From 46f5822f7841697d4aedaf4672661d7a765172cd Mon Sep 17 00:00:00 2001 From: Daniel Ribeiro Date: Sun, 7 Jun 2009 02:49:11 -0300 Subject: ASoC: Allow 32 bit registers for DAPM Replace the remaining unsigned shorts with unsigned ints. Tested with pcap2 codec (25 bits registers). Signed-off-by: Daniel Ribeiro Signed-off-by: Mark Brown --- include/sound/soc.h | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'include/sound') diff --git a/include/sound/soc.h b/include/sound/soc.h index cf6111d72b17..a167b4930447 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -216,9 +216,9 @@ void snd_soc_jack_free_gpios(struct snd_soc_jack *jack, int count, /* codec register bit access */ int snd_soc_update_bits(struct snd_soc_codec *codec, unsigned short reg, - unsigned short mask, unsigned short value); + unsigned int mask, unsigned int value); int snd_soc_test_bits(struct snd_soc_codec *codec, unsigned short reg, - unsigned short mask, unsigned short value); + unsigned int mask, unsigned int value); int snd_soc_new_ac97_codec(struct snd_soc_codec *codec, struct snd_ac97_bus_ops *ops, int num); -- cgit v1.2.1 From 291f3bbcacf278726911c713e14cedb71c486b16 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sun, 7 Jun 2009 13:57:17 +0100 Subject: ASoC: Make DAPM power sequence lists local variables They are now only accessed within dapm_power_widgets() so can be local to that function. Signed-off-by: Mark Brown --- include/sound/soc.h | 2 -- 1 file changed, 2 deletions(-) (limited to 'include/sound') diff --git a/include/sound/soc.h b/include/sound/soc.h index cf6111d72b17..5964dd65bbd3 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -369,8 +369,6 @@ struct snd_soc_codec { enum snd_soc_bias_level bias_level; enum snd_soc_bias_level suspend_bias_level; struct delayed_work delayed_work; - struct list_head up_list; - struct list_head down_list; /* codec DAI's */ struct snd_soc_dai *dai; -- cgit v1.2.1 From 831dc0f10f7b2a4856094ff160c018bf19f77527 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sat, 13 Jun 2009 19:55:02 +0100 Subject: ASoC: Add stub suspend and resume calls for ASoC subdevices Now that ASoC subdevices can be regular devices they can have normal suspend and resume calls from their buses. However, suspending them individually is not desirable since this can lead to problems such as pops and clicks from devices being suspended with their signals being amplified or clocks being stopped suddenly. This will be resolved by having the normal device model suspend and resume calls call into ASoC which will suspend the entire card while any of its components are suspended. At present this is not yet implemented but in order to aid the transition of drivers to the standard device model this patch adds API calls for the notifications. Signed-off-by: Mark Brown --- include/sound/soc.h | 5 +++++ 1 file changed, 5 insertions(+) (limited to 'include/sound') diff --git a/include/sound/soc.h b/include/sound/soc.h index 5297ba7e2c41..e6704c0a4404 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -192,6 +192,11 @@ void snd_soc_unregister_platform(struct snd_soc_platform *platform); int snd_soc_register_codec(struct snd_soc_codec *codec); void snd_soc_unregister_codec(struct snd_soc_codec *codec); +#ifdef CONFIG_PM +int snd_soc_suspend_device(struct device *dev); +int snd_soc_resume_device(struct device *dev); +#endif + /* pcm <-> DAI connect */ void snd_soc_free_pcms(struct snd_soc_device *socdev); int snd_soc_new_pcms(struct snd_soc_device *socdev, int idx, const char *xid); -- cgit v1.2.1 From 1abd91849990ed61d6468ffa8b7fc1ae61db4b1a Mon Sep 17 00:00:00 2001 From: Philipp Zabel Date: Mon, 15 Jun 2009 22:18:23 +0200 Subject: ASoC: UDA1380: refactor device registration This patch mostly follows commit 5998102b9095fdb7c67755812038612afea315c5 "ASoC: Refactor WM8731 device registration" to make UDA1380 use standard device instantiation. Similarly, the I2C device registration temporarily moves into the magician machine driver before it will find its final resting place in the board file. At the same time, platform specific configuration is moved to platform data and common power/reset GPIO handling moves into the codec driver. Signed-off-by: Philipp Zabel Signed-off-by: Mark Brown --- include/sound/uda1380.h | 22 ++++++++++++++++++++++ 1 file changed, 22 insertions(+) create mode 100644 include/sound/uda1380.h (limited to 'include/sound') diff --git a/include/sound/uda1380.h b/include/sound/uda1380.h new file mode 100644 index 000000000000..381319c7000c --- /dev/null +++ b/include/sound/uda1380.h @@ -0,0 +1,22 @@ +/* + * UDA1380 ALSA SoC Codec driver + * + * Copyright 2009 Philipp Zabel + * + * 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 __UDA1380_H +#define __UDA1380_H + +struct uda1380_platform_data { + int gpio_power; + int gpio_reset; + int dac_clk; +#define UDA1380_DAC_CLK_SYSCLK 0 +#define UDA1380_DAC_CLK_WSPLL 1 +}; + +#endif /* __UDA1380_H */ -- cgit v1.2.1 From 085f30654175a91c28d2b66b9ea6cceab627fed0 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 16 Jun 2009 13:57:07 +0200 Subject: ALSA: Add new TLV types for dBwith min/max Add new types for TLV dB scale specified with min/max values instead of min/step since the resolution can't match always with the one a device provides. For example, usb audio devices give 1/256 dB resolution while ALSA TLV is based on 1/100 dB resolution. The new min/max types have less problems because the possible rounding error happens only at min/max. Signed-off-by: Takashi Iwai --- include/sound/tlv.h | 14 ++++++++++++++ 1 file changed, 14 insertions(+) (limited to 'include/sound') diff --git a/include/sound/tlv.h b/include/sound/tlv.h index d136ea2181ed..9fd5b19ccf5c 100644 --- a/include/sound/tlv.h +++ b/include/sound/tlv.h @@ -35,6 +35,8 @@ #define SNDRV_CTL_TLVT_DB_SCALE 1 /* dB scale */ #define SNDRV_CTL_TLVT_DB_LINEAR 2 /* linear volume */ #define SNDRV_CTL_TLVT_DB_RANGE 3 /* dB range container */ +#define SNDRV_CTL_TLVT_DB_MINMAX 4 /* dB scale with min/max */ +#define SNDRV_CTL_TLVT_DB_MINMAX_MUTE 5 /* dB scale with min/max with mute */ #define TLV_DB_SCALE_ITEM(min, step, mute) \ SNDRV_CTL_TLVT_DB_SCALE, 2 * sizeof(unsigned int), \ @@ -42,6 +44,18 @@ #define DECLARE_TLV_DB_SCALE(name, min, step, mute) \ unsigned int name[] = { TLV_DB_SCALE_ITEM(min, step, mute) } +/* dB scale specified with min/max values instead of step */ +#define TLV_DB_MINMAX_ITEM(min_dB, max_dB) \ + SNDRV_CTL_TLVT_DB_MINMAX, 2 * sizeof(unsigned int), \ + (min_dB), (max_dB) +#define TLV_DB_MINMAX_MUTE_ITEM(min_dB, max_dB) \ + SNDRV_CTL_TLVT_DB_MINMAX_MUTE, 2 * sizeof(unsigned int), \ + (min_dB), (max_dB) +#define DECLARE_TLV_DB_MINMAX(name, min_dB, max_dB) \ + unsigned int name[] = { TLV_DB_MINMAX_ITEM(min_dB, max_dB) } +#define DECLARE_TLV_DB_MINMAX_MUTE(name, min_dB, max_dB) \ + unsigned int name[] = { TLV_DB_MINMAX_MUTE_ITEM(min_dB, max_dB) } + /* linear volume between min_dB and max_dB (.01dB unit) */ #define TLV_DB_LINEAR_ITEM(min_dB, max_dB) \ SNDRV_CTL_TLVT_DB_LINEAR, 2 * sizeof(unsigned int), \ -- cgit v1.2.1 From 517374704da44c1ba77c1600714fe214524af286 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 22 Jun 2009 13:16:51 +0100 Subject: ASoC: Add a shutdown callback Ensure that the audio subsystem is powered down cleanly when the system shuts down by providing a shutdown operation. This ensures that all the components have been returned to an off state cleanly which should avoid audio issues from partially charged capacitors or noise on digital inputs if the system is restarted quickly. Signed-off-by: Mark Brown Tested-by: Ben Dooks --- include/sound/soc-dapm.h | 1 + 1 file changed, 1 insertion(+) (limited to 'include/sound') diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index ec8a45f9a069..35814ced2d22 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -279,6 +279,7 @@ int snd_soc_dapm_add_routes(struct snd_soc_codec *codec, /* dapm events */ int snd_soc_dapm_stream_event(struct snd_soc_codec *codec, char *stream, int event); +void snd_soc_dapm_shutdown(struct snd_soc_device *socdev); /* dapm sys fs - used by the core */ int snd_soc_dapm_sys_add(struct device *dev); -- cgit v1.2.1 From 096e49d5e6f7bd93395e7ddf7e0239e1644d0505 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sun, 5 Jul 2009 15:12:22 +0100 Subject: ASoC: Add CODEC volatile register operation Add a volatile_register() operation to the CODEC structure providing a standard operation to query if a register is volatile. This will be used to factor out the register cache I/O operations for the CODECs. Signed-off-by: Mark Brown --- include/sound/soc.h | 2 ++ 1 file changed, 2 insertions(+) (limited to 'include/sound') diff --git a/include/sound/soc.h b/include/sound/soc.h index e6704c0a4404..94fcc65609b6 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -191,6 +191,7 @@ int snd_soc_register_platform(struct snd_soc_platform *platform); void snd_soc_unregister_platform(struct snd_soc_platform *platform); int snd_soc_register_codec(struct snd_soc_codec *codec); void snd_soc_unregister_codec(struct snd_soc_codec *codec); +int snd_soc_codec_volatile_register(struct snd_soc_codec *codec, int reg); #ifdef CONFIG_PM int snd_soc_suspend_device(struct device *dev); @@ -361,6 +362,7 @@ struct snd_soc_codec { int (*write)(struct snd_soc_codec *, unsigned int, unsigned int); int (*display_register)(struct snd_soc_codec *, char *, size_t, unsigned int); + int (*volatile_register)(unsigned int); hw_write_t hw_write; hw_read_t hw_read; void *reg_cache; -- cgit v1.2.1 From 17a52fd60a0a0e617ed94aadb1b19751a8fa219e Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sun, 5 Jul 2009 17:24:50 +0100 Subject: ASoC: Begin to factor out register cache I/O functions A lot of CODECs share the same register data formats and therefore replicate the code to manage access to and caching of the register map. In order to reduce code duplication centralised versions of this code will be introduced with drivers able to configure the use of the common code by calling the new snd_soc_codec_set_cache_io() API call during startup. As an initial user the 7 bit address/9 bit data format used by many Wolfson devices is supported for write only CODECs and the drivers with straightforward register cache implementations are converted to use it. Signed-off-by: Mark Brown --- include/sound/soc.h | 2 ++ 1 file changed, 2 insertions(+) (limited to 'include/sound') diff --git a/include/sound/soc.h b/include/sound/soc.h index 94fcc65609b6..27409dd41ae9 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -192,6 +192,8 @@ void snd_soc_unregister_platform(struct snd_soc_platform *platform); int snd_soc_register_codec(struct snd_soc_codec *codec); void snd_soc_unregister_codec(struct snd_soc_codec *codec); int snd_soc_codec_volatile_register(struct snd_soc_codec *codec, int reg); +int snd_soc_codec_set_cache_io(struct snd_soc_codec *codec, + int addr_bits, int data_bits); #ifdef CONFIG_PM int snd_soc_suspend_device(struct device *dev); -- cgit v1.2.1 From cc6a8acdeee932f6911d8b236d2c7d6bcc4616f6 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 17 Jun 2008 16:39:06 +0200 Subject: ALSA: Fix SG-buffer DMA with non-coherent architectures Using SG-buffers with dma_alloc_coherent() is often very inefficient on non-coherent architectures because a tracking record could be allocated in addition for each dma_alloc_coherent() call. Instead, simply disable SG-buffers but just allocate normal continuous buffers on non-supported (currently all but x86) architectures. Signed-off-by: Takashi Iwai --- include/sound/memalloc.h | 6 ++++++ include/sound/pcm.h | 23 +++++++++++++++++++++++ 2 files changed, 29 insertions(+) (limited to 'include/sound') diff --git a/include/sound/memalloc.h b/include/sound/memalloc.h index 7ccce94a5255..c42506212649 100644 --- a/include/sound/memalloc.h +++ b/include/sound/memalloc.h @@ -47,7 +47,11 @@ struct snd_dma_device { #define SNDRV_DMA_TYPE_UNKNOWN 0 /* not defined */ #define SNDRV_DMA_TYPE_CONTINUOUS 1 /* continuous no-DMA memory */ #define SNDRV_DMA_TYPE_DEV 2 /* generic device continuous */ +#ifdef CONFIG_SND_DMA_SGBUF #define SNDRV_DMA_TYPE_DEV_SG 3 /* generic device SG-buffer */ +#else +#define SNDRV_DMA_TYPE_DEV_SG SNDRV_DMA_TYPE_DEV /* no SG-buf support */ +#endif /* * info for buffer allocation @@ -60,6 +64,7 @@ struct snd_dma_buffer { void *private_data; /* private for allocator; don't touch */ }; +#ifdef CONFIG_SND_DMA_SGBUF /* * Scatter-Gather generic device pages */ @@ -107,6 +112,7 @@ static inline void *snd_sgbuf_get_ptr(struct snd_sg_buf *sgbuf, size_t offset) { return sgbuf->table[offset >> PAGE_SHIFT].buf + offset % PAGE_SIZE; } +#endif /* CONFIG_SND_DMA_SGBUF */ /* allocate/release a buffer */ int snd_dma_alloc_pages(int type, struct device *dev, size_t size, diff --git a/include/sound/pcm.h b/include/sound/pcm.h index 23893523dc8c..1691c7fe35af 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -902,6 +902,7 @@ int snd_pcm_lib_preallocate_pages_for_all(struct snd_pcm *pcm, int snd_pcm_lib_malloc_pages(struct snd_pcm_substream *substream, size_t size); int snd_pcm_lib_free_pages(struct snd_pcm_substream *substream); +#ifdef CONFIG_SND_DMA_SGBUF /* * SG-buffer handling */ @@ -927,6 +928,28 @@ struct page *snd_pcm_sgbuf_ops_page(struct snd_pcm_substream *substream, unsigned int snd_pcm_sgbuf_get_chunk_size(struct snd_pcm_substream *substream, unsigned int ofs, unsigned int size); +#else /* !SND_DMA_SGBUF */ +/* + * fake using a continuous buffer + */ +static inline dma_addr_t +snd_pcm_sgbuf_get_addr(struct snd_pcm_substream *substream, unsigned int ofs) +{ + return substream->runtime->dma_addr + ofs; +} + +static inline void * +snd_pcm_sgbuf_get_ptr(struct snd_pcm_substream *substream, unsigned int ofs) +{ + return substream->runtime->dma_area + ofs; +} + +#define snd_pcm_sgbuf_ops_page NULL + +#define snd_pcm_sgbuf_get_chunk_size(subs, ofs, size) (size) + +#endif /* SND_DMA_SGBUF */ + /* handle mmap counter - PCM mmap callback should handle this counter properly */ static inline void snd_pcm_mmap_data_open(struct vm_area_struct *area) { -- cgit v1.2.1 From 942c435ba79fd263a922bb114d56eccab6250662 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 5 Jun 2009 16:32:59 +0100 Subject: ASoC: Add WM8993 CODEC driver The WM8993 is a highly integrated ultra-low power hi-fi CODEC designed for portable devices such as multimedia phones. Signed-off-by: Mark Brown --- include/sound/wm8993.h | 44 ++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 44 insertions(+) create mode 100644 include/sound/wm8993.h (limited to 'include/sound') diff --git a/include/sound/wm8993.h b/include/sound/wm8993.h new file mode 100644 index 000000000000..9c661f2f8cda --- /dev/null +++ b/include/sound/wm8993.h @@ -0,0 +1,44 @@ +/* + * linux/sound/wm8993.h -- Platform data for WM8993 + * + * Copyright 2009 Wolfson Microelectronics. PLC. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef __LINUX_SND_WM8993_H +#define __LINUX_SND_WM8993_H + +/* Note that EQ1 only contains the enable/disable bit so will be + ignored but is included for simplicity. + */ +struct wm8993_retune_mobile_setting { + const char *name; + unsigned int rate; + u16 config[24]; +}; + +struct wm8993_platform_data { + struct wm8993_retune_mobile_setting *retune_configs; + int num_retune_configs; + + /* LINEOUT can be differential or single ended */ + unsigned int lineout1_diff:1; + unsigned int lineout2_diff:1; + + /* Common mode feedback */ + unsigned int lineout1fb:1; + unsigned int lineout2fb:1; + + /* Microphone biases: 0=0.9*AVDD1 1=0.65*AVVD1 */ + unsigned int micbias1_lvl:1; + unsigned int micbias2_lvl:1; + + /* Jack detect threashold levels, see datasheet for values */ + unsigned int jd_scthr:2; + unsigned int jd_thr:2; +}; + +#endif -- cgit v1.2.1 From 47db8e89ac04377fc4de9278d0a3d6e599c04b95 Mon Sep 17 00:00:00 2001 From: Peter Meerwald Date: Mon, 13 Jul 2009 23:05:11 +0100 Subject: ASoC: fixes multiple typos in comments, no functional change Signed-off-by: Peter Meerwald Signed-off-by: Mark Brown --- include/sound/soc-dai.h | 26 +++++++++++++------------- 1 file changed, 13 insertions(+), 13 deletions(-) (limited to 'include/sound') diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index 352d7eee9b6d..05991b0925e0 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -27,8 +27,8 @@ struct snd_pcm_substream; #define SND_SOC_DAIFMT_I2S 0 /* I2S mode */ #define SND_SOC_DAIFMT_RIGHT_J 1 /* Right Justified mode */ #define SND_SOC_DAIFMT_LEFT_J 2 /* Left Justified mode */ -#define SND_SOC_DAIFMT_DSP_A 3 /* L data msb after FRM LRC */ -#define SND_SOC_DAIFMT_DSP_B 4 /* L data msb during FRM LRC */ +#define SND_SOC_DAIFMT_DSP_A 3 /* L data MSB after FRM LRC */ +#define SND_SOC_DAIFMT_DSP_B 4 /* L data MSB during FRM LRC */ #define SND_SOC_DAIFMT_AC97 5 /* AC97 */ /* left and right justified also known as MSB and LSB respectively */ @@ -38,7 +38,7 @@ struct snd_pcm_substream; /* * DAI Clock gating. * - * DAI bit clocks can be be gated (disabled) when not the DAI is not + * DAI bit clocks can be be gated (disabled) when the DAI is not * sending or receiving PCM data in a frame. This can be used to save power. */ #define SND_SOC_DAIFMT_CONT (0 << 4) /* continuous clock */ @@ -51,21 +51,21 @@ struct snd_pcm_substream; * format. */ #define SND_SOC_DAIFMT_NB_NF (0 << 8) /* normal bit clock + frame */ -#define SND_SOC_DAIFMT_NB_IF (1 << 8) /* normal bclk + inv frm */ -#define SND_SOC_DAIFMT_IB_NF (2 << 8) /* invert bclk + nor frm */ -#define SND_SOC_DAIFMT_IB_IF (3 << 8) /* invert bclk + frm */ +#define SND_SOC_DAIFMT_NB_IF (1 << 8) /* normal BCLK + inv FRM */ +#define SND_SOC_DAIFMT_IB_NF (2 << 8) /* invert BCLK + nor FRM */ +#define SND_SOC_DAIFMT_IB_IF (3 << 8) /* invert BCLK + FRM */ /* * DAI hardware clock masters. * * This is wrt the codec, the inverse is true for the interface - * i.e. if the codec is clk and frm master then the interface is + * i.e. if the codec is clk and FRM master then the interface is * clk and frame slave. */ -#define SND_SOC_DAIFMT_CBM_CFM (0 << 12) /* codec clk & frm master */ -#define SND_SOC_DAIFMT_CBS_CFM (1 << 12) /* codec clk slave & frm master */ +#define SND_SOC_DAIFMT_CBM_CFM (0 << 12) /* codec clk & FRM master */ +#define SND_SOC_DAIFMT_CBS_CFM (1 << 12) /* codec clk slave & FRM master */ #define SND_SOC_DAIFMT_CBM_CFS (2 << 12) /* codec clk master & frame slave */ -#define SND_SOC_DAIFMT_CBS_CFS (3 << 12) /* codec clk & frm slave */ +#define SND_SOC_DAIFMT_CBS_CFS (3 << 12) /* codec clk & FRM slave */ #define SND_SOC_DAIFMT_FORMAT_MASK 0x000f #define SND_SOC_DAIFMT_CLOCK_MASK 0x00f0 @@ -116,12 +116,12 @@ int snd_soc_dai_digital_mute(struct snd_soc_dai *dai, int mute); /* * Digital Audio Interface. * - * Describes the Digital Audio Interface in terms of it's ALSA, DAI and AC97 - * operations an capabilities. Codec and platfom drivers will register a this + * Describes the Digital Audio Interface in terms of its ALSA, DAI and AC97 + * operations and capabilities. Codec and platform drivers will register this * structure for every DAI they have. * * This structure covers the clocking, formating and ALSA operations for each - * interface a + * interface. */ struct snd_soc_dai_ops { /* -- cgit v1.2.1 From d0af93db12639c425adee795fabadedb52182346 Mon Sep 17 00:00:00 2001 From: Joonyoung Shim Date: Wed, 15 Jul 2009 20:33:47 +0900 Subject: ASoC: add SOC_DOUBLE_EXT_TLV control type This is a macro for double controls with special callback function and TLV. The SOC_DOUBLE_EXT_TLV needs one register and two shifts for double controls. Signed-off-by: Joonyoung Shim Signed-off-by: Mark Brown --- include/sound/soc.h | 11 +++++++++++ 1 file changed, 11 insertions(+) (limited to 'include/sound') diff --git a/include/sound/soc.h b/include/sound/soc.h index 94fcc65609b6..795a6b4a67ee 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -135,6 +135,17 @@ .info = snd_soc_info_volsw, \ .get = xhandler_get, .put = xhandler_put, \ .private_value = SOC_SINGLE_VALUE(xreg, xshift, xmax, xinvert) } +#define SOC_DOUBLE_EXT_TLV(xname, xreg, shift_left, shift_right, xmax, xinvert,\ + xhandler_get, xhandler_put, tlv_array) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), \ + .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ | \ + SNDRV_CTL_ELEM_ACCESS_READWRITE, \ + .tlv.p = (tlv_array), \ + .info = snd_soc_info_volsw, \ + .get = xhandler_get, .put = xhandler_put, \ + .private_value = (unsigned long)&(struct soc_mixer_control) \ + {.reg = xreg, .shift = shift_left, .rshift = shift_right, \ + .max = xmax, .invert = xinvert} } #define SOC_SINGLE_BOOL_EXT(xname, xdata, xhandler_get, xhandler_put) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ .info = snd_soc_info_bool_ext, \ -- cgit v1.2.1 From 3ce91d5a5a47eca6308c0a64f768c7a4466e0407 Mon Sep 17 00:00:00 2001 From: Joonyoung Shim Date: Wed, 15 Jul 2009 20:33:50 +0900 Subject: ASoC: add SOC_DOUBLE_R_EXT_TLV control type This is a macro for double controls with special callback function and TLV. The SOC_DOUBLE_R_EXT_TLV needs two registers and one shift for double controls. Signed-off-by: Joonyoung Shim Signed-off-by: Mark Brown --- include/sound/soc.h | 11 +++++++++++ 1 file changed, 11 insertions(+) (limited to 'include/sound') diff --git a/include/sound/soc.h b/include/sound/soc.h index 795a6b4a67ee..756fb59772d1 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -146,6 +146,17 @@ .private_value = (unsigned long)&(struct soc_mixer_control) \ {.reg = xreg, .shift = shift_left, .rshift = shift_right, \ .max = xmax, .invert = xinvert} } +#define SOC_DOUBLE_R_EXT_TLV(xname, reg_left, reg_right, xshift, xmax, xinvert,\ + xhandler_get, xhandler_put, tlv_array) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), \ + .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ | \ + SNDRV_CTL_ELEM_ACCESS_READWRITE, \ + .tlv.p = (tlv_array), \ + .info = snd_soc_info_volsw_2r, \ + .get = xhandler_get, .put = xhandler_put, \ + .private_value = (unsigned long)&(struct soc_mixer_control) \ + {.reg = reg_left, .rreg = reg_right, .shift = xshift, \ + .max = xmax, .invert = xinvert} } #define SOC_SINGLE_BOOL_EXT(xname, xdata, xhandler_get, xhandler_put) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ .info = snd_soc_info_bool_ext, \ -- cgit v1.2.1 From 474828a40f6ddab6e2a3475a19c5c84aa3ec7d60 Mon Sep 17 00:00:00 2001 From: Marek Vasut Date: Wed, 22 Jul 2009 13:01:03 +0200 Subject: ALSA: Allow passing platform_data to devices attached to AC97 bus This patch allows passing platform_data to devices attached to AC97 bus (like touchscreens, battery measurement chips ...). Signed-off-by: Marek Vasut Signed-off-by: Mark Brown --- include/sound/ac97_codec.h | 6 ++++++ include/sound/soc-dai.h | 1 + 2 files changed, 7 insertions(+) (limited to 'include/sound') diff --git a/include/sound/ac97_codec.h b/include/sound/ac97_codec.h index 251fc1cd5002..9b1c0985480c 100644 --- a/include/sound/ac97_codec.h +++ b/include/sound/ac97_codec.h @@ -642,4 +642,10 @@ int snd_ac97_pcm_double_rate_rules(struct snd_pcm_runtime *runtime); /* ad hoc AC97 device driver access */ extern struct bus_type ac97_bus_type; +/* AC97 platform_data adding function */ +static inline void snd_ac97_dev_add_pdata(struct snd_ac97 *ac97, void *data) +{ + ac97->dev.platform_data = data; +} + #endif /* __SOUND_AC97_CODEC_H */ diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index 05991b0925e0..25d62ac53fc5 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -179,6 +179,7 @@ struct snd_soc_dai { int ac97_control; struct device *dev; + void *ac97_pdata; /* platform_data for the ac97 codec */ /* DAI callbacks */ int (*probe)(struct platform_device *pdev, -- cgit v1.2.1 From 77ee09c67e051a5ebd19a53ba3945dbdc8d21b3c Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 31 Jul 2009 18:26:51 +0100 Subject: ASoC: Allow CODECs to flag invalid registers This helps CODECs with sparse register maps work better with the register cache display interface. Signed-off-by: Mark Brown --- include/sound/soc.h | 1 + 1 file changed, 1 insertion(+) (limited to 'include/sound') diff --git a/include/sound/soc.h b/include/sound/soc.h index 756fb59772d1..55b330937260 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -385,6 +385,7 @@ struct snd_soc_codec { int (*display_register)(struct snd_soc_codec *, char *, size_t, unsigned int); int (*volatile_register)(unsigned int); + int (*readable_register)(unsigned int); hw_write_t hw_write; hw_read_t hw_read; void *reg_cache; -- cgit v1.2.1 From 7084a42b965d972079201414d19a399e65b26099 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 10 Jul 2009 22:24:27 +0100 Subject: ASoC: Add I/O control bus information to factored out cache setup While writes tend to be able to use a fairly bus independant format to do the writes reads are all bus specific. To allow us to factor out this code include the bus type as a parameter when setting up the cache. Initially just use this to factor out hw_write_t for I2C. Signed-off-by: Mark Brown --- include/sound/soc.h | 9 ++++++++- 1 file changed, 8 insertions(+), 1 deletion(-) (limited to 'include/sound') diff --git a/include/sound/soc.h b/include/sound/soc.h index 27409dd41ae9..d0b29a509bdd 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -187,13 +187,20 @@ typedef int (*hw_read_t)(void *,char* ,int); extern struct snd_ac97_bus_ops soc_ac97_ops; +enum snd_soc_control_type { + SND_SOC_CUSTOM, + SND_SOC_I2C, + SND_SOC_SPI, +}; + int snd_soc_register_platform(struct snd_soc_platform *platform); void snd_soc_unregister_platform(struct snd_soc_platform *platform); int snd_soc_register_codec(struct snd_soc_codec *codec); void snd_soc_unregister_codec(struct snd_soc_codec *codec); int snd_soc_codec_volatile_register(struct snd_soc_codec *codec, int reg); int snd_soc_codec_set_cache_io(struct snd_soc_codec *codec, - int addr_bits, int data_bits); + int addr_bits, int data_bits, + enum snd_soc_control_type control); #ifdef CONFIG_PM int snd_soc_suspend_device(struct device *dev); -- cgit v1.2.1 From afa2f1066e7288a9e4f8e3fda277da245219dffc Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 10 Jul 2009 23:11:24 +0100 Subject: ASoC: Factor out I2C 8 bit address 16 bit data I/O As part of this refactoring the type of the CODEC hw_read operation is changed to match the regular read operation. Signed-off-by: Mark Brown --- include/sound/soc.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'include/sound') diff --git a/include/sound/soc.h b/include/sound/soc.h index d0b29a509bdd..4a5846e72473 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -373,7 +373,7 @@ struct snd_soc_codec { size_t, unsigned int); int (*volatile_register)(unsigned int); hw_write_t hw_write; - hw_read_t hw_read; + unsigned int (*hw_read)(struct snd_soc_codec *, unsigned int); void *reg_cache; short reg_cache_size; short reg_cache_step; -- cgit v1.2.1 From a5479e389e989acfeca9c32eeb0083d086202280 Mon Sep 17 00:00:00 2001 From: Daniel Ribeiro Date: Mon, 15 Jun 2009 21:44:31 -0300 Subject: ASoC: change set_tdm_slot api to allow slot_width override. Extend set_tdm_slot to allow the user to arbitrarily set the frame width and active TX/RX slots. Updates magician.c and wm9081.c for the new set_tdm_slot(). wm9081.c still doesn't handle the slot_width override. While being there, correct an incorrect use of SlotsPerFrm(7) use in bitmask on pxa-ssp.c (SSCR0_SlotsPerFrm(x) is (((x) - 1) << 24)) ). (this series is meant for Mark's for-2.6.32 branch) Signed-off-by: Daniel Ribeiro Signed-off-by: Mark Brown --- include/sound/soc-dai.h | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) (limited to 'include/sound') diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index 25d62ac53fc5..2d3fa2950aa1 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -106,7 +106,7 @@ int snd_soc_dai_set_pll(struct snd_soc_dai *dai, int snd_soc_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt); int snd_soc_dai_set_tdm_slot(struct snd_soc_dai *dai, - unsigned int mask, int slots); + unsigned int tx_mask, unsigned int rx_mask, int slots, int slot_width); int snd_soc_dai_set_tristate(struct snd_soc_dai *dai, int tristate); @@ -140,7 +140,8 @@ struct snd_soc_dai_ops { */ int (*set_fmt)(struct snd_soc_dai *dai, unsigned int fmt); int (*set_tdm_slot)(struct snd_soc_dai *dai, - unsigned int mask, int slots); + unsigned int tx_mask, unsigned int rx_mask, + int slots, int slot_width); int (*set_tristate)(struct snd_soc_dai *dai, int tristate); /* -- cgit v1.2.1 From 8f738d58425625faf0c1a6dbfdd4458545338551 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sun, 9 Aug 2009 20:08:31 +0100 Subject: ASoC: Define more formats for the AC97 CODECs Signed-off-by: Mark Brown --- include/sound/soc-dai.h | 8 +++++++- 1 file changed, 7 insertions(+), 1 deletion(-) (limited to 'include/sound') diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index 25d62ac53fc5..50c6a0e295b7 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -78,7 +78,13 @@ struct snd_pcm_substream; #define SND_SOC_CLOCK_IN 0 #define SND_SOC_CLOCK_OUT 1 -#define SND_SOC_STD_AC97_FMTS (SNDRV_PCM_FMTBIT_S16_LE |\ +#define SND_SOC_STD_AC97_FMTS (SNDRV_PCM_FMTBIT_S8 |\ + SNDRV_PCM_FMTBIT_S16_LE |\ + SNDRV_PCM_FMTBIT_S16_BE |\ + SNDRV_PCM_FMTBIT_S20_3LE |\ + SNDRV_PCM_FMTBIT_S20_3BE |\ + SNDRV_PCM_FMTBIT_S24_3LE |\ + SNDRV_PCM_FMTBIT_S24_3BE |\ SNDRV_PCM_FMTBIT_S32_LE |\ SNDRV_PCM_FMTBIT_S32_BE) -- cgit v1.2.1 From 6e2efaacb3579fd9643d0dc59963b58b801c03a1 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Mon, 10 Aug 2009 10:06:53 +0200 Subject: sound: ymfpci: increase timer resolution to 96 kHz Allow the interval timer to be programmed with its full 96 kHz precision. Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai --- include/sound/ymfpci.h | 1 + 1 file changed, 1 insertion(+) (limited to 'include/sound') diff --git a/include/sound/ymfpci.h b/include/sound/ymfpci.h index 05ead6698434..444cd6ba0ba7 100644 --- a/include/sound/ymfpci.h +++ b/include/sound/ymfpci.h @@ -331,6 +331,7 @@ struct snd_ymfpci { struct snd_ac97 *ac97; struct snd_rawmidi *rawmidi; struct snd_timer *timer; + unsigned int timer_ticks; struct pci_dev *pci; struct snd_card *card; -- cgit v1.2.1 From 4ac0478f2afaf8e778b4190d6218459a9dbf2a8f Mon Sep 17 00:00:00 2001 From: Marek Vasut Date: Thu, 30 Jul 2009 02:55:01 +0200 Subject: ALSA: Allow passing platform_data for pxa2xx-ac97 This patch adds support for passing platform data to ac97 bus devices from PXA2xx-AC97 driver.. Signed-off-by: Marek Vasut Signed-off-by: Mark Brown --- include/sound/ac97_codec.h | 3 +++ 1 file changed, 3 insertions(+) (limited to 'include/sound') diff --git a/include/sound/ac97_codec.h b/include/sound/ac97_codec.h index 9b1c0985480c..3dae3f799b9b 100644 --- a/include/sound/ac97_codec.h +++ b/include/sound/ac97_codec.h @@ -32,6 +32,9 @@ #include "control.h" #include "info.h" +/* maximum number of devices on the AC97 bus */ +#define AC97_BUS_MAX_DEVICES 4 + /* * AC'97 codec registers */ -- cgit v1.2.1 From 010ff262269c6ad84acba98eab2d7843919c7ccf Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 17 Aug 2009 17:39:22 +0100 Subject: ASoC: Add input and output AIF widgets Currently DAPM interfaces with the audio streams to and from the processor at the DAC and ADC widgets. As the digital capabilities of parts increases this is becoming a less and less able to meet the needs of parts. To meet the needs of these devices create new widgets interfacing with the TDM bus but not integrated into any other functionality. Audio can then be routed to and from these widgets using existing routing widgets. A slot number is provided in the definition but this is currently not used yet. This is intended to support devices which can use more than one TDM slot on a single interface. Signed-off-by: Mark Brown --- include/sound/soc-dapm.h | 8 ++++++++ 1 file changed, 8 insertions(+) (limited to 'include/sound') diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index 35814ced2d22..338840510617 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -137,6 +137,12 @@ .event_flags = SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD} /* stream domain */ +#define SND_SOC_DAPM_AIF_IN(wname, stname, wslot, wreg, wshift, winvert) \ +{ .id = snd_soc_dapm_aif_in, .name = wname, .sname = stname, \ + .reg = wreg, .shift = wshift, .invert = winvert } +#define SND_SOC_DAPM_AIF_OUT(wname, stname, wslot, wreg, wshift, winvert) \ +{ .id = snd_soc_dapm_aif_out, .name = wname, .sname = stname, \ + .reg = wreg, .shift = wshift, .invert = winvert } #define SND_SOC_DAPM_DAC(wname, stname, wreg, wshift, winvert) \ { .id = snd_soc_dapm_dac, .name = wname, .sname = stname, .reg = wreg, \ .shift = wshift, .invert = winvert} @@ -312,6 +318,8 @@ enum snd_soc_dapm_type { snd_soc_dapm_pre, /* machine specific pre widget - exec first */ snd_soc_dapm_post, /* machine specific post widget - exec last */ snd_soc_dapm_supply, /* power/clock supply */ + snd_soc_dapm_aif_in, /* audio interface input */ + snd_soc_dapm_aif_out, /* audio interface output */ }; /* -- cgit v1.2.1 From a4d7d550a9cfdfbc615383a08e9afa39d5a6d875 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Thu, 20 Aug 2009 21:01:05 +0900 Subject: ASoC: Add SuperH FSI driver support for ALSA This driver is very simple. It support playback only now. This patch is tested by ms7724se board. Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- include/sound/sh_fsi.h | 83 ++++++++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 83 insertions(+) create mode 100644 include/sound/sh_fsi.h (limited to 'include/sound') diff --git a/include/sound/sh_fsi.h b/include/sound/sh_fsi.h new file mode 100644 index 000000000000..c0227361a876 --- /dev/null +++ b/include/sound/sh_fsi.h @@ -0,0 +1,83 @@ +#ifndef __SOUND_FSI_H +#define __SOUND_FSI_H + +/* + * Fifo-attached Serial Interface (FSI) support for SH7724 + * + * Copyright (C) 2009 Renesas Solutions Corp. + * Kuninori Morimoto + * + * 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. + */ + +/* flags format + + * 0xABCDEEFF + * + * A: channel size for TDM (input) + * B: channel size for TDM (ooutput) + * C: inversion + * D: mode + * E: input format + * F: output format + */ + +#include +#include + +/* TDM channel */ +#define SH_FSI_SET_CH_I(x) ((x & 0xF) << 28) +#define SH_FSI_SET_CH_O(x) ((x & 0xF) << 24) + +#define SH_FSI_CH_IMASK 0xF0000000 +#define SH_FSI_CH_OMASK 0x0F000000 +#define SH_FSI_GET_CH_I(x) ((x & SH_FSI_CH_IMASK) >> 28) +#define SH_FSI_GET_CH_O(x) ((x & SH_FSI_CH_OMASK) >> 24) + +/* clock inversion */ +#define SH_FSI_INVERSION_MASK 0x00F00000 +#define SH_FSI_LRM_INV (1 << 20) +#define SH_FSI_BRM_INV (1 << 21) +#define SH_FSI_LRS_INV (1 << 22) +#define SH_FSI_BRS_INV (1 << 23) + +/* mode */ +#define SH_FSI_MODE_MASK 0x000F0000 +#define SH_FSI_IN_SLAVE_MODE (1 << 16) /* default master mode */ +#define SH_FSI_OUT_SLAVE_MODE (1 << 17) /* default master mode */ + +/* DI format */ +#define SH_FSI_FMT_MASK 0x000000FF +#define SH_FSI_IFMT(x) (((SH_FSI_FMT_ ## x) & SH_FSI_FMT_MASK) << 8) +#define SH_FSI_OFMT(x) (((SH_FSI_FMT_ ## x) & SH_FSI_FMT_MASK) << 0) +#define SH_FSI_GET_IFMT(x) ((x >> 8) & SH_FSI_FMT_MASK) +#define SH_FSI_GET_OFMT(x) ((x >> 0) & SH_FSI_FMT_MASK) + +#define SH_FSI_FMT_MONO (1 << 0) +#define SH_FSI_FMT_MONO_DELAY (1 << 1) +#define SH_FSI_FMT_PCM (1 << 2) +#define SH_FSI_FMT_I2S (1 << 3) +#define SH_FSI_FMT_TDM (1 << 4) +#define SH_FSI_FMT_TDM_DELAY (1 << 5) + +#define SH_FSI_IFMT_TDM_CH(x) \ + (SH_FSI_IFMT(TDM) | SH_FSI_SET_CH_I(x)) +#define SH_FSI_IFMT_TDM_DELAY_CH(x) \ + (SH_FSI_IFMT(TDM_DELAY) | SH_FSI_SET_CH_I(x)) + +#define SH_FSI_OFMT_TDM_CH(x) \ + (SH_FSI_OFMT(TDM) | SH_FSI_SET_CH_O(x)) +#define SH_FSI_OFMT_TDM_DELAY_CH(x) \ + (SH_FSI_OFMT(TDM_DELAY) | SH_FSI_SET_CH_O(x)) + +struct sh_fsi_platform_info { + unsigned long porta_flags; + unsigned long portb_flags; +}; + +extern struct snd_soc_dai fsi_soc_dai[2]; +extern struct snd_soc_platform fsi_soc_platform; + +#endif /* __SOUND_FSI_H */ -- cgit v1.2.1 From 79fb9387f88b6b44bbc46e19cae26d2c9fe3bb6a Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 21 Aug 2009 16:38:13 +0100 Subject: ASoC: Add DAPM widget power decision debugfs files Currently when built with DEBUG DAPM will dump information about the power state decisions it is taking for each widget to dmesg. This isn't an ideal way of getting the information - it requires a kernel build to turn it on and off and for large hub CODECs the volume of information is so large as to be illegible. When the output goes to the console it can also cause a noticable impact on performance simply to print it out. Improve the situation by adding a dapm directory to our debugfs tree containing a file per widget with the same information in it. This still requires a decision to build with debugfs support but is easier to navigate and much less intrusive. In addition to the previously displayed information active streams are also shown in these files. Signed-off-by: Mark Brown --- include/sound/soc-dapm.h | 1 + include/sound/soc.h | 1 + 2 files changed, 2 insertions(+) (limited to 'include/sound') diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index 35814ced2d22..1673f0b2cf58 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -283,6 +283,7 @@ void snd_soc_dapm_shutdown(struct snd_soc_device *socdev); /* dapm sys fs - used by the core */ int snd_soc_dapm_sys_add(struct device *dev); +void snd_soc_dapm_debugfs_init(struct snd_soc_codec *codec); /* dapm audio pin control and status */ int snd_soc_dapm_enable_pin(struct snd_soc_codec *codec, const char *pin); diff --git a/include/sound/soc.h b/include/sound/soc.h index dbb1702688cd..0758a1b3ca44 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -416,6 +416,7 @@ struct snd_soc_codec { #ifdef CONFIG_DEBUG_FS struct dentry *debugfs_reg; struct dentry *debugfs_pop_time; + struct dentry *debugfs_dapm; #endif }; -- cgit v1.2.1 From 36ce99c1dcab2978fc1900f19b431adedd8f99f6 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 27 Aug 2009 16:45:07 +0200 Subject: ALSA: Add debug module option Add debug module option to snd core. This controls the debug print level. When CONFIG_SND_DEBUG_VERBOSE is set, you can suppress the debug messages by giving or changing this parameter to a lower value. debug=0 means no debug messsages. As default, it's set to the verbose level 2. Since this option can be changed dynamically via sysfs file, you can suppress the verbose debug messages on the fly, which wasn't possible before. Signed-off-by: Takashi Iwai --- include/sound/core.h | 38 ++++++++++++++------------------------ 1 file changed, 14 insertions(+), 24 deletions(-) (limited to 'include/sound') diff --git a/include/sound/core.h b/include/sound/core.h index 309cb9659a05..a89728db5584 100644 --- a/include/sound/core.h +++ b/include/sound/core.h @@ -340,18 +340,17 @@ unsigned int snd_dma_pointer(unsigned long dma, unsigned int size); struct resource; void release_and_free_resource(struct resource *res); -#ifdef CONFIG_SND_VERBOSE_PRINTK -void snd_verbose_printk(const char *file, int line, const char *format, ...) - __attribute__ ((format (printf, 3, 4))); -#endif -#if defined(CONFIG_SND_DEBUG) && defined(CONFIG_SND_VERBOSE_PRINTK) -void snd_verbose_printd(const char *file, int line, const char *format, ...) - __attribute__ ((format (printf, 3, 4))); -#endif - /* --- */ -#ifdef CONFIG_SND_VERBOSE_PRINTK +#if defined(CONFIG_SND_DEBUG) || defined(CONFIG_SND_VERBOSE_PRINTK) +void __snd_printk(unsigned int level, const char *file, int line, + const char *format, ...) + __attribute__ ((format (printf, 4, 5))); +#else +#define __snd_printk(level, file, line, format, args...) \ + prinkt(format, ##args) +#endif + /** * snd_printk - printk wrapper * @fmt: format string @@ -360,15 +359,9 @@ void snd_verbose_printd(const char *file, int line, const char *format, ...) * when configured with CONFIG_SND_VERBOSE_PRINTK. */ #define snd_printk(fmt, args...) \ - snd_verbose_printk(__FILE__, __LINE__, fmt ,##args) -#else -#define snd_printk(fmt, args...) \ - printk(fmt ,##args) -#endif + __snd_printk(0, __FILE__, __LINE__, fmt, ##args) #ifdef CONFIG_SND_DEBUG - -#ifdef CONFIG_SND_VERBOSE_PRINTK /** * snd_printd - debug printk * @fmt: format string @@ -377,11 +370,7 @@ void snd_verbose_printd(const char *file, int line, const char *format, ...) * Ignored when CONFIG_SND_DEBUG is not set. */ #define snd_printd(fmt, args...) \ - snd_verbose_printd(__FILE__, __LINE__, fmt ,##args) -#else -#define snd_printd(fmt, args...) \ - printk(fmt ,##args) -#endif + __snd_printk(1, __FILE__, __LINE__, fmt, ##args) /** * snd_BUG - give a BUG warning message and stack trace @@ -428,9 +417,10 @@ static inline int __snd_bug_on(int cond) * Works like snd_printk() for debugging purposes. * Ignored when CONFIG_SND_DEBUG_VERBOSE is not set. */ -#define snd_printdd(format, args...) snd_printk(format, ##args) +#define snd_printdd(format, args...) \ + __snd_printk(2, __FILE__, __LINE__, format, ##args) #else -#define snd_printdd(format, args...) /* nothing */ +#define snd_printdd(format, args...) do { } while (0) #endif -- cgit v1.2.1 From 5a53a7640a7af7acf904ed805c6fd1bf9fea829c Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 27 Aug 2009 21:04:12 +0200 Subject: ALSA: pcm - Increase protocol version Increase the PCM protocol version to indicate the drain ioctl behavior change. Signed-off-by: Takashi Iwai --- include/sound/asound.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'include/sound') diff --git a/include/sound/asound.h b/include/sound/asound.h index 82aed3f47534..1f57bb92eb5a 100644 --- a/include/sound/asound.h +++ b/include/sound/asound.h @@ -138,7 +138,7 @@ struct snd_hwdep_dsp_image { * * *****************************************************************************/ -#define SNDRV_PCM_VERSION SNDRV_PROTOCOL_VERSION(2, 0, 9) +#define SNDRV_PCM_VERSION SNDRV_PROTOCOL_VERSION(2, 0, 10) typedef unsigned long snd_pcm_uframes_t; typedef signed long snd_pcm_sframes_t; -- cgit v1.2.1 From cf0baf16c3a3b3dd67ea3df346479032ab10e988 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 28 Aug 2009 07:22:05 +0200 Subject: ALSA: Fixed a typo of printk() Fixed a silly typo of printk() included in the previous patch... Signed-off-by: Takashi Iwai --- include/sound/core.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'include/sound') diff --git a/include/sound/core.h b/include/sound/core.h index a89728db5584..1ec992b3f1d0 100644 --- a/include/sound/core.h +++ b/include/sound/core.h @@ -348,7 +348,7 @@ void __snd_printk(unsigned int level, const char *file, int line, __attribute__ ((format (printf, 4, 5))); #else #define __snd_printk(level, file, line, format, args...) \ - prinkt(format, ##args) + printk(format, ##args) #endif /** -- cgit v1.2.1 From 9d32e03d01649d2dd837923470f3f323e3b88253 Mon Sep 17 00:00:00 2001 From: Jaroslav Kysela Date: Mon, 31 Aug 2009 15:07:23 +0200 Subject: ALSA: Release v1.0.21 Signed-off-by: Jaroslav Kysela Signed-off-by: Takashi Iwai --- include/sound/version.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'include/sound') diff --git a/include/sound/version.h b/include/sound/version.h index 456f1359e1c0..22939142dd23 100644 --- a/include/sound/version.h +++ b/include/sound/version.h @@ -1,3 +1,3 @@ /* include/version.h */ -#define CONFIG_SND_VERSION "1.0.20" +#define CONFIG_SND_VERSION "1.0.21" #define CONFIG_SND_DATE "" -- cgit v1.2.1 From 236cc52856f6ebe47f52d50ba5431b0e172fd0d1 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 7 Sep 2009 12:46:42 +0100 Subject: ASoC: Remove unuused hw_read_t Signed-off-by: Mark Brown --- include/sound/soc.h | 1 - 1 file changed, 1 deletion(-) (limited to 'include/sound') diff --git a/include/sound/soc.h b/include/sound/soc.h index 0758a1b3ca44..475cb7ed6bec 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -205,7 +205,6 @@ struct snd_soc_jack_gpio; #endif typedef int (*hw_write_t)(void *,const char* ,int); -typedef int (*hw_read_t)(void *,char* ,int); extern struct snd_ac97_bus_ops soc_ac97_ops; -- cgit v1.2.1 From 82a783f4bcb878e6c4f02e24c7cd0687bdea7443 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 7 Sep 2009 15:50:18 +0200 Subject: ALSA: Remove struct snd_monitor_file from public sound/core.h The struct snd_monitor_file is used locally only in sound/core/init.c, thus it should be moved there from the public sound/core.h. Signed-off-by: Takashi Iwai --- include/sound/core.h | 9 --------- 1 file changed, 9 deletions(-) (limited to 'include/sound') diff --git a/include/sound/core.h b/include/sound/core.h index 309cb9659a05..3bb07bc62255 100644 --- a/include/sound/core.h +++ b/include/sound/core.h @@ -93,15 +93,6 @@ struct snd_device { #define snd_device(n) list_entry(n, struct snd_device, list) -/* monitor files for graceful shutdown (hotplug) */ - -struct snd_monitor_file { - struct file *file; - const struct file_operations *disconnected_f_op; - struct list_head shutdown_list; /* still need to shutdown */ - struct list_head list; /* link of monitor files */ -}; - /* main structure for soundcard */ struct snd_card { -- cgit v1.2.1 From b8c60ede6abf8e96a892c114131700b0cfb0be89 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 7 Sep 2009 15:52:30 +0200 Subject: ALSA: Remove unneeded ifdef from sound/core.h Remove the old hack that was needed for building alsa-driver modules externally for old kernels. Signed-off-by: Takashi Iwai --- include/sound/core.h | 4 ---- 1 file changed, 4 deletions(-) (limited to 'include/sound') diff --git a/include/sound/core.h b/include/sound/core.h index 3bb07bc62255..f545efcf03f2 100644 --- a/include/sound/core.h +++ b/include/sound/core.h @@ -302,9 +302,7 @@ int snd_component_add(struct snd_card *card, const char *component); int snd_card_file_add(struct snd_card *card, struct file *file); int snd_card_file_remove(struct snd_card *card, struct file *file); -#ifndef snd_card_set_dev #define snd_card_set_dev(card, devptr) ((card)->dev = (devptr)) -#endif /* device.c */ @@ -429,12 +427,10 @@ static inline int __snd_bug_on(int cond) /* for easier backward-porting */ #if defined(CONFIG_GAMEPORT) || defined(CONFIG_GAMEPORT_MODULE) -#ifndef gameport_set_dev_parent #define gameport_set_dev_parent(gp,xdev) ((gp)->dev.parent = (xdev)) #define gameport_set_port_data(gp,r) ((gp)->port_data = (r)) #define gameport_get_port_data(gp) (gp)->port_data #endif -#endif /* PCI quirk list helper */ struct snd_pci_quirk { -- cgit v1.2.1 From 6e5265ec34d3b9578973841ddec8b925e986136a Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 8 Sep 2009 14:26:51 +0200 Subject: ALSA: Re-export snd_pcm_format_name() function Re-export snd_pcm_format_name() function to be used outside the PCM core. As a first example, usbaudio is changed to use it now again. Signed-off-by: Takashi Iwai --- include/sound/pcm.h | 2 ++ 1 file changed, 2 insertions(+) (limited to 'include/sound') diff --git a/include/sound/pcm.h b/include/sound/pcm.h index 23893523dc8c..4d5b2407514e 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -965,4 +965,6 @@ static inline void snd_pcm_limit_isa_dma_size(int dma, size_t *max) #define PCM_RUNTIME_CHECK(sub) snd_BUG_ON(!(sub) || !(sub)->runtime) +const char *snd_pcm_format_name(snd_pcm_format_t format); + #endif /* __SOUND_PCM_H */ -- cgit v1.2.1 From 4f7454a9970fa0f3e9f1a68201520e3df1bb5224 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 8 Sep 2009 14:29:58 +0200 Subject: ALSA: Add const prefix to proc helper functions Add appropriate const prefix to char * arguments in proc helper functions. Also fixed the caller side to be proper const pointers. Signed-off-by: Takashi Iwai --- include/sound/info.h | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'include/sound') diff --git a/include/sound/info.h b/include/sound/info.h index 7c2ee1a21b00..112e8949e1a7 100644 --- a/include/sound/info.h +++ b/include/sound/info.h @@ -110,13 +110,13 @@ void snd_card_info_read_oss(struct snd_info_buffer *buffer); static inline void snd_card_info_read_oss(struct snd_info_buffer *buffer) {} #endif -int snd_iprintf(struct snd_info_buffer *buffer, char *fmt, ...) \ +int snd_iprintf(struct snd_info_buffer *buffer, const char *fmt, ...) \ __attribute__ ((format (printf, 2, 3))); int snd_info_init(void); int snd_info_done(void); int snd_info_get_line(struct snd_info_buffer *buffer, char *line, int len); -char *snd_info_get_str(char *dest, char *src, int len); +const char *snd_info_get_str(char *dest, const char *src, int len); struct snd_info_entry *snd_info_create_module_entry(struct module *module, const char *name, struct snd_info_entry *parent); -- cgit v1.2.1