before setup */ @@ -113,16 +113,12 @@ static int davinci_vcif_hw_params(struct snd_pcm_substream *substream, /* Determine xfer data type */ switch (params_format(params)) { case SNDRV_PCM_FORMAT_U8: - dma_params->data_type = 0; - MOD_REG_BIT(w, DAVINCI_VC_CTRL_RD_BITS_8 | DAVINCI_VC_CTRL_RD_UNSIGNED | DAVINCI_VC_CTRL_WD_BITS_8 | DAVINCI_VC_CTRL_WD_UNSIGNED, 1); break; case SNDRV_PCM_FORMAT_S8: - dma_params->data_type = 1; - MOD_REG_BIT(w, DAVINCI_VC_CTRL_RD_BITS_8 | DAVINCI_VC_CTRL_WD_BITS_8, 1); @@ -130,8 +126,6 @@ static int davinci_vcif_hw_params(struct snd_pcm_substream *substream, DAVINCI_VC_CTRL_WD_UNSIGNED, 0); break; case SNDRV_PCM_FORMAT_S16_LE: - dma_params->data_type = 2; - MOD_REG_BIT(w, DAVINCI_VC_CTRL_RD_BITS_8 | DAVINCI_VC_CTRL_RD_UNSIGNED | DAVINCI_VC_CTRL_WD_BITS_8 | @@ -142,8 +136,6 @@ static int davinci_vcif_hw_params(struct snd_pcm_substream *substream, return -EINVAL; } - dma_params->acnt = dma_params->data_type; - writel(w, davinci_vc->base + DAVINCI_VC_CTRL); return 0; @@ -172,24 +164,25 @@ static int davinci_vcif_trigger(struct snd_pcm_substream *substream, int cmd, return ret; } -static int davinci_vcif_startup(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai) -{ - struct davinci_vcif_dev *dev = snd_soc_dai_get_drvdata(dai); - - snd_soc_dai_set_dma_data(dai, substream, dev->dma_params); - return 0; -} - #define DAVINCI_VCIF_RATES SNDRV_PCM_RATE_8000_48000 static const struct snd_soc_dai_ops davinci_vcif_dai_ops = { - .startup = davinci_vcif_startup, .trigger = davinci_vcif_trigger, .hw_params = davinci_vcif_hw_params, }; +static int davinci_vcif_dai_probe(struct snd_soc_dai *dai) +{ + struct davinci_vcif_dev *dev = snd_soc_dai_get_drvdata(dai); + + dai->playback_dma_data = &dev->dma_data[SNDRV_PCM_STREAM_PLAYBACK]; + dai->capture_dma_data = &dev->dma_data[SNDRV_PCM_STREAM_CAPTURE]; + + return 0; +} + static struct snd_soc_dai_driver davinci_vcif_dai = { + .probe = davinci_vcif_dai_probe, .playback = { .channels_min = 1, .channels_max = 2, @@ -225,16 +218,16 @@ static int davinci_vcif_probe(struct platform_device *pdev) /* DMA tx params */ davinci_vcif_dev->davinci_vc = davinci_vc; - davinci_vcif_dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK].channel = - davinci_vc->davinci_vcif.dma_tx_channel; - davinci_vcif_dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK].dma_addr = - davinci_vc->davinci_vcif.dma_tx_addr; + davinci_vcif_dev->dma_data[SNDRV_PCM_STREAM_PLAYBACK].filter_data = + &davinci_vc->davinci_vcif.dma_tx_channel; + davinci_vcif_dev->dma_data[SNDRV_PCM_STREAM_PLAYBACK].addr = + davinci_vc->davinci_vcif.dma_tx_addr; /* DMA rx params */ - davinci_vcif_dev->dma_params[SNDRV_PCM_STREAM_CAPTURE].channel = - davinci_vc->davinci_vcif.dma_rx_channel; - davinci_vcif_dev->dma_params[SNDRV_PCM_STREAM_CAPTURE].dma_addr = - davinci_vc->davinci_vcif.dma_rx_addr; + davinci_vcif_dev->dma_data[SNDRV_PCM_STREAM_CAPTURE].filter_data = + &davinci_vc->davinci_vcif.dma_rx_channel; + davinci_vcif_dev->dma_data[SNDRV_PCM_STREAM_CAPTURE].addr = + davinci_vc->davinci_vcif.dma_rx_addr; dev_set_drvdata(&pdev->dev, davinci_vcif_dev); @@ -245,7 +238,7 @@ static int davinci_vcif_probe(struct platform_device *pdev) return ret; } - ret = davinci_soc_platform_register(&pdev->dev); + ret = edma_pcm_platform_register(&pdev->dev); if (ret) { dev_err(&pdev->dev, "register PCM failed: %d\n", ret); snd_soc_unregister_component(&pdev->dev); diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c index 3f6959c..de43887 100644 --- a/sound/soc/fsl/fsl-asoc-card.c +++ b/sound/soc/fsl/fsl-asoc-card.c @@ -512,6 +512,12 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) memcpy(priv->dai_link, fsl_asoc_card_dai, sizeof(struct snd_soc_dai_link) * ARRAY_SIZE(priv->dai_link)); + ret = snd_soc_of_parse_audio_routing(&priv->card, "audio-routing"); + if (ret) { + dev_err(&pdev->dev, "failed to parse audio-routing: %d\n", ret); + goto asrc_fail; + } + /* Normal DAI Link */ priv->dai_link[0].cpu_of_node = cpu_np; priv->dai_link[0].codec_of_node = codec_np; diff --git a/sound/soc/fsl/imx-es8328.c b/sound/soc/fsl/imx-es8328.c index f8cf10e..20e7400 100644 --- a/sound/soc/fsl/imx-es8328.c +++ b/sound/soc/fsl/imx-es8328.c @@ -53,9 +53,9 @@ static int imx_es8328_dai_init(struct snd_soc_pcm_runtime *rtd) /* Headphone jack detection */ if (gpio_is_valid(data->jack_gpio)) { - ret = snd_soc_jack_new(rtd->codec, "Headphone", - SND_JACK_HEADPHONE | SND_JACK_BTN_0, - &headset_jack); + ret = snd_soc_card_jack_new(rtd->card, "Headphone", + SND_JACK_HEADPHONE | SND_JACK_BTN_0, + &headset_jack, NULL, 0); if (ret) return ret; diff --git a/sound/soc/fsl/wm1133-ev1.c b/sound/soc/fsl/wm1133-ev1.c index a958937..0653aa8 100644 --- a/sound/soc/fsl/wm1133-ev1.c +++ b/sound/soc/fsl/wm1133-ev1.c @@ -205,16 +205,14 @@ static int wm1133_ev1_init(struct snd_soc_pcm_runtime *rtd) struct snd_soc_dapm_context *dapm = &codec->dapm; /* Headphone jack detection */ - snd_soc_jack_new(codec, "Headphone", SND_JACK_HEADPHONE, &hp_jack); - snd_soc_jack_add_pins(&hp_jack, ARRAY_SIZE(hp_jack_pins), - hp_jack_pins); + snd_soc_card_jack_new(rtd->card, "Headphone", SND_JACK_HEADPHONE, + &hp_jack, hp_jack_pins, ARRAY_SIZE(hp_jack_pins)); wm8350_hp_jack_detect(codec, WM8350_JDR, &hp_jack, SND_JACK_HEADPHONE); /* Microphone jack detection */ - snd_soc_jack_new(codec, "Microphone", - SND_JACK_MICROPHONE | SND_JACK_BTN_0, &mic_jack); - snd_soc_jack_add_pins(&mic_jack, ARRAY_SIZE(mic_jack_pins), - mic_jack_pins); + snd_soc_card_jack_new(rtd->card, "Microphone", + SND_JACK_MICROPHONE | SND_JACK_BTN_0, &mic_jack, + mic_jack_pins, ARRAY_SIZE(mic_jack_pins)); wm8350_mic_jack_detect(codec, &mic_jack, SND_JACK_MICROPHONE, SND_JACK_BTN_0); diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index fb550b5..c49a408 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -176,11 +176,11 @@ static int asoc_simple_card_dai_init(struct snd_soc_pcm_runtime *rtd) return ret; if (gpio_is_valid(priv->gpio_hp_det)) { - snd_soc_jack_new(codec->codec, "Headphones", SND_JACK_HEADPHONE, - &simple_card_hp_jack); - snd_soc_jack_add_pins(&simple_card_hp_jack, - ARRAY_SIZE(simple_card_hp_jack_pins), - simple_card_hp_jack_pins); + snd_soc_card_jack_new(rtd->card, "Headphones", + SND_JACK_HEADPHONE, + &simple_card_hp_jack, + simple_card_hp_jack_pins, + ARRAY_SIZE(simple_card_hp_jack_pins)); simple_card_hp_jack_gpio.gpio = priv->gpio_hp_det; simple_card_hp_jack_gpio.invert = priv->gpio_hp_det_invert; @@ -189,11 +189,11 @@ static int asoc_simple_card_dai_init(struct snd_soc_pcm_runtime *rtd) } if (gpio_is_valid(priv->gpio_mic_det)) { - snd_soc_jack_new(codec->codec, "Mic Jack", SND_JACK_MICROPHONE, - &simple_card_mic_jack); - snd_soc_jack_add_pins(&simple_card_mic_jack, - ARRAY_SIZE(simple_card_mic_jack_pins), - simple_card_mic_jack_pins); + snd_soc_card_jack_new(rtd->card, "Mic Jack", + SND_JACK_MICROPHONE, + &simple_card_mic_jack, + simple_card_mic_jack_pins, + ARRAY_SIZE(simple_card_mic_jack_pins)); simple_card_mic_jack_gpio.gpio = priv->gpio_mic_det; simple_card_mic_jack_gpio.invert = priv->gpio_mic_det_invert; snd_soc_jack_add_gpios(&simple_card_mic_jack, 1, diff --git a/sound/soc/intel/broadwell.c b/sound/soc/intel/broadwell.c index 9cf7d01..fc55420 100644 --- a/sound/soc/intel/broadwell.c +++ b/sound/soc/intel/broadwell.c @@ -80,15 +80,9 @@ static int broadwell_rt286_codec_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_codec *codec = rtd->codec; int ret = 0; - ret = snd_soc_jack_new(codec, "Headset", - SND_JACK_HEADSET | SND_JACK_BTN_0, &broadwell_headset); - - if (ret) - return ret; - - ret = snd_soc_jack_add_pins(&broadwell_headset, - ARRAY_SIZE(broadwell_headset_pins), - broadwell_headset_pins); + ret = snd_soc_card_jack_new(rtd->card, "Headset", + SND_JACK_HEADSET | SND_JACK_BTN_0, &broadwell_headset, + broadwell_headset_pins, ARRAY_SIZE(broadwell_headset_pins)); if (ret) return ret; @@ -110,9 +104,7 @@ static int broadwell_ssp0_fixup(struct snd_soc_pcm_runtime *rtd, channels->min = channels->max = 2; /* set SSP0 to 16 bit */ - snd_mask_set(¶ms->masks[SNDRV_PCM_HW_PARAM_FORMAT - - SNDRV_PCM_HW_PARAM_FIRST_MASK], - SNDRV_PCM_FORMAT_S16_LE); + params_set_format(params, SNDRV_PCM_FORMAT_S16_LE); return 0; } diff --git a/sound/soc/intel/byt-max98090.c b/sound/soc/intel/byt-max98090.c index 9832afe..d8b1f03 100644 --- a/sound/soc/intel/byt-max98090.c +++ b/sound/soc/intel/byt-max98090.c @@ -84,7 +84,6 @@ static struct snd_soc_jack_gpio hs_jack_gpios[] = { static int byt_max98090_init(struct snd_soc_pcm_runtime *runtime) { int ret; - struct snd_soc_codec *codec = runtime->codec; struct snd_soc_card *card = runtime->card; struct byt_max98090_private *drv = snd_soc_card_get_drvdata(card); struct snd_soc_jack *jack = &drv->jack; @@ -100,13 +99,9 @@ static int byt_max98090_init(struct snd_soc_pcm_runtime *runtime) } /* Enable jack detection */ - ret = snd_soc_jack_new(codec, "Headset", - SND_JACK_LINEOUT | SND_JACK_HEADSET, jack); - if (ret) - return ret; - - ret = snd_soc_jack_add_pins(jack, ARRAY_SIZE(hs_jack_pins), - hs_jack_pins); + ret = snd_soc_card_jack_new(runtime->card, "Headset", + SND_JACK_LINEOUT | SND_JACK_HEADSET, jack, + hs_jack_pins, ARRAY_SIZE(hs_jack_pins)); if (ret) return ret; diff --git a/sound/soc/intel/bytcr_dpcm_rt5640.c b/sound/soc/intel/bytcr_dpcm_rt5640.c index 5930862..3b262d0 100644 --- a/sound/soc/intel/bytcr_dpcm_rt5640.c +++ b/sound/soc/intel/bytcr_dpcm_rt5640.c @@ -113,9 +113,7 @@ static int byt_codec_fixup(struct snd_soc_pcm_runtime *rtd, channels->min = channels->max = 2; /* set SSP2 to 24-bit */ - snd_mask_set(¶ms->masks[SNDRV_PCM_HW_PARAM_FORMAT - - SNDRV_PCM_HW_PARAM_FIRST_MASK], - SNDRV_PCM_FORMAT_S24_LE); + params_set_format(params, SNDRV_PCM_FORMAT_S24_LE); return 0; } diff --git a/sound/soc/intel/cht_bsw_rt5645.c b/sound/soc/intel/cht_bsw_rt5645.c index bd29617..0122279 100644 --- a/sound/soc/intel/cht_bsw_rt5645.c +++ b/sound/soc/intel/cht_bsw_rt5645.c @@ -169,17 +169,17 @@ static int cht_codec_init(struct snd_soc_pcm_runtime *runtime) return ret; } - ret = snd_soc_jack_new(codec, "Headphone Jack", - SND_JACK_HEADPHONE, - &ctx->hp_jack); + ret = snd_soc_card_jack_new(runtime->card, "Headphone Jack", + SND_JACK_HEADPHONE, &ctx->hp_jack, + NULL, 0); if (ret) { dev_err(runtime->dev, "HP jack creation failed %d\n", ret); return ret; } - ret = snd_soc_jack_new(codec, "Mic Jack", - SND_JACK_MICROPHONE, - &ctx->mic_jack); + ret = snd_soc_card_jack_new(runtime->card, "Mic Jack", + SND_JACK_MICROPHONE, &ctx->mic_jack, + NULL, 0); if (ret) { dev_err(runtime->dev, "Mic jack creation failed %d\n", ret); return ret; @@ -203,9 +203,7 @@ static int cht_codec_fixup(struct snd_soc_pcm_runtime *rtd, channels->min = channels->max = 2; /* set SSP2 to 24-bit */ - snd_mask_set(¶ms->masks[SNDRV_PCM_HW_PARAM_FORMAT - - SNDRV_PCM_HW_PARAM_FIRST_MASK], - SNDRV_PCM_FORMAT_S24_LE); + params_set_format(params, SNDRV_PCM_FORMAT_S24_LE); return 0; } diff --git a/sound/soc/intel/cht_bsw_rt5672.c b/sound/soc/intel/cht_bsw_rt5672.c index ff01662..bc8dcac 100644 --- a/sound/soc/intel/cht_bsw_rt5672.c +++ b/sound/soc/intel/cht_bsw_rt5672.c @@ -178,9 +178,7 @@ static int cht_codec_fixup(struct snd_soc_pcm_runtime *rtd, channels->min = channels->max = 2; /* set SSP2 to 24-bit */ - snd_mask_set(¶ms->masks[SNDRV_PCM_HW_PARAM_FORMAT - - SNDRV_PCM_HW_PARAM_FIRST_MASK], - SNDRV_PCM_FORMAT_S24_LE); + params_set_format(params, SNDRV_PCM_FORMAT_S24_LE); return 0; } @@ -217,7 +215,7 @@ static struct snd_soc_dai_link cht_dailink[] = { .codec_dai_name = "snd-soc-dummy-dai", .codec_name = "snd-soc-dummy", .platform_name = "sst-mfld-platform", - .ignore_suspend = 1, + .nonatomic = true, .dynamic = 1, .dpcm_playback = 1, .dpcm_capture = 1, @@ -240,13 +238,13 @@ static struct snd_soc_dai_link cht_dailink[] = { .cpu_dai_name = "ssp2-port", .platform_name = "sst-mfld-platform", .no_pcm = 1, + .nonatomic = true, .codec_dai_name = "rt5670-aif1", .codec_name = "i2c-10EC5670:00", .dai_fmt = SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_IB_NF | SND_SOC_DAIFMT_CBS_CFS, .init = cht_codec_init, .be_hw_params_fixup = cht_codec_fixup, - .ignore_suspend = 1, .dpcm_playback = 1, .dpcm_capture = 1, .ops = &cht_be_ssp2_ops, @@ -285,7 +283,6 @@ static int snd_cht_mc_probe(struct platform_device *pdev) static struct platform_driver snd_cht_mc_driver = { .driver = { .name = "cht-bsw-rt5672", - .pm = &snd_soc_pm_ops, }, .probe = snd_cht_mc_probe, }; diff --git a/sound/soc/intel/haswell.c b/sound/soc/intel/haswell.c index 35edf51..00fddd3 100644 --- a/sound/soc/intel/haswell.c +++ b/sound/soc/intel/haswell.c @@ -56,9 +56,7 @@ static int haswell_ssp0_fixup(struct snd_soc_pcm_runtime *rtd, channels->min = channels->max = 2; /* set SSP0 to 16 bit */ - snd_mask_set(¶ms->masks[SNDRV_PCM_HW_PARAM_FORMAT - - SNDRV_PCM_HW_PARAM_FIRST_MASK], - SNDRV_PCM_FORMAT_S16_LE); + params_set_format(params, SNDRV_PCM_FORMAT_S16_LE); return 0; } diff --git a/sound/soc/intel/mfld_machine.c b/sound/soc/intel/mfld_machine.c index 90b7a57..49c09a0 100644 --- a/sound/soc/intel/mfld_machine.c +++ b/sound/soc/intel/mfld_machine.c @@ -228,10 +228,13 @@ static void mfld_jack_check(unsigned int intr_status) { struct mfld_jack_data jack_data; + if (!mfld_codec) + return; + jack_data.mfld_jack = &mfld_jack; jack_data.intr_id = intr_status; - sn95031_jack_detection(&jack_data); + sn95031_jack_detection(mfld_codec, &jack_data); /* TODO: add american headset detection post gpiolib support */ } @@ -240,8 +243,6 @@ static int mfld_init(struct snd_soc_pcm_runtime *runtime) struct snd_soc_dapm_context *dapm = &runtime->card->dapm; int ret_val; - mfld_codec = runtime->codec; - /* default is earpiece pin, userspace sets it explcitly */ snd_soc_dapm_disable_pin(dapm, "Headphones"); /* default is lineout NC, userspace sets it explcitly */ @@ -254,20 +255,15 @@ static int mfld_init(struct snd_soc_pcm_runtime *runtime) snd_soc_dapm_disable_pin(dapm, "LINEINR"); /* Headset and button jack detection */ - ret_val = snd_soc_jack_new(mfld_codec, "Intel(R) MID Audio Jack", - SND_JACK_HEADSET | SND_JACK_BTN_0 | - SND_JACK_BTN_1, &mfld_jack); + ret_val = snd_soc_card_jack_new(runtime->card, + "Intel(R) MID Audio Jack", SND_JACK_HEADSET | + SND_JACK_BTN_0 | SND_JACK_BTN_1, &mfld_jack, + mfld_jack_pins, ARRAY_SIZE(mfld_jack_pins)); if (ret_val) { pr_err("jack creation failed\n"); return ret_val; } - ret_val = snd_soc_jack_add_pins(&mfld_jack, - ARRAY_SIZE(mfld_jack_pins), mfld_jack_pins); - if (ret_val) { - pr_err("adding jack pins failed\n"); - return ret_val; - } ret_val = snd_soc_jack_add_zones(&mfld_jack, ARRAY_SIZE(mfld_zones), mfld_zones); if (ret_val) { @@ -275,6 +271,8 @@ static int mfld_init(struct snd_soc_pcm_runtime *runtime) return ret_val; } + mfld_codec = runtime->codec; + /* we want to check if anything is inserted at boot, * so send a fake event to codec and it will read adc * to find if anything is there or not */ @@ -359,8 +357,6 @@ static irqreturn_t snd_mfld_jack_detection(int irq, void *data) { struct mfld_mc_private *mc_drv_ctx = (struct mfld_mc_private *) data; - if (mfld_jack.codec == NULL) - return IRQ_HANDLED; mfld_jack_check(mc_drv_ctx->interrupt_status); return IRQ_HANDLED; diff --git a/sound/soc/intel/sst-mfld-platform-pcm.c b/sound/soc/intel/sst-mfld-platform-pcm.c index 7523cbe..2fbaf2c 100644 --- a/sound/soc/intel/sst-mfld-platform-pcm.c +++ b/sound/soc/intel/sst-mfld-platform-pcm.c @@ -594,11 +594,13 @@ static int sst_platform_pcm_trigger(struct snd_pcm_substream *substream, ret_val = stream->ops->stream_drop(sst->dev, str_id); break; case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + case SNDRV_PCM_TRIGGER_SUSPEND: dev_dbg(rtd->dev, "sst: in pause\n"); status = SST_PLATFORM_PAUSED; ret_val = stream->ops->stream_pause(sst->dev, str_id); break; case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + case SNDRV_PCM_TRIGGER_RESUME: dev_dbg(rtd->dev, "sst: in pause release\n"); status = SST_PLATFORM_RUNNING; ret_val = stream->ops->stream_pause_release(sst->dev, str_id); @@ -665,6 +667,9 @@ static int sst_pcm_new(struct snd_soc_pcm_runtime *rtd) static int sst_soc_probe(struct snd_soc_platform *platform) { + struct sst_data *drv = dev_get_drvdata(platform->dev); + + drv->soc_card = platform->component.card; return sst_dsp_init_v2_dpcm(platform); } @@ -727,9 +732,64 @@ static int sst_platform_remove(struct platform_device *pdev) return 0; } +#ifdef CONFIG_PM_SLEEP + +static int sst_soc_prepare(struct device *dev) +{ + struct sst_data *drv = dev_get_drvdata(dev); + int i; + + /* suspend all pcms first */ + snd_soc_suspend(drv->soc_card->dev); + snd_soc_poweroff(drv->soc_card->dev); + + /* set the SSPs to idle */ + for (i = 0; i < drv->soc_card->num_rtd; i++) { + struct snd_soc_dai *dai = drv->soc_card->rtd[i].cpu_dai; + + if (dai->active) { + send_ssp_cmd(dai, dai->name, 0); + sst_handle_vb_timer(dai, false); + } + } + + return 0; +} + +static void sst_soc_complete(struct device *dev) +{ + struct sst_data *drv = dev_get_drvdata(dev); + int i; + + /* restart SSPs */ + for (i = 0; i < drv->soc_card->num_rtd; i++) { + struct snd_soc_dai *dai = drv->soc_card->rtd[i].cpu_dai; + + if (dai->active) { + sst_handle_vb_timer(dai, true); + send_ssp_cmd(dai, dai->name, 1); + } + } + snd_soc_resume(drv->soc_card->dev); +} + +#else + +#define sst_soc_prepare NULL +#define sst_soc_complete NULL + +#endif + + +static const struct dev_pm_ops sst_platform_pm = { + .prepare = sst_soc_prepare, + .complete = sst_soc_complete, +}; + static struct platform_driver sst_platform_driver = { .driver = { .name = "sst-mfld-platform", + .pm = &sst_platform_pm, }, .probe = sst_platform_probe, .remove = sst_platform_remove, diff --git a/sound/soc/intel/sst-mfld-platform.h b/sound/soc/intel/sst-mfld-platform.h index 79c8d12..9094314 100644 --- a/sound/soc/intel/sst-mfld-platform.h +++ b/sound/soc/intel/sst-mfld-platform.h @@ -174,6 +174,7 @@ struct sst_data { struct sst_platform_data *pdata; struct snd_sst_bytes_v2 *byte_stream; struct mutex lock; + struct snd_soc_card *soc_card; }; int sst_register_dsp(struct sst_device *sst); int sst_unregister_dsp(struct sst_device *sst); diff --git a/sound/soc/intel/sst/sst.c b/sound/soc/intel/sst/sst.c index 11c5786..1a7eeec 100644 --- a/sound/soc/intel/sst/sst.c +++ b/sound/soc/intel/sst/sst.c @@ -423,23 +423,135 @@ static int intel_sst_runtime_suspend(struct device *dev) return ret; } -static int intel_sst_runtime_resume(struct device *dev) +static int intel_sst_suspend(struct device *dev) { - int ret = 0; struct intel_sst_drv *ctx = dev_get_drvdata(dev); + struct sst_fw_save *fw_save; + int i, ret = 0; - if (ctx->sst_state == SST_RESET) { - ret = sst_load_fw(ctx); - if (ret) { - dev_err(dev, "FW download fail %d\n", ret); - sst_set_fw_state_locked(ctx, SST_RESET); + /* check first if we are already in SW reset */ + if (ctx->sst_state == SST_RESET) + return 0; + + /* + * check if any stream is active and running + * they should already by suspend by soc_suspend + */ + for (i = 1; i <= ctx->info.max_streams; i++) { + struct stream_info *stream = &ctx->streams[i]; + + if (stream->status == STREAM_RUNNING) { + dev_err(dev, "stream %d is running, cant susupend, abort\n", i); + return -EBUSY; } } + synchronize_irq(ctx->irq_num); + flush_workqueue(ctx->post_msg_wq); + + /* Move the SST state to Reset */ + sst_set_fw_state_locked(ctx, SST_RESET); + + /* tell DSP we are suspending */ + if (ctx->ops->save_dsp_context(ctx)) + return -EBUSY; + + /* save the memories */ + fw_save = kzalloc(sizeof(*fw_save), GFP_KERNEL); + if (!fw_save) + return -ENOMEM; + fw_save->iram = kzalloc(ctx->iram_end - ctx->iram_base, GFP_KERNEL); + if (!fw_save->iram) { + ret = -ENOMEM; + goto iram; + } + fw_save->dram = kzalloc(ctx->dram_end - ctx->dram_base, GFP_KERNEL); + if (!fw_save->dram) { + ret = -ENOMEM; + goto dram; + } + fw_save->sram = kzalloc(SST_MAILBOX_SIZE, GFP_KERNEL); + if (!fw_save->sram) { + ret = -ENOMEM; + goto sram; + } + + fw_save->ddr = kzalloc(ctx->ddr_end - ctx->ddr_base, GFP_KERNEL); + if (!fw_save->ddr) { + ret = -ENOMEM; + goto ddr; + } + + memcpy32_fromio(fw_save->iram, ctx->iram, ctx->iram_end - ctx->iram_base); + memcpy32_fromio(fw_save->dram, ctx->dram, ctx->dram_end - ctx->dram_base); + memcpy32_fromio(fw_save->sram, ctx->mailbox, SST_MAILBOX_SIZE); + memcpy32_fromio(fw_save->ddr, ctx->ddr, ctx->ddr_end - ctx->ddr_base); + + ctx->fw_save = fw_save; + ctx->ops->reset(ctx); + return 0; +ddr: + kfree(fw_save->sram); +sram: + kfree(fw_save->dram); +dram: + kfree(fw_save->iram); +iram: + kfree(fw_save); + return ret; +} + +static int intel_sst_resume(struct device *dev) +{ + struct intel_sst_drv *ctx = dev_get_drvdata(dev); + struct sst_fw_save *fw_save = ctx->fw_save; + int ret = 0; + struct sst_block *block; + + if (!fw_save) + return 0; + + sst_set_fw_state_locked(ctx, SST_FW_LOADING); + + /* we have to restore the memory saved */ + ctx->ops->reset(ctx); + + ctx->fw_save = NULL; + + memcpy32_toio(ctx->iram, fw_save->iram, ctx->iram_end - ctx->iram_base); + memcpy32_toio(ctx->dram, fw_save->dram, ctx->dram_end - ctx->dram_base); + memcpy32_toio(ctx->mailbox, fw_save->sram, SST_MAILBOX_SIZE); + memcpy32_toio(ctx->ddr, fw_save->ddr, ctx->ddr_end - ctx->ddr_base); + + kfree(fw_save->sram); + kfree(fw_save->dram); + kfree(fw_save->iram); + kfree(fw_save->ddr); + kfree(fw_save); + + block = sst_create_block(ctx, 0, FW_DWNL_ID); + if (block == NULL) + return -ENOMEM; + + + /* start and wait for ack */ + ctx->ops->start(ctx); + ret = sst_wait_timeout(ctx, block); + if (ret) { + dev_err(ctx->dev, "fw download failed %d\n", ret); + /* FW download failed due to timeout */ + ret = -EBUSY; + + } else { + sst_set_fw_state_locked(ctx, SST_FW_RUNNING); + } + + sst_free_block(ctx, block); return ret; } const struct dev_pm_ops intel_sst_pm = { + .suspend = intel_sst_suspend, + .resume = intel_sst_resume, .runtime_suspend = intel_sst_runtime_suspend, - .runtime_resume = intel_sst_runtime_resume, }; EXPORT_SYMBOL_GPL(intel_sst_pm); diff --git a/sound/soc/intel/sst/sst.h b/sound/soc/intel/sst/sst.h index 562bc48..3f49386 100644 --- a/sound/soc/intel/sst/sst.h +++ b/sound/soc/intel/sst/sst.h @@ -337,6 +337,13 @@ struct sst_shim_regs64 { u64 csr2; }; +struct sst_fw_save { + void *iram; + void *dram; + void *sram; + void *ddr; +}; + /** * struct intel_sst_drv - driver ops * @@ -428,6 +435,8 @@ struct intel_sst_drv { * persistent till worker thread gets called */ char firmware_name[FW_NAME_SIZE]; + + struct sst_fw_save *fw_save; }; /* misc definitions */ @@ -544,4 +553,7 @@ int sst_alloc_drv_context(struct intel_sst_drv **ctx, int sst_context_init(struct intel_sst_drv *ctx); void sst_context_cleanup(struct intel_sst_drv *ctx); void sst_configure_runtime_pm(struct intel_sst_drv *ctx); +void memcpy32_toio(void __iomem *dst, const void *src, int count); +void memcpy32_fromio(void *dst, const void __iomem *src, int count); + #endif diff --git a/sound/soc/intel/sst/sst_drv_interface.c b/sound/soc/intel/sst/sst_drv_interface.c index 5f75ef3..f0e4b99b 100644 --- a/sound/soc/intel/sst/sst_drv_interface.c +++ b/sound/soc/intel/sst/sst_drv_interface.c @@ -138,12 +138,36 @@ int sst_get_stream(struct intel_sst_drv *ctx, static int sst_power_control(struct device *dev, bool state) { struct intel_sst_drv *ctx = dev_get_drvdata(dev); - - dev_dbg(ctx->dev, "state:%d", state); - if (state == true) - return pm_runtime_get_sync(dev); - else + int ret = 0; + int usage_count = 0; + +#ifdef CONFIG_PM + usage_count = atomic_read(&dev->power.usage_count); +#else + usage_count = 1; +#endif + + if (state == true) { + ret = pm_runtime_get_sync(dev); + + dev_dbg(ctx->dev, "Enable: pm usage count: %d\n", usage_count); + if (ret < 0) { + dev_err(ctx->dev, "Runtime get failed with err: %d\n", ret); + return ret; + } + if ((ctx->sst_state == SST_RESET) && (usage_count == 1)) { + ret = sst_load_fw(ctx); + if (ret) { + dev_err(dev, "FW download fail %d\n", ret); + sst_set_fw_state_locked(ctx, SST_RESET); + ret = sst_pm_runtime_put(ctx); + } + } + } else { + dev_dbg(ctx->dev, "Disable: pm usage count: %d\n", usage_count); return sst_pm_runtime_put(ctx); + } + return ret; } /* @@ -572,6 +596,35 @@ static int sst_stream_drop(struct device *dev, int str_id) return sst_drop_stream(ctx, str_id); } +static int sst_stream_pause(struct device *dev, int str_id) +{ + struct stream_info *str_info; + struct intel_sst_drv *ctx = dev_get_drvdata(dev); + + if (ctx->sst_state != SST_FW_RUNNING) + return 0; + + str_info = get_stream_info(ctx, str_id); + if (!str_info) + return -EINVAL; + + return sst_pause_stream(ctx, str_id); +} + +static int sst_stream_resume(struct device *dev, int str_id) +{ + struct stream_info *str_info; + struct intel_sst_drv *ctx = dev_get_drvdata(dev); + + if (ctx->sst_state != SST_FW_RUNNING) + return 0; + + str_info = get_stream_info(ctx, str_id); + if (!str_info) + return -EINVAL; + return sst_resume_stream(ctx, str_id); +} + static int sst_stream_init(struct device *dev, struct pcm_stream_info *str_info) { int str_id = 0; @@ -633,6 +686,8 @@ static struct sst_ops pcm_ops = { .stream_init = sst_stream_init, .stream_start = sst_stream_start, .stream_drop = sst_stream_drop, + .stream_pause = sst_stream_pause, + .stream_pause_release = sst_stream_resume, .stream_read_tstamp = sst_read_timestamp, .send_byte_stream = sst_send_byte_stream, .close = sst_close_pcm_stream, diff --git a/sound/soc/intel/sst/sst_loader.c b/sound/soc/intel/sst/sst_loader.c index 7888cd7..e88907a 100644 --- a/sound/soc/intel/sst/sst_loader.c +++ b/sound/soc/intel/sst/sst_loader.c @@ -39,7 +39,15 @@ #include "sst.h" #include "../sst-dsp.h" -static inline void memcpy32_toio(void __iomem *dst, const void *src, int count) +void memcpy32_toio(void __iomem *dst, const void *src, int count) +{ + /* __iowrite32_copy uses 32-bit count values so divide by 4 for + * right count in words + */ + __iowrite32_copy(dst, src, count/4); +} + +void memcpy32_fromio(void *dst, const void __iomem *src, int count) { /* __iowrite32_copy uses 32-bit count values so divide by 4 for * right count in words diff --git a/sound/soc/omap/Kconfig b/sound/soc/omap/Kconfig index a2cd348..e7c78b0 100644 --- a/sound/soc/omap/Kconfig +++ b/sound/soc/omap/Kconfig @@ -100,17 +100,19 @@ config SND_OMAP_SOC_OMAP_TWL4030 config SND_OMAP_SOC_OMAP_ABE_TWL6040 tristate "SoC Audio support for OMAP boards using ABE and twl6040 codec" - depends on TWL6040_CORE && SND_OMAP_SOC && (ARCH_OMAP4 || COMPILE_TEST) + depends on TWL6040_CORE && SND_OMAP_SOC && (ARCH_OMAP4 || SOC_OMAP5 || COMPILE_TEST) select SND_OMAP_SOC_DMIC select SND_OMAP_SOC_MCPDM select SND_SOC_TWL6040 select SND_SOC_DMIC + select COMMON_CLK_PALMAS if SOC_OMAP5 help Say Y if you want to add support for SoC audio on OMAP boards using ABE and twl6040 codec. This driver currently supports: - SDP4430/Blaze boards - PandaBoard (4430) - PandaBoardES (4460) + - omap5-uevm (5432) config SND_OMAP_SOC_OMAP3_PANDORA tristate "SoC Audio support for OMAP3 Pandora" diff --git a/sound/soc/omap/ams-delta.c b/sound/soc/omap/ams-delta.c index 7066130..16cc95f 100644 --- a/sound/soc/omap/ams-delta.c +++ b/sound/soc/omap/ams-delta.c @@ -479,8 +479,8 @@ static int ams_delta_cx20442_init(struct snd_soc_pcm_runtime *rtd) /* Add hook switch - can be used to control the codec from userspace * even if line discipline fails */ - ret = snd_soc_jack_new(rtd->codec, "hook_switch", - SND_JACK_HEADSET, &ams_delta_hook_switch); + ret = snd_soc_card_jack_new(card, "hook_switch", SND_JACK_HEADSET, + &ams_delta_hook_switch, NULL, 0); if (ret) dev_warn(card->dev, "Failed to allocate resources for hook switch, " diff --git a/sound/soc/omap/omap-abe-twl6040.c b/sound/soc/omap/omap-abe-twl6040.c index b9c65f1..0843a68 100644 --- a/sound/soc/omap/omap-abe-twl6040.c +++ b/sound/soc/omap/omap-abe-twl6040.c @@ -182,17 +182,17 @@ static int omap_abe_twl6040_init(struct snd_soc_pcm_runtime *rtd) /* Headset jack detection only if it is supported */ if (priv->jack_detection) { - ret = snd_soc_jack_new(codec, "Headset Jack", - SND_JACK_HEADSET, &hs_jack); + ret = snd_soc_card_jack_new(rtd->card, "Headset Jack", + SND_JACK_HEADSET, &hs_jack, + hs_jack_pins, + ARRAY_SIZE(hs_jack_pins)); if (ret) return ret; - ret = snd_soc_jack_add_pins(&hs_jack, ARRAY_SIZE(hs_jack_pins), - hs_jack_pins); twl6040_hs_jack_detect(codec, &hs_jack, SND_JACK_HEADSET); } - return ret; + return 0; } static const struct snd_soc_dapm_route dmic_audio_map[] = { diff --git a/sound/soc/omap/omap-pcm.c b/sound/soc/omap/omap-pcm.c index 1343ecb..6bb623a 100644 --- a/sound/soc/omap/omap-pcm.c +++ b/sound/soc/omap/omap-pcm.c @@ -39,7 +39,7 @@ #define pcm_omap1510() 0 #endif -static const struct snd_pcm_hardware omap_pcm_hardware = { +static struct snd_pcm_hardware omap_pcm_hardware = { .info = SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID | SNDRV_PCM_INFO_INTERLEAVED | @@ -53,6 +53,24 @@ static const struct snd_pcm_hardware omap_pcm_hardware = { .buffer_bytes_max = 128 * 1024, }; +/* sDMA supports only 1, 2, and 4 byte transfer elements. */ +static void omap_pcm_limit_supported_formats(void) +{ + int i; + + for (i = 0; i < SNDRV_PCM_FORMAT_LAST; i++) { + switch (snd_pcm_format_physical_width(i)) { + case 8: + case 16: + case 32: + omap_pcm_hardware.formats |= (1LL << i); + break; + default: + break; + } + } +} + /* this may get called several times by oss emulation */ static int omap_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) @@ -235,6 +253,7 @@ static struct snd_soc_platform_driver omap_soc_platform = { int omap_pcm_platform_register(struct device *dev) { + omap_pcm_limit_supported_formats(); return devm_snd_soc_register_platform(dev, &omap_soc_platform); } EXPORT_SYMBOL_GPL(omap_pcm_platform_register); diff --git a/sound/soc/omap/omap-twl4030.c b/sound/soc/omap/omap-twl4030.c index fb1f6bb..3673ada 100644 --- a/sound/soc/omap/omap-twl4030.c +++ b/sound/soc/omap/omap-twl4030.c @@ -170,14 +170,10 @@ static int omap_twl4030_init(struct snd_soc_pcm_runtime *rtd) if (priv->jack_detect > 0) { hs_jack_gpios[0].gpio = priv->jack_detect; - ret = snd_soc_jack_new(codec, "Headset Jack", SND_JACK_HEADSET, - &priv->hs_jack); - if (ret) - return ret; - - ret = snd_soc_jack_add_pins(&priv->hs_jack, - ARRAY_SIZE(hs_jack_pins), - hs_jack_pins); + ret = snd_soc_card_jack_new(rtd->card, "Headset Jack", + SND_JACK_HEADSET, &priv->hs_jack, + hs_jack_pins, + ARRAY_SIZE(hs_jack_pins)); if (ret) return ret; diff --git a/sound/soc/omap/rx51.c b/sound/soc/omap/rx51.c index 7f29935..c2ddf0f 100644 --- a/sound/soc/omap/rx51.c +++ b/sound/soc/omap/rx51.c @@ -311,9 +311,9 @@ static int rx51_aic34_init(struct snd_soc_pcm_runtime *rtd) } /* AV jack detection */ - err = snd_soc_jack_new(codec, "AV Jack", - SND_JACK_HEADSET | SND_JACK_VIDEOOUT, - &rx51_av_jack); + err = snd_soc_card_jack_new(rtd->card, "AV Jack", + SND_JACK_HEADSET | SND_JACK_VIDEOOUT, + &rx51_av_jack, NULL, 0); if (err) { dev_err(card->dev, "Failed to add AV Jack\n"); return err; diff --git a/sound/soc/pxa/hx4700.c b/sound/soc/pxa/hx4700.c index 73eb5dd..9f8be7c 100644 --- a/sound/soc/pxa/hx4700.c +++ b/sound/soc/pxa/hx4700.c @@ -126,17 +126,12 @@ static const struct snd_soc_dapm_route hx4700_audio_map[] = { */ static int hx4700_ak4641_init(struct snd_soc_pcm_runtime *rtd) { - struct snd_soc_codec *codec = rtd->codec; int err; /* Jack detection API stuff */ - err = snd_soc_jack_new(codec, "Headphone Jack", - SND_JACK_HEADPHONE, &hs_jack); - if (err) - return err; - - err = snd_soc_jack_add_pins(&hs_jack, ARRAY_SIZE(hs_jack_pin), - hs_jack_pin); + err = snd_soc_card_jack_new(rtd->card, "Headphone Jack", + SND_JACK_HEADPHONE, &hs_jack, hs_jack_pin, + ARRAY_SIZE(hs_jack_pin)); if (err) return err; diff --git a/sound/soc/pxa/palm27x.c b/sound/soc/pxa/palm27x.c index 910336c..c20bbc0 100644 --- a/sound/soc/pxa/palm27x.c +++ b/sound/soc/pxa/palm27x.c @@ -75,17 +75,12 @@ static struct snd_soc_card palm27x_asoc; static int palm27x_ac97_init(struct snd_soc_pcm_runtime *rtd) { - struct snd_soc_codec *codec = rtd->codec; int err; /* Jack detection API stuff */ - err = snd_soc_jack_new(codec, "Headphone Jack", - SND_JACK_HEADPHONE, &hs_jack); - if (err) - return err; - - err = snd_soc_jack_add_pins(&hs_jack, ARRAY_SIZE(hs_jack_pins), - hs_jack_pins); + err = snd_soc_card_jack_new(rtd->card, "Headphone Jack", + SND_JACK_HEADPHONE, &hs_jack, hs_jack_pins, + ARRAY_SIZE(hs_jack_pins)); if (err) return err; diff --git a/sound/soc/pxa/ttc-dkb.c b/sound/soc/pxa/ttc-dkb.c index 5001dbb..1753c7d 100644 --- a/sound/soc/pxa/ttc-dkb.c +++ b/sound/soc/pxa/ttc-dkb.c @@ -78,15 +78,12 @@ static int ttc_pm860x_init(struct snd_soc_pcm_runtime *rtd) struct snd_soc_codec *codec = rtd->codec; /* Headset jack detection */ - snd_soc_jack_new(codec, "Headphone Jack", SND_JACK_HEADPHONE - | SND_JACK_BTN_0 | SND_JACK_BTN_1 | SND_JACK_BTN_2, - &hs_jack); - snd_soc_jack_add_pins(&hs_jack, ARRAY_SIZE(hs_jack_pins), - hs_jack_pins); - snd_soc_jack_new(codec, "Microphone Jack", SND_JACK_MICROPHONE, - &mic_jack); - snd_soc_jack_add_pins(&mic_jack, ARRAY_SIZE(mic_jack_pins), - mic_jack_pins); + snd_soc_card_jack_new(rtd->card, "Headphone Jack", SND_JACK_HEADPHONE | + SND_JACK_BTN_0 | SND_JACK_BTN_1 | SND_JACK_BTN_2, + &hs_jack, hs_jack_pins, ARRAY_SIZE(hs_jack_pins)); + snd_soc_card_jack_new(rtd->card, "Microphone Jack", SND_JACK_MICROPHONE, + &mic_jack, mic_jack_pins, + ARRAY_SIZE(mic_jack_pins)); /* headphone, microphone detection & headset short detection */ pm860x_hs_jack_detect(codec, &hs_jack, SND_JACK_HEADPHONE, diff --git a/sound/soc/pxa/z2.c b/sound/soc/pxa/z2.c index 76ccb172..bcbfbe8 100644 --- a/sound/soc/pxa/z2.c +++ b/sound/soc/pxa/z2.c @@ -143,13 +143,9 @@ static int z2_wm8750_init(struct snd_soc_pcm_runtime *rtd) snd_soc_dapm_disable_pin(dapm, "MONO1"); /* Jack detection API stuff */ - ret = snd_soc_jack_new(codec, "Headset Jack", SND_JACK_HEADSET, - &hs_jack); - if (ret) - goto err; - - ret = snd_soc_jack_add_pins(&hs_jack, ARRAY_SIZE(hs_jack_pins), - hs_jack_pins); + ret = snd_soc_card_jack_new(rtd->card, "Headset Jack", SND_JACK_HEADSET, + &hs_jack, hs_jack_pins, + ARRAY_SIZE(hs_jack_pins)); if (ret) goto err; diff --git a/sound/soc/samsung/h1940_uda1380.c b/sound/soc/samsung/h1940_uda1380.c index 59b0442..c72e9fb 100644 --- a/sound/soc/samsung/h1940_uda1380.c +++ b/sound/soc/samsung/h1940_uda1380.c @@ -162,13 +162,8 @@ static struct platform_device *s3c24xx_snd_device; static int h1940_uda1380_init(struct snd_soc_pcm_runtime *rtd) { - struct snd_soc_codec *codec = rtd->codec; - - snd_soc_jack_new(codec, "Headphone Jack", SND_JACK_HEADPHONE, - &hp_jack); - - snd_soc_jack_add_pins(&hp_jack, ARRAY_SIZE(hp_jack_pins), - hp_jack_pins); + snd_soc_card_jack_new(rtd->card, "Headphone Jack", SND_JACK_HEADPHONE, + &hp_jack, hp_jack_pins, ARRAY_SIZE(hp_jack_pins)); snd_soc_jack_add_gpios(&hp_jack, ARRAY_SIZE(hp_jack_gpios), hp_jack_gpios); diff --git a/sound/soc/samsung/littlemill.c b/sound/soc/samsung/littlemill.c index 141519c..31a820e 100644 --- a/sound/soc/samsung/littlemill.c +++ b/sound/soc/samsung/littlemill.c @@ -260,12 +260,12 @@ static int littlemill_late_probe(struct snd_soc_card *card) if (ret < 0) return ret; - ret = snd_soc_jack_new(codec, "Headset", - SND_JACK_HEADSET | SND_JACK_MECHANICAL | - SND_JACK_BTN_0 | SND_JACK_BTN_1 | - SND_JACK_BTN_2 | SND_JACK_BTN_3 | - SND_JACK_BTN_4 | SND_JACK_BTN_5, - &littlemill_headset); + ret = snd_soc_card_jack_new(card, "Headset", + SND_JACK_HEADSET | SND_JACK_MECHANICAL | + SND_JACK_BTN_0 | SND_JACK_BTN_1 | + SND_JACK_BTN_2 | SND_JACK_BTN_3 | + SND_JACK_BTN_4 | SND_JACK_BTN_5, + &littlemill_headset, NULL, 0); if (ret) return ret; diff --git a/sound/soc/samsung/lowland.c b/sound/soc/samsung/lowland.c index 243dea7..5f15609 100644 --- a/sound/soc/samsung/lowland.c +++ b/sound/soc/samsung/lowland.c @@ -56,16 +56,10 @@ static int lowland_wm5100_init(struct snd_soc_pcm_runtime *rtd) return ret; } - ret = snd_soc_jack_new(codec, "Headset", - SND_JACK_LINEOUT | SND_JACK_HEADSET | - SND_JACK_BTN_0, - &lowland_headset); - if (ret) - return ret; - - ret = snd_soc_jack_add_pins(&lowland_headset, - ARRAY_SIZE(lowland_headset_pins), - lowland_headset_pins); + ret = snd_soc_card_jack_new(rtd->card, "Headset", SND_JACK_LINEOUT | + SND_JACK_HEADSET | SND_JACK_BTN_0, + &lowland_headset, lowland_headset_pins, + ARRAY_SIZE(lowland_headset_pins)); if (ret) return ret; diff --git a/sound/soc/samsung/rx1950_uda1380.c b/sound/soc/samsung/rx1950_uda1380.c index 873f2cb..35e37c4 100644 --- a/sound/soc/samsung/rx1950_uda1380.c +++ b/sound/soc/samsung/rx1950_uda1380.c @@ -211,13 +211,8 @@ static int rx1950_hw_params(struct snd_pcm_substream *substream, static int rx1950_uda1380_init(struct snd_soc_pcm_runtime *rtd) { - struct snd_soc_codec *codec = rtd->codec; - - snd_soc_jack_new(codec, "Headphone Jack", SND_JACK_HEADPHONE, - &hp_jack); - - snd_soc_jack_add_pins(&hp_jack, ARRAY_SIZE(hp_jack_pins), - hp_jack_pins); + snd_soc_card_jack_new(rtd->card, "Headphone Jack", SND_JACK_HEADPHONE, + &hp_jack, hp_jack_pins, ARRAY_SIZE(hp_jack_pins)); snd_soc_jack_add_gpios(&hp_jack, ARRAY_SIZE(hp_jack_gpios), hp_jack_gpios); diff --git a/sound/soc/samsung/smartq_wm8987.c b/sound/soc/samsung/smartq_wm8987.c index 8291d2a..dfbe2db 100644 --- a/sound/soc/samsung/smartq_wm8987.c +++ b/sound/soc/samsung/smartq_wm8987.c @@ -151,13 +151,10 @@ static int smartq_wm8987_init(struct snd_soc_pcm_runtime *rtd) snd_soc_dapm_disable_pin(dapm, "Headphone Jack"); /* Headphone jack detection */ - err = snd_soc_jack_new(codec, "Headphone Jack", - SND_JACK_HEADPHONE, &smartq_jack); - if (err) - return err; - - err = snd_soc_jack_add_pins(&smartq_jack, ARRAY_SIZE(smartq_jack_pins), - smartq_jack_pins); + err = snd_soc_card_jack_new(rtd->card, "Headphone Jack", + SND_JACK_HEADPHONE, &smartq_jack, + smartq_jack_pins, + ARRAY_SIZE(smartq_jack_pins)); if (err) return err; diff --git a/sound/soc/samsung/speyside.c b/sound/soc/samsung/speyside.c index 5ec7c52..2dcb988 100644 --- a/sound/soc/samsung/speyside.c +++ b/sound/soc/samsung/speyside.c @@ -153,16 +153,10 @@ static int speyside_wm8996_init(struct snd_soc_pcm_runtime *rtd) pr_err("Failed to request HP_SEL GPIO: %d\n", ret); gpio_direction_output(WM8996_HPSEL_GPIO, speyside_jack_polarity); - ret = snd_soc_jack_new(codec, "Headset", - SND_JACK_LINEOUT | SND_JACK_HEADSET | - SND_JACK_BTN_0, - &speyside_headset); - if (ret) - return ret; - - ret = snd_soc_jack_add_pins(&speyside_headset, - ARRAY_SIZE(speyside_headset_pins), - speyside_headset_pins); + ret = snd_soc_card_jack_new(rtd->card, "Headset", SND_JACK_LINEOUT | + SND_JACK_HEADSET | SND_JACK_BTN_0, + &speyside_headset, speyside_headset_pins, + ARRAY_SIZE(speyside_headset_pins)); if (ret) return ret; diff --git a/sound/soc/samsung/tobermory.c b/sound/soc/samsung/tobermory.c index 9c80506..85ccfb7 100644 --- a/sound/soc/samsung/tobermory.c +++ b/sound/soc/samsung/tobermory.c @@ -179,15 +179,10 @@ static int tobermory_late_probe(struct snd_soc_card *card) if (ret < 0) return ret; - ret = snd_soc_jack_new(codec, "Headset", - SND_JACK_HEADSET | SND_JACK_BTN_0, - &tobermory_headset); - if (ret) - return ret; - - ret = snd_soc_jack_add_pins(&tobermory_headset, - ARRAY_SIZE(tobermory_headset_pins), - tobermory_headset_pins); + ret = snd_soc_card_jack_new(card, "Headset", SND_JACK_HEADSET | + SND_JACK_BTN_0, &tobermory_headset, + tobermory_headset_pins, + ARRAY_SIZE(tobermory_headset_pins)); if (ret) return ret; diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index e5c9908..07aa543 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1578,6 +1578,10 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card) snd_soc_dapm_new_controls(&card->dapm, card->dapm_widgets, card->num_dapm_widgets); + if (card->of_dapm_widgets) + snd_soc_dapm_new_controls(&card->dapm, card->of_dapm_widgets, + card->num_of_dapm_widgets); + /* initialise the sound card only once */ if (card->probe) { ret = card->probe(card); @@ -1633,6 +1637,10 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card) snd_soc_dapm_add_routes(&card->dapm, card->dapm_routes, card->num_dapm_routes); + if (card->of_dapm_routes) + snd_soc_dapm_add_routes(&card->dapm, card->of_dapm_routes, + card->num_of_dapm_routes); + for (i = 0; i < card->num_links; i++) { if (card->dai_link[i].dai_fmt) snd_soc_runtime_set_dai_fmt(&card->rtd[i], @@ -3242,8 +3250,8 @@ int snd_soc_of_parse_audio_simple_widgets(struct snd_soc_card *card, widgets[i].name = wname; } - card->dapm_widgets = widgets; - card->num_dapm_widgets = num_widgets; + card->of_dapm_widgets = widgets; + card->num_of_dapm_widgets = num_widgets; return 0; } @@ -3327,8 +3335,8 @@ int snd_soc_of_parse_audio_routing(struct snd_soc_card *card, } } - card->num_dapm_routes = num_routes; - card->dapm_routes = routes; + card->num_of_dapm_routes = num_routes; + card->of_dapm_routes = routes; return 0; } diff --git a/sound/soc/soc-jack.c b/sound/soc/soc-jack.c index 4380dcc..9f60c25 100644 --- a/sound/soc/soc-jack.c +++ b/sound/soc/soc-jack.c @@ -22,30 +22,42 @@ #include <trace/events/asoc.h> /** - * snd_soc_jack_new - Create a new jack - * @codec: ASoC codec + * snd_soc_card_jack_new - Create a new jack + * @card: ASoC card * @id: an identifying string for this jack * @type: a bitmask of enum snd_jack_type values that can be detected by * this jack * @jack: structure to use for the jack + * @pins: Array of jack pins to be added to the jack or NULL + * @num_pins: Number of elements in the @pins array * * Creates a new jack object. * * Returns zero if successful, or a negative error code on failure. * On success jack will be initialised. */ -int snd_soc_jack_new(struct snd_soc_codec *codec, const char *id, int type, - struct snd_soc_jack *jack) +int snd_soc_card_jack_new(struct snd_soc_card *card, const char *id, int type, + struct snd_soc_jack *jack, struct snd_soc_jack_pin *pins, + unsigned int num_pins) { + int ret; + mutex_init(&jack->mutex); - jack->codec = codec; + jack->card = card; INIT_LIST_HEAD(&jack->pins); INIT_LIST_HEAD(&jack->jack_zones); BLOCKING_INIT_NOTIFIER_HEAD(&jack->notifier); - return snd_jack_new(codec->component.card->snd_card, id, type, &jack->jack); + ret = snd_jack_new(card->snd_card, id, type, &jack->jack); + if (ret) + return ret; + + if (num_pins) + return snd_soc_jack_add_pins(jack, num_pins, pins); + + return 0; } -EXPORT_SYMBOL_GPL(snd_soc_jack_new); +EXPORT_SYMBOL_GPL(snd_soc_card_jack_new); /** * snd_soc_jack_report - Report the current status for a jack @@ -63,7 +75,6 @@ EXPORT_SYMBOL_GPL(snd_soc_jack_new); */ void snd_soc_jack_report(struct snd_soc_jack *jack, int status, int mask) { - struct snd_soc_codec *codec; struct snd_soc_dapm_context *dapm; struct snd_soc_jack_pin *pin; unsigned int sync = 0; @@ -74,8 +85,7 @@ void snd_soc_jack_report(struct snd_soc_jack *jack, int status, int mask) if (!jack) return; - codec = jack->codec; - dapm = &codec->dapm; + dapm = &jack->card->dapm; mutex_lock(&jack->mutex); @@ -175,12 +185,12 @@ int snd_soc_jack_add_pins(struct snd_soc_jack *jack, int count, for (i = 0; i < count; i++) { if (!pins[i].pin) { - dev_err(jack->codec->dev, "ASoC: No name for pin %d\n", + dev_err(jack->card->dev, "ASoC: No name for pin %d\n", i); return -EINVAL; } if (!pins[i].mask) { - dev_err(jack->codec->dev, "ASoC: No mask for pin %d" + dev_err(jack->card->dev, "ASoC: No mask for pin %d" " (%s)\n", i, pins[i].pin); return -EINVAL; } @@ -260,7 +270,7 @@ static void snd_soc_jack_gpio_detect(struct snd_soc_jack_gpio *gpio) static irqreturn_t gpio_handler(int irq, void *data) { struct snd_soc_jack_gpio *gpio = data; - struct device *dev = gpio->jack->codec->component.card->dev; + struct device *dev = gpio->jack->card->dev; trace_snd_soc_jack_irq(gpio->name); @@ -299,7 +309,7 @@ int snd_soc_jack_add_gpios(struct snd_soc_jack *jack, int count, for (i = 0; i < count; i++) { if (!gpios[i].name) { - dev_err(jack->codec->dev, + dev_err(jack->card->dev, "ASoC: No name for gpio at index %d\n", i); ret = -EINVAL; goto undo; @@ -320,7 +330,7 @@ int snd_soc_jack_add_gpios(struct snd_soc_jack *jack, int count, } else { /* legacy GPIO number */ if (!gpio_is_valid(gpios[i].gpio)) { - dev_err(jack->codec->dev, + dev_err(jack->card->dev, "ASoC: Invalid gpio %d\n", gpios[i].gpio); ret = -EINVAL; @@ -350,7 +360,7 @@ int snd_soc_jack_add_gpios(struct snd_soc_jack *jack, int count, if (gpios[i].wake) { ret = irq_set_irq_wake(gpiod_to_irq(gpios[i].desc), 1); if (ret != 0) - dev_err(jack->codec->dev, + dev_err(jack->card->dev, "ASoC: Failed to mark GPIO at index %d as wake source: %d\n", i, ret); } diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 6b0136e..6e3781e 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -2511,6 +2511,7 @@ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num) /* DAPM dai link stream work */ INIT_DELAYED_WORK(&rtd->delayed_work, close_delayed_work); + pcm->nonatomic = rtd->dai_link->nonatomic; rtd->pcm = pcm; pcm->private_data = rtd; diff --git a/sound/soc/tegra/tegra_alc5632.c b/sound/soc/tegra/tegra_alc5632.c index 769aca2..6dcd06a 100644 --- a/sound/soc/tegra/tegra_alc5632.c +++ b/sound/soc/tegra/tegra_alc5632.c @@ -106,11 +106,10 @@ static int tegra_alc5632_asoc_init(struct snd_soc_pcm_runtime *rtd) struct snd_soc_dapm_context *dapm = &codec->dapm; struct tegra_alc5632 *machine = snd_soc_card_get_drvdata(rtd->card); - snd_soc_jack_new(codec, "Headset Jack", SND_JACK_HEADSET, - &tegra_alc5632_hs_jack); - snd_soc_jack_add_pins(&tegra_alc5632_hs_jack, - ARRAY_SIZE(tegra_alc5632_hs_jack_pins), - tegra_alc5632_hs_jack_pins); + snd_soc_card_jack_new(rtd->card, "Headset Jack", SND_JACK_HEADSET, + &tegra_alc5632_hs_jack, + tegra_alc5632_hs_jack_pins, + ARRAY_SIZE(tegra_alc5632_hs_jack_pins)); if (gpio_is_valid(machine->gpio_hp_det)) { tegra_alc5632_hp_jack_gpio.gpio = machine->gpio_hp_det; diff --git a/sound/soc/tegra/tegra_max98090.c b/sound/soc/tegra/tegra_max98090.c index af3fb99..902da36 100644 --- a/sound/soc/tegra/tegra_max98090.c +++ b/sound/soc/tegra/tegra_max98090.c @@ -133,24 +133,26 @@ static const struct snd_soc_dapm_widget tegra_max98090_dapm_widgets[] = { SND_SOC_DAPM_HP("Headphones", NULL), SND_SOC_DAPM_SPK("Speakers", NULL), SND_SOC_DAPM_MIC("Mic Jack", NULL), + SND_SOC_DAPM_MIC("Int Mic", NULL), }; static const struct snd_kcontrol_new tegra_max98090_controls[] = { + SOC_DAPM_PIN_SWITCH("Headphones"), SOC_DAPM_PIN_SWITCH("Speakers"), + SOC_DAPM_PIN_SWITCH("Mic Jack"), + SOC_DAPM_PIN_SWITCH("Int Mic"), }; static int tegra_max98090_asoc_init(struct snd_soc_pcm_runtime *rtd) { - struct snd_soc_dai *codec_dai = rtd->codec_dai; - struct snd_soc_codec *codec = codec_dai->codec; struct tegra_max98090 *machine = snd_soc_card_get_drvdata(rtd->card); if (gpio_is_valid(machine->gpio_hp_det)) { - snd_soc_jack_new(codec, "Headphones", SND_JACK_HEADPHONE, - &tegra_max98090_hp_jack); - snd_soc_jack_add_pins(&tegra_max98090_hp_jack, - ARRAY_SIZE(tegra_max98090_hp_jack_pins), - tegra_max98090_hp_jack_pins); + snd_soc_card_jack_new(rtd->card, "Headphones", + SND_JACK_HEADPHONE, + &tegra_max98090_hp_jack, + tegra_max98090_hp_jack_pins, + ARRAY_SIZE(tegra_max98090_hp_jack_pins)); tegra_max98090_hp_jack_gpio.gpio = machine->gpio_hp_det; snd_soc_jack_add_gpios(&tegra_max98090_hp_jack, @@ -159,11 +161,11 @@ static int tegra_max98090_asoc_init(struct snd_soc_pcm_runtime *rtd) } if (gpio_is_valid(machine->gpio_mic_det)) { - snd_soc_jack_new(codec, "Mic Jack", SND_JACK_MICROPHONE, - &tegra_max98090_mic_jack); - snd_soc_jack_add_pins(&tegra_max98090_mic_jack, - ARRAY_SIZE(tegra_max98090_mic_jack_pins), - tegra_max98090_mic_jack_pins); + snd_soc_card_jack_new(rtd->card, "Mic Jack", + SND_JACK_MICROPHONE, + &tegra_max98090_mic_jack, + tegra_max98090_mic_jack_pins, + ARRAY_SIZE(tegra_max98090_mic_jack_pins)); tegra_max98090_mic_jack_gpio.gpio = machine->gpio_mic_det; snd_soc_jack_add_gpios(&tegra_max98090_mic_jack, diff --git a/sound/soc/tegra/tegra_rt5640.c b/sound/soc/tegra/tegra_rt5640.c index ed759a3..773daec 100644 --- a/sound/soc/tegra/tegra_rt5640.c +++ b/sound/soc/tegra/tegra_rt5640.c @@ -108,15 +108,11 @@ static const struct snd_kcontrol_new tegra_rt5640_controls[] = { static int tegra_rt5640_asoc_init(struct snd_soc_pcm_runtime *rtd) { - struct snd_soc_dai *codec_dai = rtd->codec_dai; - struct snd_soc_codec *codec = codec_dai->codec; struct tegra_rt5640 *machine = snd_soc_card_get_drvdata(rtd->card); - snd_soc_jack_new(codec, "Headphones", SND_JACK_HEADPHONE, - &tegra_rt5640_hp_jack); - snd_soc_jack_add_pins(&tegra_rt5640_hp_jack, - ARRAY_SIZE(tegra_rt5640_hp_jack_pins), - tegra_rt5640_hp_jack_pins); + snd_soc_card_jack_new(rtd->card, "Headphones", SND_JACK_HEADPHONE, + &tegra_rt5640_hp_jack, tegra_rt5640_hp_jack_pins, + ARRAY_SIZE(tegra_rt5640_hp_jack_pins)); if (gpio_is_valid(machine->gpio_hp_det)) { tegra_rt5640_hp_jack_gpio.gpio = machine->gpio_hp_det; diff --git a/sound/soc/tegra/tegra_rt5677.c b/sound/soc/tegra/tegra_rt5677.c index e4cf978..68d8b67 100644 --- a/sound/soc/tegra/tegra_rt5677.c +++ b/sound/soc/tegra/tegra_rt5677.c @@ -146,10 +146,9 @@ static int tegra_rt5677_asoc_init(struct snd_soc_pcm_runtime *rtd) struct snd_soc_dapm_context *dapm = &codec->dapm; struct tegra_rt5677 *machine = snd_soc_card_get_drvdata(rtd->card); - snd_soc_jack_new(codec, "Headphone Jack", SND_JACK_HEADPHONE, - &tegra_rt5677_hp_jack); - snd_soc_jack_add_pins(&tegra_rt5677_hp_jack, 1, - &tegra_rt5677_hp_jack_pins); + snd_soc_card_jack_new(rtd->card, "Headphone Jack", SND_JACK_HEADPHONE, + &tegra_rt5677_hp_jack, + &tegra_rt5677_hp_jack_pins, 1); if (gpio_is_valid(machine->gpio_hp_det)) { tegra_rt5677_hp_jack_gpio.gpio = machine->gpio_hp_det; @@ -158,10 +157,9 @@ static int tegra_rt5677_asoc_init(struct snd_soc_pcm_runtime *rtd) } - snd_soc_jack_new(codec, "Mic Jack", SND_JACK_MICROPHONE, - &tegra_rt5677_mic_jack); - snd_soc_jack_add_pins(&tegra_rt5677_mic_jack, 1, - &tegra_rt5677_mic_jack_pins); + snd_soc_card_jack_new(rtd->card, "Mic Jack", SND_JACK_MICROPHONE, + &tegra_rt5677_mic_jack, + &tegra_rt5677_mic_jack_pins, 1); if (gpio_is_valid(machine->gpio_mic_present)) { tegra_rt5677_mic_jack_gpio.gpio = machine->gpio_mic_present; diff --git a/sound/soc/tegra/tegra_wm8903.c b/sound/soc/tegra/tegra_wm8903.c index e52420d..4a95b70 100644 --- a/sound/soc/tegra/tegra_wm8903.c +++ b/sound/soc/tegra/tegra_wm8903.c @@ -177,21 +177,19 @@ static int tegra_wm8903_init(struct snd_soc_pcm_runtime *rtd) if (gpio_is_valid(machine->gpio_hp_det)) { tegra_wm8903_hp_jack_gpio.gpio = machine->gpio_hp_det; - snd_soc_jack_new(codec, "Headphone Jack", SND_JACK_HEADPHONE, - &tegra_wm8903_hp_jack); - snd_soc_jack_add_pins(&tegra_wm8903_hp_jack, - ARRAY_SIZE(tegra_wm8903_hp_jack_pins), - tegra_wm8903_hp_jack_pins); + snd_soc_card_jack_new(rtd->card, "Headphone Jack", + SND_JACK_HEADPHONE, &tegra_wm8903_hp_jack, + tegra_wm8903_hp_jack_pins, + ARRAY_SIZE(tegra_wm8903_hp_jack_pins)); snd_soc_jack_add_gpios(&tegra_wm8903_hp_jack, 1, &tegra_wm8903_hp_jack_gpio); } - snd_soc_jack_new(codec, "Mic Jack", SND_JACK_MICROPHONE, - &tegra_wm8903_mic_jack); - snd_soc_jack_add_pins(&tegra_wm8903_mic_jack, - ARRAY_SIZE(tegra_wm8903_mic_jack_pins), - tegra_wm8903_mic_jack_pins); + snd_soc_card_jack_new(rtd->card, "Mic Jack", SND_JACK_MICROPHONE, + &tegra_wm8903_mic_jack, + tegra_wm8903_mic_jack_pins, + ARRAY_SIZE(tegra_wm8903_mic_jack_pins)); wm8903_mic_detect(codec, &tegra_wm8903_mic_jack, SND_JACK_MICROPHONE, 0); diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 753a47d..353532b 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1120,17 +1120,24 @@ bool snd_usb_get_sample_rate_quirk(struct snd_usb_audio *chip) /* Marantz/Denon USB DACs need a vendor cmd to switch * between PCM and native DSD mode */ +static bool is_marantz_denon_dac(unsigned int id) +{ + switch (id) { + case USB_ID(0x154e, 0x1003): /* Denon DA-300USB */ + case USB_ID(0x154e, 0x3005): /* Marantz HD-DAC1 */ + case USB_ID(0x154e, 0x3006): /* Marantz SA-14S1 */ + return true; + } + return false; +} + int snd_usb_select_mode_quirk(struct snd_usb_substream *subs, struct audioformat *fmt) { struct usb_device *dev = subs->dev; int err; - switch (subs->stream->chip->usb_id) { - case USB_ID(0x154e, 0x1003): /* Denon DA-300USB */ - case USB_ID(0x154e, 0x3005): /* Marantz HD-DAC1 */ - case USB_ID(0x154e, 0x3006): /* Marantz SA-14S1 */ - + if (is_marantz_denon_dac(subs->stream->chip->usb_id)) { /* First switch to alt set 0, otherwise the mode switch cmd * will not be accepted by the DAC */ @@ -1203,17 +1210,10 @@ void snd_usb_ctl_msg_quirk(struct usb_device *dev, unsigned int pipe, /* Marantz/Denon devices with USB DAC functionality need a delay * after each class compliant request */ - if ((le16_to_cpu(dev->descriptor.idVendor) == 0x154e) && - (requesttype & USB_TYPE_MASK) == USB_TYPE_CLASS) { - - switch (le16_to_cpu(dev->descriptor.idProduct)) { - case 0x1003: /* Denon DA300-USB */ - case 0x3005: /* Marantz HD-DAC1 */ - case 0x3006: /* Marantz SA-14S1 */ - mdelay(20); - break; - } - } + if (is_marantz_denon_dac(USB_ID(le16_to_cpu(dev->descriptor.idVendor), + le16_to_cpu(dev->descriptor.idProduct))) + && (requesttype & USB_TYPE_MASK) == USB_TYPE_CLASS) + mdelay(20); /* Zoom R16/24 needs a tiny delay here, otherwise requests like * get/set frequency return as failed despite actually succeeding. @@ -1268,15 +1268,9 @@ u64 snd_usb_interface_dsd_format_quirks(struct snd_usb_audio *chip, } /* Denon/Marantz devices with USB DAC functionality */ - switch (chip->usb_id) { - case USB_ID(0x154e, 0x1003): /* Denon DA300-USB */ - case USB_ID(0x154e, 0x3005): /* Marantz HD-DAC1 */ - case USB_ID(0x154e, 0x3006): /* Marantz SA-14S1 */ + if (is_marantz_denon_dac(chip->usb_id)) { if (fp->altsetting == 2) return SNDRV_PCM_FMTBIT_DSD_U32_BE; - break; - default: - break; } return 0; |