From 1aad779fccdbb4d79af7b9de93dfd2bfe807e052 Mon Sep 17 00:00:00 2001 From: Ola Lilja Date: Thu, 24 May 2012 15:26:03 +0200 Subject: ALSA: pcm: Add debug-print helper function Adds a function getting the stream-name as a string for a specific stream. Signed-off-by: Ola Lilja Reviewed-by: Takashi Iwai Signed-off-by: Mark Brown --- include/sound/pcm.h | 11 +++++++++++ 1 file changed, 11 insertions(+) (limited to 'include') diff --git a/include/sound/pcm.h b/include/sound/pcm.h index 0d1112815be3..a55d5db7eb5a 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -1073,4 +1073,15 @@ static inline void snd_pcm_limit_isa_dma_size(int dma, size_t *max) const char *snd_pcm_format_name(snd_pcm_format_t format); +/** + * Get a string naming the direction of a stream + */ +static inline const char *snd_pcm_stream_str(struct snd_pcm_substream *substream) +{ + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + return "Playback"; + else + return "Capture"; +} + #endif /* __SOUND_PCM_H */ -- cgit v1.2.3 From d7e7eb91551ad99244b989d71d092cb0375648fa Mon Sep 17 00:00:00 2001 From: Ola Lilja Date: Thu, 24 May 2012 15:26:25 +0200 Subject: ASoC: core: Add widget SND_SOC_DAPM_CLOCK_SUPPLY Adds a supply-widget variant for connection to the clock-framework. This widget-type corresponds to the variant for regulators. Signed-off-by: Ola Lilja Signed-off-by: Mark Brown --- include/sound/soc-dapm.h | 10 ++++++++++ 1 file changed, 10 insertions(+) (limited to 'include') diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index e3833d9f1914..05559e571d44 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -229,6 +229,10 @@ struct device; { .id = snd_soc_dapm_adc, .name = wname, .sname = stname, .reg = wreg, \ .shift = wshift, .invert = winvert, \ .event = wevent, .event_flags = wflags} +#define SND_SOC_DAPM_CLOCK_SUPPLY(wname) \ +{ .id = snd_soc_dapm_clock_supply, .name = wname, \ + .reg = SND_SOC_NOPM, .event = dapm_clock_event, \ + .event_flags = SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD } /* generic widgets */ #define SND_SOC_DAPM_REG(wid, wname, wreg, wshift, wmask, won_val, woff_val) \ @@ -245,6 +249,7 @@ struct device; .reg = SND_SOC_NOPM, .shift = wdelay, .event = dapm_regulator_event, \ .event_flags = SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD } + /* dapm kcontrol types */ #define SOC_DAPM_SINGLE(xname, reg, shift, max, invert) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ @@ -327,6 +332,8 @@ int dapm_reg_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event); int dapm_regulator_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event); +int dapm_clock_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event); /* dapm controls */ int snd_soc_dapm_put_volsw(struct snd_kcontrol *kcontrol, @@ -432,6 +439,7 @@ enum snd_soc_dapm_type { snd_soc_dapm_post, /* machine specific post widget - exec last */ snd_soc_dapm_supply, /* power/clock supply */ snd_soc_dapm_regulator_supply, /* external regulator */ + snd_soc_dapm_clock_supply, /* external clock */ snd_soc_dapm_aif_in, /* audio interface input */ snd_soc_dapm_aif_out, /* audio interface output */ snd_soc_dapm_siggen, /* signal generator */ @@ -537,6 +545,8 @@ struct snd_soc_dapm_widget { struct list_head dirty; int inputs; int outputs; + + struct clk *clk; }; struct snd_soc_dapm_update { -- cgit v1.2.3 From bc92657a11c0982783979bbb84ceaf58ba222124 Mon Sep 17 00:00:00 2001 From: Stephen Warren Date: Fri, 25 May 2012 18:22:11 -0600 Subject: ASoC: make snd_soc_dai_link more symmetrical Prior to this patch, the CPU side of a DAI link was specified using a single name. Often, this was the result of calling dev_name() on the device providing the DAI, but in the case of a CPU DAI driver that provided multiple DAIs, it needed to mix together both the device name and some device-relative name, in order to form a single globally unique name. However, the CODEC side of the DAI link was specified using separate fields for device (name or OF node) and device-relative DAI name. This patch allows the CPU side of a DAI link to be specified in the same way as the CODEC side, separating concepts of device and device-relative DAI name. I believe this will be important in multi-codec and/or dynamic PCM scenarios, where a single CPU driver provides multiple DAIs, while also booting using device tree, with accompanying desire not to hard-code the CPU side device's name into the original .cpu_dai_name field. Ideally, both the CPU DAI and CODEC DAI loops in soc_bind_dai_link() would now be identical. However, two things prevent that at present: 1) The need to save rtd->codec for the CODEC side, which means we have to search for the CODEC explicitly, and not just the CODEC side DAI. 2) Since we know the CODEC side DAI is part of a codec, and not just a standalone DAI, it's slightly more efficient to convert .codec_name/ .codec_of_node into a codec first, and then compare each DAI's .codec field, since this avoids strcmp() on each DAI's CODEC's name within the loop. However, the two loops are essentially semantically equivalent. Signed-off-by: Stephen Warren Signed-off-by: Mark Brown --- include/sound/soc.h | 33 ++++++++++++++++++++++++++++----- 1 file changed, 28 insertions(+), 5 deletions(-) (limited to 'include') diff --git a/include/sound/soc.h b/include/sound/soc.h index c703871f5f65..23c4efbe13a6 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -785,13 +785,36 @@ struct snd_soc_dai_link { /* config - must be set by machine driver */ const char *name; /* Codec name */ const char *stream_name; /* Stream name */ - const char *codec_name; /* for multi-codec */ - const struct device_node *codec_of_node; - const char *platform_name; /* for multi-platform */ - const struct device_node *platform_of_node; + /* + * You MAY specify the link's CPU-side device, either by device name, + * or by DT/OF node, but not both. If this information is omitted, + * the CPU-side DAI is matched using .cpu_dai_name only, which hence + * must be globally unique. These fields are currently typically used + * only for codec to codec links, or systems using device tree. + */ + const char *cpu_name; + const struct device_node *cpu_of_node; + /* + * You MAY specify the DAI name of the CPU DAI. If this information is + * omitted, the CPU-side DAI is matched using .cpu_name/.cpu_of_node + * only, which only works well when that device exposes a single DAI. + */ const char *cpu_dai_name; - const struct device_node *cpu_dai_of_node; + /* + * You MUST specify the link's codec, either by device name, or by + * DT/OF node, but not both. + */ + const char *codec_name; + const struct device_node *codec_of_node; + /* You MUST specify the DAI name within the codec */ const char *codec_dai_name; + /* + * You MAY specify the link's platform/PCM/DMA driver, either by + * device name, or by DT/OF node, but not both. Some forms of link + * do not need a platform. + */ + const char *platform_name; + const struct device_node *platform_of_node; int be_id; /* optional ID for machine driver BE identification */ const struct snd_soc_pcm_stream *params; -- cgit v1.2.3 From 6c9d8cf6372ed2995a3d982f5c1f966e842101cc Mon Sep 17 00:00:00 2001 From: Adam Thomson Date: Thu, 31 May 2012 15:18:01 +0100 Subject: ASoC: core: Add single controls with specified range of values Control type added for cases where a specific range of values within a register are required for control. Added convenience macros: SOC_SINGLE_RANGE SOC_SINGLE_RANGE_TLV Added accessor implementations: snd_soc_info_volsw_range snd_soc_put_volsw_range snd_soc_get_volsw_range Signed-off-by: Michal Hajduk Signed-off-by: Adam Thomson Signed-off-by: Mark Brown --- include/sound/soc.h | 23 +++++++++++++++++++++++ 1 file changed, 23 insertions(+) (limited to 'include') diff --git a/include/sound/soc.h b/include/sound/soc.h index 23c4efbe13a6..e4348d25fca3 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -47,6 +47,13 @@ .info = snd_soc_info_volsw, .get = snd_soc_get_volsw,\ .put = snd_soc_put_volsw, \ .private_value = SOC_SINGLE_VALUE(reg, shift, max, invert) } +#define SOC_SINGLE_RANGE(xname, xreg, xshift, xmin, xmax, xinvert) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname),\ + .info = snd_soc_info_volsw_range, .get = snd_soc_get_volsw_range, \ + .put = snd_soc_put_volsw_range, \ + .private_value = (unsigned long)&(struct soc_mixer_control) \ + {.reg = xreg, .shift = xshift, .min = xmin,\ + .max = xmax, .platform_max = xmax, .invert = xinvert} } #define SOC_SINGLE_TLV(xname, reg, shift, max, invert, tlv_array) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ |\ @@ -67,6 +74,16 @@ {.reg = xreg, .rreg = xreg, \ .shift = xshift, .rshift = xshift, \ .max = xmax, .min = xmin} } +#define SOC_SINGLE_RANGE_TLV(xname, xreg, xshift, xmin, xmax, xinvert, 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_range, \ + .get = snd_soc_get_volsw_range, .put = snd_soc_put_volsw_range, \ + .private_value = (unsigned long)&(struct soc_mixer_control) \ + {.reg = xreg, .shift = xshift, .min = xmin,\ + .max = xmax, .platform_max = xmax, .invert = xinvert} } #define SOC_DOUBLE(xname, reg, shift_left, shift_right, max, invert) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname),\ .info = snd_soc_info_volsw, .get = snd_soc_get_volsw, \ @@ -460,6 +477,12 @@ int snd_soc_get_volsw_s8(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol); int snd_soc_put_volsw_s8(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol); +int snd_soc_info_volsw_range(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo); +int snd_soc_put_volsw_range(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol); +int snd_soc_get_volsw_range(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol); int snd_soc_limit_volume(struct snd_soc_codec *codec, const char *name, int max); int snd_soc_bytes_info(struct snd_kcontrol *kcontrol, -- cgit v1.2.3 From f242e50eee1ec7692c4854d94e8cd543991cce71 Mon Sep 17 00:00:00 2001 From: Ola Lilja Date: Thu, 7 Jun 2012 14:00:46 +0200 Subject: mfd/ab8500: Move platform-data for ab8500-codec into mfd-driver The platform-data used by the Ux500 ASoC-driver is moved from the machine-driver context into the codec-driver context. This means adding the platform-data for 'ab8500-codec' into the main AB8500 platform-data. Signed-off-by: Ola Lilja Signed-off-by: Mark Brown --- include/linux/mfd/abx500/ab8500-codec.h | 52 +++++++++++++++++++++++++++++++++ include/linux/mfd/abx500/ab8500.h | 2 ++ 2 files changed, 54 insertions(+) create mode 100644 include/linux/mfd/abx500/ab8500-codec.h (limited to 'include') diff --git a/include/linux/mfd/abx500/ab8500-codec.h b/include/linux/mfd/abx500/ab8500-codec.h new file mode 100644 index 000000000000..dc6529202cdd --- /dev/null +++ b/include/linux/mfd/abx500/ab8500-codec.h @@ -0,0 +1,52 @@ +/* + * Copyright (C) ST-Ericsson SA 2012 + * + * Author: Ola Lilja + * for ST-Ericsson. + * + * License terms: + * + * 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 AB8500_CORE_CODEC_H +#define AB8500_CORE_CODEC_H + +/* Mic-types */ +enum amic_type { + AMIC_TYPE_SINGLE_ENDED, + AMIC_TYPE_DIFFERENTIAL +}; + +/* Mic-biases */ +enum amic_micbias { + AMIC_MICBIAS_VAMIC1, + AMIC_MICBIAS_VAMIC2 +}; + +/* Bias-voltage */ +enum ear_cm_voltage { + EAR_CMV_0_95V, + EAR_CMV_1_10V, + EAR_CMV_1_27V, + EAR_CMV_1_58V +}; + +/* Analog microphone settings */ +struct amic_settings { + enum amic_type mic1_type; + enum amic_type mic2_type; + enum amic_micbias mic1a_micbias; + enum amic_micbias mic1b_micbias; + enum amic_micbias mic2_micbias; +}; + +/* Platform data structure for the audio-parts of the AB8500 */ +struct ab8500_codec_platform_data { + struct amic_settings amics; + enum ear_cm_voltage ear_cmv; +}; + +#endif diff --git a/include/linux/mfd/abx500/ab8500.h b/include/linux/mfd/abx500/ab8500.h index 91dd3ef63e99..bc9b84b60ec6 100644 --- a/include/linux/mfd/abx500/ab8500.h +++ b/include/linux/mfd/abx500/ab8500.h @@ -266,6 +266,7 @@ struct ab8500 { struct regulator_reg_init; struct regulator_init_data; struct ab8500_gpio_platform_data; +struct ab8500_codec_platform_data; /** * struct ab8500_platform_data - AB8500 platform data @@ -284,6 +285,7 @@ struct ab8500_platform_data { int num_regulator; struct regulator_init_data *regulator; struct ab8500_gpio_platform_data *gpio; + struct ab8500_codec_platform_data *codec; }; extern int __devinit ab8500_init(struct ab8500 *ab8500, -- cgit v1.2.3 From 7a824e214e25a49442fe868dac0af8a904b24f58 Mon Sep 17 00:00:00 2001 From: Zhangfei Gao Date: Mon, 11 Jun 2012 18:04:38 +0800 Subject: ASoC: mmp: add audio dma support mmp-pcm handle audio dma based on soc-dmaengine Support mmp and pxa910 Signed-off-by: Zhangfei Gao Signed-off-by: Leo Yan Signed-off-by: Qiao Zhou Signed-off-by: Mark Brown --- include/linux/platform_data/mmp_audio.h | 22 ++++++++++++++++++++++ 1 file changed, 22 insertions(+) create mode 100644 include/linux/platform_data/mmp_audio.h (limited to 'include') diff --git a/include/linux/platform_data/mmp_audio.h b/include/linux/platform_data/mmp_audio.h new file mode 100644 index 000000000000..0f25d165abd6 --- /dev/null +++ b/include/linux/platform_data/mmp_audio.h @@ -0,0 +1,22 @@ +/* + * MMP Platform AUDIO Management + * + * Copyright (c) 2011 Marvell Semiconductors Inc. + * + * 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 MMP_AUDIO_H +#define MMP_AUDIO_H + +struct mmp_audio_platdata { + u32 period_max_capture; + u32 buffer_max_capture; + u32 period_max_playback; + u32 buffer_max_playback; +}; + +#endif /* MMP_AUDIO_H */ -- cgit v1.2.3 From 4be77a530be1ea62574f31c20dd9848e7e2ab0f6 Mon Sep 17 00:00:00 2001 From: Dimitris Papastamos Date: Fri, 15 Jun 2012 16:35:28 +0100 Subject: ALSA: pcm: Add snd_pcm_rate_bit_to_rate() This is essentially the reverse of snd_pcm_rate_to_rate_bit(). This is generally useful as the Compress API uses the rate bit directly and it helps to be able to map back to the actual sample rate. Signed-off-by: Dimitris Papastamos Signed-off-by: Takashi Iwai --- include/sound/pcm.h | 1 + 1 file changed, 1 insertion(+) (limited to 'include') diff --git a/include/sound/pcm.h b/include/sound/pcm.h index 0d1112815be3..68372bc1e11b 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -893,6 +893,7 @@ extern const struct snd_pcm_hw_constraint_list snd_pcm_known_rates; int snd_pcm_limit_hw_rates(struct snd_pcm_runtime *runtime); unsigned int snd_pcm_rate_to_rate_bit(unsigned int rate); +unsigned int snd_pcm_rate_bit_to_rate(unsigned int rate_bit); static inline void snd_pcm_set_runtime_buffer(struct snd_pcm_substream *substream, struct snd_dma_buffer *bufp) -- cgit v1.2.3 From 33eb3311f3ad4a14f2e55d36fdb0d3ec54712231 Mon Sep 17 00:00:00 2001 From: Ezequiel Garcia Date: Tue, 19 Jun 2012 16:20:24 -0300 Subject: sound: Remove unused include/linux/ac97_codec.h header This file has been superseded by include/sound/ac97_codec.h, and has currently no users. Cc: Ralf Baechle Cc: Jaroslav Kysela Cc: Clemens Ladisch Signed-off-by: Ezequiel Garcia Signed-off-by: Takashi Iwai --- include/linux/ac97_codec.h | 362 --------------------------------------------- 1 file changed, 362 deletions(-) delete mode 100644 include/linux/ac97_codec.h (limited to 'include') diff --git a/include/linux/ac97_codec.h b/include/linux/ac97_codec.h deleted file mode 100644 index 0260c3e79fdd..000000000000 --- a/include/linux/ac97_codec.h +++ /dev/null @@ -1,362 +0,0 @@ -#ifndef _AC97_CODEC_H_ -#define _AC97_CODEC_H_ - -#include -#include - -/* AC97 1.0 */ -#define AC97_RESET 0x0000 // -#define AC97_MASTER_VOL_STEREO 0x0002 // Line Out -#define AC97_HEADPHONE_VOL 0x0004 // -#define AC97_MASTER_VOL_MONO 0x0006 // TAD Output -#define AC97_MASTER_TONE 0x0008 // -#define AC97_PCBEEP_VOL 0x000a // none -#define AC97_PHONE_VOL 0x000c // TAD Input (mono) -#define AC97_MIC_VOL 0x000e // MIC Input (mono) -#define AC97_LINEIN_VOL 0x0010 // Line Input (stereo) -#define AC97_CD_VOL 0x0012 // CD Input (stereo) -#define AC97_VIDEO_VOL 0x0014 // none -#define AC97_AUX_VOL 0x0016 // Aux Input (stereo) -#define AC97_PCMOUT_VOL 0x0018 // Wave Output (stereo) -#define AC97_RECORD_SELECT 0x001a // -#define AC97_RECORD_GAIN 0x001c -#define AC97_RECORD_GAIN_MIC 0x001e -#define AC97_GENERAL_PURPOSE 0x0020 -#define AC97_3D_CONTROL 0x0022 -#define AC97_MODEM_RATE 0x0024 -#define AC97_POWER_CONTROL 0x0026 - -/* AC'97 2.0 */ -#define AC97_EXTENDED_ID 0x0028 /* Extended Audio ID */ -#define AC97_EXTENDED_STATUS 0x002A /* Extended Audio Status */ -#define AC97_PCM_FRONT_DAC_RATE 0x002C /* PCM Front DAC Rate */ -#define AC97_PCM_SURR_DAC_RATE 0x002E /* PCM Surround DAC Rate */ -#define AC97_PCM_LFE_DAC_RATE 0x0030 /* PCM LFE DAC Rate */ -#define AC97_PCM_LR_ADC_RATE 0x0032 /* PCM LR ADC Rate */ -#define AC97_PCM_MIC_ADC_RATE 0x0034 /* PCM MIC ADC Rate */ -#define AC97_CENTER_LFE_MASTER 0x0036 /* Center + LFE Master Volume */ -#define AC97_SURROUND_MASTER 0x0038 /* Surround (Rear) Master Volume */ -#define AC97_RESERVED_3A 0x003A /* Reserved in AC '97 < 2.2 */ - -/* AC'97 2.2 */ -#define AC97_SPDIF_CONTROL 0x003A /* S/PDIF Control */ - -/* range 0x3c-0x58 - MODEM */ -#define AC97_EXTENDED_MODEM_ID 0x003C -#define AC97_EXTEND_MODEM_STAT 0x003E -#define AC97_LINE1_RATE 0x0040 -#define AC97_LINE2_RATE 0x0042 -#define AC97_HANDSET_RATE 0x0044 -#define AC97_LINE1_LEVEL 0x0046 -#define AC97_LINE2_LEVEL 0x0048 -#define AC97_HANDSET_LEVEL 0x004A -#define AC97_GPIO_CONFIG 0x004C -#define AC97_GPIO_POLARITY 0x004E -#define AC97_GPIO_STICKY 0x0050 -#define AC97_GPIO_WAKE_UP 0x0052 -#define AC97_GPIO_STATUS 0x0054 -#define AC97_MISC_MODEM_STAT 0x0056 -#define AC97_RESERVED_58 0x0058 - -/* registers 0x005a - 0x007a are vendor reserved */ - -#define AC97_VENDOR_ID1 0x007c -#define AC97_VENDOR_ID2 0x007e - -/* volume control bit defines */ -#define AC97_MUTE 0x8000 -#define AC97_MICBOOST 0x0040 -#define AC97_LEFTVOL 0x3f00 -#define AC97_RIGHTVOL 0x003f - -/* record mux defines */ -#define AC97_RECMUX_MIC 0x0000 -#define AC97_RECMUX_CD 0x0101 -#define AC97_RECMUX_VIDEO 0x0202 -#define AC97_RECMUX_AUX 0x0303 -#define AC97_RECMUX_LINE 0x0404 -#define AC97_RECMUX_STEREO_MIX 0x0505 -#define AC97_RECMUX_MONO_MIX 0x0606 -#define AC97_RECMUX_PHONE 0x0707 - -/* general purpose register bit defines */ -#define AC97_GP_LPBK 0x0080 /* Loopback mode */ -#define AC97_GP_MS 0x0100 /* Mic Select 0=Mic1, 1=Mic2 */ -#define AC97_GP_MIX 0x0200 /* Mono output select 0=Mix, 1=Mic */ -#define AC97_GP_RLBK 0x0400 /* Remote Loopback - Modem line codec */ -#define AC97_GP_LLBK 0x0800 /* Local Loopback - Modem Line codec */ -#define AC97_GP_LD 0x1000 /* Loudness 1=on */ -#define AC97_GP_3D 0x2000 /* 3D Enhancement 1=on */ -#define AC97_GP_ST 0x4000 /* Stereo Enhancement 1=on */ -#define AC97_GP_POP 0x8000 /* Pcm Out Path, 0=pre 3D, 1=post 3D */ - -/* extended audio status and control bit defines */ -#define AC97_EA_VRA 0x0001 /* Variable bit rate enable bit */ -#define AC97_EA_DRA 0x0002 /* Double-rate audio enable bit */ -#define AC97_EA_SPDIF 0x0004 /* S/PDIF Enable bit */ -#define AC97_EA_VRM 0x0008 /* Variable bit rate for MIC enable bit */ -#define AC97_EA_CDAC 0x0040 /* PCM Center DAC is ready (Read only) */ -#define AC97_EA_SDAC 0x0040 /* PCM Surround DACs are ready (Read only) */ -#define AC97_EA_LDAC 0x0080 /* PCM LFE DAC is ready (Read only) */ -#define AC97_EA_MDAC 0x0100 /* MIC ADC is ready (Read only) */ -#define AC97_EA_SPCV 0x0400 /* S/PDIF configuration valid (Read only) */ -#define AC97_EA_PRI 0x0800 /* Turns the PCM Center DAC off */ -#define AC97_EA_PRJ 0x1000 /* Turns the PCM Surround DACs off */ -#define AC97_EA_PRK 0x2000 /* Turns the PCM LFE DAC off */ -#define AC97_EA_PRL 0x4000 /* Turns the MIC ADC off */ -#define AC97_EA_SLOT_MASK 0xffcf /* Mask for slot assignment bits */ -#define AC97_EA_SPSA_3_4 0x0000 /* Slot assigned to 3 & 4 */ -#define AC97_EA_SPSA_7_8 0x0010 /* Slot assigned to 7 & 8 */ -#define AC97_EA_SPSA_6_9 0x0020 /* Slot assigned to 6 & 9 */ -#define AC97_EA_SPSA_10_11 0x0030 /* Slot assigned to 10 & 11 */ - -/* S/PDIF control bit defines */ -#define AC97_SC_PRO 0x0001 /* Professional status */ -#define AC97_SC_NAUDIO 0x0002 /* Non audio stream */ -#define AC97_SC_COPY 0x0004 /* Copyright status */ -#define AC97_SC_PRE 0x0008 /* Preemphasis status */ -#define AC97_SC_CC_MASK 0x07f0 /* Category Code mask */ -#define AC97_SC_L 0x0800 /* Generation Level status */ -#define AC97_SC_SPSR_MASK 0xcfff /* S/PDIF Sample Rate bits */ -#define AC97_SC_SPSR_44K 0x0000 /* Use 44.1kHz Sample rate */ -#define AC97_SC_SPSR_48K 0x2000 /* Use 48kHz Sample rate */ -#define AC97_SC_SPSR_32K 0x3000 /* Use 32kHz Sample rate */ -#define AC97_SC_DRS 0x4000 /* Double Rate S/PDIF */ -#define AC97_SC_V 0x8000 /* Validity status */ - -/* powerdown control and status bit defines */ - -/* status */ -#define AC97_PWR_MDM 0x0010 /* Modem section ready */ -#define AC97_PWR_REF 0x0008 /* Vref nominal */ -#define AC97_PWR_ANL 0x0004 /* Analog section ready */ -#define AC97_PWR_DAC 0x0002 /* DAC section ready */ -#define AC97_PWR_ADC 0x0001 /* ADC section ready */ - -/* control */ -#define AC97_PWR_PR0 0x0100 /* ADC and Mux powerdown */ -#define AC97_PWR_PR1 0x0200 /* DAC powerdown */ -#define AC97_PWR_PR2 0x0400 /* Output mixer powerdown (Vref on) */ -#define AC97_PWR_PR3 0x0800 /* Output mixer powerdown (Vref off) */ -#define AC97_PWR_PR4 0x1000 /* AC-link powerdown */ -#define AC97_PWR_PR5 0x2000 /* Internal Clk disable */ -#define AC97_PWR_PR6 0x4000 /* HP amp powerdown */ -#define AC97_PWR_PR7 0x8000 /* Modem off - if supported */ - -/* extended audio ID register bit defines */ -#define AC97_EXTID_VRA 0x0001 -#define AC97_EXTID_DRA 0x0002 -#define AC97_EXTID_SPDIF 0x0004 -#define AC97_EXTID_VRM 0x0008 -#define AC97_EXTID_DSA0 0x0010 -#define AC97_EXTID_DSA1 0x0020 -#define AC97_EXTID_CDAC 0x0040 -#define AC97_EXTID_SDAC 0x0080 -#define AC97_EXTID_LDAC 0x0100 -#define AC97_EXTID_AMAP 0x0200 -#define AC97_EXTID_REV0 0x0400 -#define AC97_EXTID_REV1 0x0800 -#define AC97_EXTID_ID0 0x4000 -#define AC97_EXTID_ID1 0x8000 - -/* extended status register bit defines */ -#define AC97_EXTSTAT_VRA 0x0001 -#define AC97_EXTSTAT_DRA 0x0002 -#define AC97_EXTSTAT_SPDIF 0x0004 -#define AC97_EXTSTAT_VRM 0x0008 -#define AC97_EXTSTAT_SPSA0 0x0010 -#define AC97_EXTSTAT_SPSA1 0x0020 -#define AC97_EXTSTAT_CDAC 0x0040 -#define AC97_EXTSTAT_SDAC 0x0080 -#define AC97_EXTSTAT_LDAC 0x0100 -#define AC97_EXTSTAT_MADC 0x0200 -#define AC97_EXTSTAT_SPCV 0x0400 -#define AC97_EXTSTAT_PRI 0x0800 -#define AC97_EXTSTAT_PRJ 0x1000 -#define AC97_EXTSTAT_PRK 0x2000 -#define AC97_EXTSTAT_PRL 0x4000 - -/* extended audio ID register bit defines */ -#define AC97_EXTID_VRA 0x0001 -#define AC97_EXTID_DRA 0x0002 -#define AC97_EXTID_SPDIF 0x0004 -#define AC97_EXTID_VRM 0x0008 -#define AC97_EXTID_DSA0 0x0010 -#define AC97_EXTID_DSA1 0x0020 -#define AC97_EXTID_CDAC 0x0040 -#define AC97_EXTID_SDAC 0x0080 -#define AC97_EXTID_LDAC 0x0100 -#define AC97_EXTID_AMAP 0x0200 -#define AC97_EXTID_REV0 0x0400 -#define AC97_EXTID_REV1 0x0800 -#define AC97_EXTID_ID0 0x4000 -#define AC97_EXTID_ID1 0x8000 - -/* extended status register bit defines */ -#define AC97_EXTSTAT_VRA 0x0001 -#define AC97_EXTSTAT_DRA 0x0002 -#define AC97_EXTSTAT_SPDIF 0x0004 -#define AC97_EXTSTAT_VRM 0x0008 -#define AC97_EXTSTAT_SPSA0 0x0010 -#define AC97_EXTSTAT_SPSA1 0x0020 -#define AC97_EXTSTAT_CDAC 0x0040 -#define AC97_EXTSTAT_SDAC 0x0080 -#define AC97_EXTSTAT_LDAC 0x0100 -#define AC97_EXTSTAT_MADC 0x0200 -#define AC97_EXTSTAT_SPCV 0x0400 -#define AC97_EXTSTAT_PRI 0x0800 -#define AC97_EXTSTAT_PRJ 0x1000 -#define AC97_EXTSTAT_PRK 0x2000 -#define AC97_EXTSTAT_PRL 0x4000 - -/* useful power states */ -#define AC97_PWR_D0 0x0000 /* everything on */ -#define AC97_PWR_D1 AC97_PWR_PR0|AC97_PWR_PR1|AC97_PWR_PR4 -#define AC97_PWR_D2 AC97_PWR_PR0|AC97_PWR_PR1|AC97_PWR_PR2|AC97_PWR_PR3|AC97_PWR_PR4 -#define AC97_PWR_D3 AC97_PWR_PR0|AC97_PWR_PR1|AC97_PWR_PR2|AC97_PWR_PR3|AC97_PWR_PR4 -#define AC97_PWR_ANLOFF AC97_PWR_PR2|AC97_PWR_PR3 /* analog section off */ - -/* Total number of defined registers. */ -#define AC97_REG_CNT 64 - - -/* OSS interface to the ac97s.. */ -#define AC97_STEREO_MASK (SOUND_MASK_VOLUME|SOUND_MASK_PCM|\ - SOUND_MASK_LINE|SOUND_MASK_CD|\ - SOUND_MASK_ALTPCM|SOUND_MASK_IGAIN|\ - SOUND_MASK_LINE1|SOUND_MASK_VIDEO) - -#define AC97_SUPPORTED_MASK (AC97_STEREO_MASK | \ - SOUND_MASK_BASS|SOUND_MASK_TREBLE|\ - SOUND_MASK_SPEAKER|SOUND_MASK_MIC|\ - SOUND_MASK_PHONEIN|SOUND_MASK_PHONEOUT) - -#define AC97_RECORD_MASK (SOUND_MASK_MIC|\ - SOUND_MASK_CD|SOUND_MASK_IGAIN|SOUND_MASK_VIDEO|\ - SOUND_MASK_LINE1| SOUND_MASK_LINE|\ - SOUND_MASK_PHONEIN) - -/* original check is not good enough in case FOO is greater than - * SOUND_MIXER_NRDEVICES because the supported_mixers has exactly - * SOUND_MIXER_NRDEVICES elements. - * before matching the given mixer against the bitmask in supported_mixers we - * check if mixer number exceeds maximum allowed size which is as mentioned - * above SOUND_MIXER_NRDEVICES */ -#define supported_mixer(CODEC,FOO) ((FOO >= 0) && \ - (FOO < SOUND_MIXER_NRDEVICES) && \ - (CODEC)->supported_mixers & (1< Date: Tue, 19 Jun 2012 19:31:48 +0100 Subject: ALSA: Add missing include of pcm.h to pcm_params.h There's a dependency but no #include. Signed-off-by: Mark Brown Signed-off-by: Takashi Iwai --- include/sound/pcm_params.h | 2 ++ 1 file changed, 2 insertions(+) (limited to 'include') diff --git a/include/sound/pcm_params.h b/include/sound/pcm_params.h index f494f1e3c900..37ae12e0ab06 100644 --- a/include/sound/pcm_params.h +++ b/include/sound/pcm_params.h @@ -22,6 +22,8 @@ * */ +#include + int snd_pcm_hw_param_first(struct snd_pcm_substream *pcm, struct snd_pcm_hw_params *params, snd_pcm_hw_param_t var, int *dir); -- cgit v1.2.3 From c32c44cb58d212513243744878423abd207bc8a8 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Mon, 11 Jun 2012 20:11:40 +0200 Subject: dmaengine: Add wrapper for device_tx_status callback This patch adds a small inline wrapper for the devivce_tx_status callback of a dma device. This makes the source code of users of this function a bit more compact and a bit more legible. E.g.: -status = chan->device->device_tx_status(chan, cookie, &state) +status = dmaengine_tx_status(chan, cookie, &state) Signed-off-by: Lars-Peter Clausen Acked-by Vinod Koul Signed-off-by: Mark Brown --- include/linux/dmaengine.h | 6 ++++++ 1 file changed, 6 insertions(+) (limited to 'include') diff --git a/include/linux/dmaengine.h b/include/linux/dmaengine.h index 56377df39124..cc0756a35ae3 100644 --- a/include/linux/dmaengine.h +++ b/include/linux/dmaengine.h @@ -670,6 +670,12 @@ static inline int dmaengine_resume(struct dma_chan *chan) return dmaengine_device_control(chan, DMA_RESUME, 0); } +static inline enum dma_status dmaengine_tx_status(struct dma_chan *chan, + dma_cookie_t cookie, struct dma_tx_state *state) +{ + return chan->device->device_tx_status(chan, cookie, state); +} + static inline dma_cookie_t dmaengine_submit(struct dma_async_tx_descriptor *desc) { return desc->tx_submit(desc); -- cgit v1.2.3 From 9883ab229d61b884323f9186b1bd4a41373a491b Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Mon, 11 Jun 2012 20:11:41 +0200 Subject: ASoC: dmaengine-pcm: Rename and deprecate snd_dmaengine_pcm_pointer Currently the sound dmaengine pcm helper functions implement the pcm_pointer callback by trying to count the number of elapsed periods. This is done by advancing the stream position in the dmaengine callback by one period. Unfortunately there is no guarantee that the callback will be called for each elapsed period. It may be possible that under high system load it is only called once for multiple elapsed periods. This patch renames the current implementation and documents its shortcomings and that it should not be used anymore in new drivers. The next patch will introduce a new snd_dmaengine_pcm_pointer which will be implemented based on querying the current stream position from the dma device. Signed-off-by: Lars-Peter Clausen Acked-by Vinod Koul Acked-by: Dong Aisheng --- include/sound/dmaengine_pcm.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'include') diff --git a/include/sound/dmaengine_pcm.h b/include/sound/dmaengine_pcm.h index a8fcaa6d531f..ea5791583fed 100644 --- a/include/sound/dmaengine_pcm.h +++ b/include/sound/dmaengine_pcm.h @@ -38,7 +38,7 @@ void *snd_dmaengine_pcm_get_data(struct snd_pcm_substream *substream); int snd_hwparams_to_dma_slave_config(const struct snd_pcm_substream *substream, const struct snd_pcm_hw_params *params, struct dma_slave_config *slave_config); int snd_dmaengine_pcm_trigger(struct snd_pcm_substream *substream, int cmd); -snd_pcm_uframes_t snd_dmaengine_pcm_pointer(struct snd_pcm_substream *substream); +snd_pcm_uframes_t snd_dmaengine_pcm_pointer_no_residue(struct snd_pcm_substream *substream); int snd_dmaengine_pcm_open(struct snd_pcm_substream *substream, dma_filter_fn filter_fn, void *filter_data); -- cgit v1.2.3 From 3528f27a5d4ac299e2d8cbe7297c1e9edd601ee6 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Mon, 11 Jun 2012 20:11:42 +0200 Subject: ASoC: dmaengine-pcm: Add support for querying stream position from DMA driver Currently the sound dmaengine pcm helper functions implement the pcm_pointer callback by trying to count the number of elapsed periods. This is done by advancing the stream position in the dmaengine callback by one period. Unfortunately there is no guarantee that the callback will be called for each elapsed period. It may be possible that under high system load it is only called once for multiple elapsed periods. This patch addresses the issue by implementing support for querying the current stream position directly from the dmaengine driver. Since not all dmaengine drivers support reporting the stream position yet the old period counting implementation is kept for now. Furthermore the new mechanism allows to report the stream position with a sub-period granularity, given that the dmaengine driver supports this. Signed-off-by: Lars-Peter Clausen Acked-by: Vinod Koul Signed-off-by: Mark Brown --- include/sound/dmaengine_pcm.h | 1 + 1 file changed, 1 insertion(+) (limited to 'include') diff --git a/include/sound/dmaengine_pcm.h b/include/sound/dmaengine_pcm.h index ea5791583fed..b877334bbb0f 100644 --- a/include/sound/dmaengine_pcm.h +++ b/include/sound/dmaengine_pcm.h @@ -38,6 +38,7 @@ void *snd_dmaengine_pcm_get_data(struct snd_pcm_substream *substream); int snd_hwparams_to_dma_slave_config(const struct snd_pcm_substream *substream, const struct snd_pcm_hw_params *params, struct dma_slave_config *slave_config); int snd_dmaengine_pcm_trigger(struct snd_pcm_substream *substream, int cmd); +snd_pcm_uframes_t snd_dmaengine_pcm_pointer(struct snd_pcm_substream *substream); snd_pcm_uframes_t snd_dmaengine_pcm_pointer_no_residue(struct snd_pcm_substream *substream); int snd_dmaengine_pcm_open(struct snd_pcm_substream *substream, -- cgit v1.2.3 From 3a9cf8efd7b64f26f1e0f02afb70382f90cc11ca Mon Sep 17 00:00:00 2001 From: Rajeev Kumar Date: Thu, 21 Jun 2012 15:54:51 +0530 Subject: ASoC: Add support for synopsys i2s controller as per ASoC framework. This patch add support for synopsys I2S controller as per the ASoC framework. Signed-off-by: Rajeev Kumar Signed-off-by: Mark Brown --- include/sound/designware_i2s.h | 69 ++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 69 insertions(+) create mode 100644 include/sound/designware_i2s.h (limited to 'include') diff --git a/include/sound/designware_i2s.h b/include/sound/designware_i2s.h new file mode 100644 index 000000000000..26f406e0f673 --- /dev/null +++ b/include/sound/designware_i2s.h @@ -0,0 +1,69 @@ +/* + * Copyright (ST) 2012 Rajeev Kumar (rajeev-dlh.kumar@st.com) + * + * 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. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + * + */ + +#ifndef __SOUND_DESIGNWARE_I2S_H +#define __SOUND_DESIGNWARE_I2S_H + +#include +#include + +/* + * struct i2s_clk_config_data - represent i2s clk configuration data + * @chan_nr: number of channel + * @data_width: number of bits per sample (8/16/24/32 bit) + * @sample_rate: sampling frequency (8Khz, 16Khz, 32Khz, 44Khz, 48Khz) + */ +struct i2s_clk_config_data { + int chan_nr; + u32 data_width; + u32 sample_rate; +}; + +struct i2s_platform_data { + #define DWC_I2S_PLAY (1 << 0) + #define DWC_I2S_RECORD (1 << 1) + unsigned int cap; + int channel; + u32 snd_fmts; + u32 snd_rates; + + void *play_dma_data; + void *capture_dma_data; + bool (*filter)(struct dma_chan *chan, void *slave); + int (*i2s_clk_cfg)(struct i2s_clk_config_data *config); +}; + +struct i2s_dma_data { + void *data; + dma_addr_t addr; + u32 max_burst; + enum dma_slave_buswidth addr_width; + bool (*filter)(struct dma_chan *chan, void *slave); +}; + +/* I2S DMA registers */ +#define I2S_RXDMA 0x01C0 +#define I2S_TXDMA 0x01C8 + +#define TWO_CHANNEL_SUPPORT 2 /* up to 2.0 */ +#define FOUR_CHANNEL_SUPPORT 4 /* up to 3.1 */ +#define SIX_CHANNEL_SUPPORT 6 /* up to 5.1 */ +#define EIGHT_CHANNEL_SUPPORT 8 /* up to 7.1 */ + +#endif /* __SOUND_DESIGNWARE_I2S_H */ -- cgit v1.2.3 From 241b446f30de171b627524c107ce03e5ecee0124 Mon Sep 17 00:00:00 2001 From: Rajeev Kumar Date: Thu, 21 Jun 2012 15:54:52 +0530 Subject: ASoC: Add support for SPEAr ASoC pcm layer. This patch add support for the SPEAr ASoC pcm layer in ASoC framework. The pcm layer uses common snd_dmaengine framework. Signed-off-by: Rajeev Kumar Signed-off-by: Mark Brown --- include/sound/spear_dma.h | 35 +++++++++++++++++++++++++++++++++++ 1 file changed, 35 insertions(+) create mode 100644 include/sound/spear_dma.h (limited to 'include') diff --git a/include/sound/spear_dma.h b/include/sound/spear_dma.h new file mode 100644 index 000000000000..1b365bfdfb37 --- /dev/null +++ b/include/sound/spear_dma.h @@ -0,0 +1,35 @@ +/* +* linux/spear_dma.h +* +* Copyright (ST) 2012 Rajeev Kumar (rajeev-dlh.kumar@st.com) +* +* 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. +* +* You should have received a copy of the GNU General Public License +* along with this program; if not, write to the Free Software +* Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA +* +*/ + +#ifndef SPEAR_DMA_H +#define SPEAR_DMA_H + +#include + +struct spear_dma_data { + void *data; + dma_addr_t addr; + u32 max_burst; + enum dma_slave_buswidth addr_width; + bool (*filter)(struct dma_chan *chan, void *slave); +}; + +#endif /* SPEAR_DMA_H */ -- cgit v1.2.3 From ace36d85809f6005b559802f194d44c6aa61af88 Mon Sep 17 00:00:00 2001 From: Vipin Kumar Date: Thu, 21 Jun 2012 15:54:53 +0530 Subject: ASoC: SPEAr spdif_in: Add spdif IN support This patch implements the spdif IN driver for ST peripheral Signed-off-by: Vipin Kumar Signed-off-by: Rajeev Kumar Signed-off-by: Mark Brown --- include/sound/spear_spdif.h | 29 +++++++++++++++++++++++++++++ 1 file changed, 29 insertions(+) create mode 100644 include/sound/spear_spdif.h (limited to 'include') diff --git a/include/sound/spear_spdif.h b/include/sound/spear_spdif.h new file mode 100644 index 000000000000..a12f39695610 --- /dev/null +++ b/include/sound/spear_spdif.h @@ -0,0 +1,29 @@ +/* + * Copyright (ST) 2012 Vipin Kumar (vipin.kumar@st.com) + * + * 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. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + */ + +#ifndef __SOUND_SPDIF_H +#define __SOUND_SPDIF_H + +struct spear_spdif_platform_data { + /* DMA params */ + void *dma_params; + bool (*filter)(struct dma_chan *chan, void *slave); + void (*reset_perip)(void); +}; + +#endif /* SOUND_SPDIF_H */ -- cgit v1.2.3 From 229e3fdc1ba49b747e9434b55b3f6bd68a4db251 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 22 Jun 2012 11:40:55 +0100 Subject: ASoC: core: Add DOUBLE_R variants of the _RANGE controls The code handles this fine already, we just need new macros in the header for drivers to create the controls. Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- include/sound/soc.h | 21 +++++++++++++++++++++ 1 file changed, 21 insertions(+) (limited to 'include') diff --git a/include/sound/soc.h b/include/sound/soc.h index e4348d25fca3..e063380f63a2 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -42,6 +42,10 @@ ((unsigned long)&(struct soc_mixer_control) \ {.reg = xlreg, .rreg = xrreg, .shift = xshift, .rshift = xshift, \ .max = xmax, .platform_max = xmax, .invert = xinvert}) +#define SOC_DOUBLE_R_RANGE_VALUE(xlreg, xrreg, xshift, xmin, xmax, xinvert) \ + ((unsigned long)&(struct soc_mixer_control) \ + {.reg = xlreg, .rreg = xrreg, .shift = xshift, .rshift = xshift, \ + .min = xmin, .max = xmax, .platform_max = xmax, .invert = xinvert}) #define SOC_SINGLE(xname, reg, shift, max, invert) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ .info = snd_soc_info_volsw, .get = snd_soc_get_volsw,\ @@ -96,6 +100,13 @@ .get = snd_soc_get_volsw, .put = snd_soc_put_volsw, \ .private_value = SOC_DOUBLE_R_VALUE(reg_left, reg_right, xshift, \ xmax, xinvert) } +#define SOC_DOUBLE_R_RANGE(xname, reg_left, reg_right, xshift, xmin, \ + xmax, xinvert) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname),\ + .info = snd_soc_info_volsw_range, \ + .get = snd_soc_get_volsw_range, .put = snd_soc_put_volsw_range, \ + .private_value = SOC_DOUBLE_R_RANGE_VALUE(reg_left, reg_right, \ + xshift, xmin, xmax, xinvert) } #define SOC_DOUBLE_TLV(xname, reg, shift_left, shift_right, max, invert, tlv_array) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname),\ .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ |\ @@ -114,6 +125,16 @@ .get = snd_soc_get_volsw, .put = snd_soc_put_volsw, \ .private_value = SOC_DOUBLE_R_VALUE(reg_left, reg_right, xshift, \ xmax, xinvert) } +#define SOC_DOUBLE_R_RANGE_TLV(xname, reg_left, reg_right, xshift, xmin, \ + xmax, xinvert, 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_range, \ + .get = snd_soc_get_volsw_range, .put = snd_soc_put_volsw_range, \ + .private_value = SOC_DOUBLE_R_RANGE_VALUE(reg_left, reg_right, \ + xshift, xmin, xmax, xinvert) } #define SOC_DOUBLE_R_SX_TLV(xname, xreg, xrreg, xshift, xmin, xmax, tlv_array) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), \ .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ | \ -- cgit v1.2.3 From 68cb2b559278858ef9f3a7722f95df88797c7840 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 2 Jul 2012 15:20:37 +0200 Subject: ALSA: Convert to new pm_ops for PCI drivers Straightforward conversion to the new pm_ops from the legacy suspend/resume ops. Since we change vx222, vx_core and vxpocket have to be converted, too. Signed-off-by: Takashi Iwai --- include/sound/cs46xx.h | 3 +-- include/sound/trident.h | 3 +-- include/sound/vx_core.h | 2 +- include/sound/ymfpci.h | 3 +-- 4 files changed, 4 insertions(+), 7 deletions(-) (limited to 'include') diff --git a/include/sound/cs46xx.h b/include/sound/cs46xx.h index e3005a674a24..34a2dd1614fa 100644 --- a/include/sound/cs46xx.h +++ b/include/sound/cs46xx.h @@ -1730,8 +1730,7 @@ int snd_cs46xx_create(struct snd_card *card, struct pci_dev *pci, int external_amp, int thinkpad, struct snd_cs46xx **rcodec); -int snd_cs46xx_suspend(struct pci_dev *pci, pm_message_t state); -int snd_cs46xx_resume(struct pci_dev *pci); +extern const struct dev_pm_ops snd_cs46xx_pm; int snd_cs46xx_pcm(struct snd_cs46xx *chip, int device, struct snd_pcm **rpcm); int snd_cs46xx_pcm_rear(struct snd_cs46xx *chip, int device, struct snd_pcm **rpcm); diff --git a/include/sound/trident.h b/include/sound/trident.h index 9f191a0a1e19..06f0478103db 100644 --- a/include/sound/trident.h +++ b/include/sound/trident.h @@ -430,8 +430,7 @@ void snd_trident_free_voice(struct snd_trident * trident, struct snd_trident_voi void snd_trident_start_voice(struct snd_trident * trident, unsigned int voice); void snd_trident_stop_voice(struct snd_trident * trident, unsigned int voice); void snd_trident_write_voice_regs(struct snd_trident * trident, struct snd_trident_voice *voice); -int snd_trident_suspend(struct pci_dev *pci, pm_message_t state); -int snd_trident_resume(struct pci_dev *pci); +extern const struct dev_pm_ops snd_trident_pm; /* TLB memory allocation */ struct snd_util_memblk *snd_trident_alloc_pages(struct snd_trident *trident, diff --git a/include/sound/vx_core.h b/include/sound/vx_core.h index 5456343ebe4c..4f67c762cd74 100644 --- a/include/sound/vx_core.h +++ b/include/sound/vx_core.h @@ -341,7 +341,7 @@ int vx_change_frequency(struct vx_core *chip); /* * PM */ -int snd_vx_suspend(struct vx_core *card, pm_message_t state); +int snd_vx_suspend(struct vx_core *card); int snd_vx_resume(struct vx_core *card); /* diff --git a/include/sound/ymfpci.h b/include/sound/ymfpci.h index 41199664666b..238f118de6e1 100644 --- a/include/sound/ymfpci.h +++ b/include/sound/ymfpci.h @@ -377,8 +377,7 @@ int snd_ymfpci_create(struct snd_card *card, struct snd_ymfpci ** rcodec); void snd_ymfpci_free_gameport(struct snd_ymfpci *chip); -int snd_ymfpci_suspend(struct pci_dev *pci, pm_message_t state); -int snd_ymfpci_resume(struct pci_dev *pci); +extern const struct dev_pm_ops snd_ymfpci_pm; int snd_ymfpci_pcm(struct snd_ymfpci *chip, int device, struct snd_pcm **rpcm); int snd_ymfpci_pcm2(struct snd_ymfpci *chip, int device, struct snd_pcm **rpcm); -- cgit v1.2.3 From 81fcb170852d58d7ebd8101a8ef970c82056426e Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 2 Jul 2012 16:37:05 +0200 Subject: ALSA: Move some headers to local directories from include/sound This is a bit clean up of public sound header directory. Some header files in include/sound aren't really necessary to be located there but can be moved to their local directories gracefully. Signed-off-by: Takashi Iwai --- include/sound/cs46xx.h | 1744 --------------------------------- include/sound/cs46xx_dsp_scb_types.h | 1213 ----------------------- include/sound/cs46xx_dsp_spos.h | 234 ----- include/sound/cs46xx_dsp_task_types.h | 252 ----- include/sound/trident.h | 444 --------- include/sound/ymfpci.h | 389 -------- 6 files changed, 4276 deletions(-) delete mode 100644 include/sound/cs46xx.h delete mode 100644 include/sound/cs46xx_dsp_scb_types.h delete mode 100644 include/sound/cs46xx_dsp_spos.h delete mode 100644 include/sound/cs46xx_dsp_task_types.h delete mode 100644 include/sound/trident.h delete mode 100644 include/sound/ymfpci.h (limited to 'include') diff --git a/include/sound/cs46xx.h b/include/sound/cs46xx.h deleted file mode 100644 index 34a2dd1614fa..000000000000 --- a/include/sound/cs46xx.h +++ /dev/null @@ -1,1744 +0,0 @@ -#ifndef __SOUND_CS46XX_H -#define __SOUND_CS46XX_H - -/* - * Copyright (c) by Jaroslav Kysela , - * Cirrus Logic, Inc. - * Definitions for Cirrus Logic CS46xx chips - * - * - * 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. - * - * You should have received a copy of the GNU General Public License - * along with this program; if not, write to the Free Software - * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA - * - */ - -#include "pcm.h" -#include "pcm-indirect.h" -#include "rawmidi.h" -#include "ac97_codec.h" -#include "cs46xx_dsp_spos.h" - -/* - * Direct registers - */ - -/* - * The following define the offsets of the registers accessed via base address - * register zero on the CS46xx part. - */ -#define BA0_HISR 0x00000000 -#define BA0_HSR0 0x00000004 -#define BA0_HICR 0x00000008 -#define BA0_DMSR 0x00000100 -#define BA0_HSAR 0x00000110 -#define BA0_HDAR 0x00000114 -#define BA0_HDMR 0x00000118 -#define BA0_HDCR 0x0000011C -#define BA0_PFMC 0x00000200 -#define BA0_PFCV1 0x00000204 -#define BA0_PFCV2 0x00000208 -#define BA0_PCICFG00 0x00000300 -#define BA0_PCICFG04 0x00000304 -#define BA0_PCICFG08 0x00000308 -#define BA0_PCICFG0C 0x0000030C -#define BA0_PCICFG10 0x00000310 -#define BA0_PCICFG14 0x00000314 -#define BA0_PCICFG18 0x00000318 -#define BA0_PCICFG1C 0x0000031C -#define BA0_PCICFG20 0x00000320 -#define BA0_PCICFG24 0x00000324 -#define BA0_PCICFG28 0x00000328 -#define BA0_PCICFG2C 0x0000032C -#define BA0_PCICFG30 0x00000330 -#define BA0_PCICFG34 0x00000334 -#define BA0_PCICFG38 0x00000338 -#define BA0_PCICFG3C 0x0000033C -#define BA0_CLKCR1 0x00000400 -#define BA0_CLKCR2 0x00000404 -#define BA0_PLLM 0x00000408 -#define BA0_PLLCC 0x0000040C -#define BA0_FRR 0x00000410 -#define BA0_CFL1 0x00000414 -#define BA0_CFL2 0x00000418 -#define BA0_SERMC1 0x00000420 -#define BA0_SERMC2 0x00000424 -#define BA0_SERC1 0x00000428 -#define BA0_SERC2 0x0000042C -#define BA0_SERC3 0x00000430 -#define BA0_SERC4 0x00000434 -#define BA0_SERC5 0x00000438 -#define BA0_SERBSP 0x0000043C -#define BA0_SERBST 0x00000440 -#define BA0_SERBCM 0x00000444 -#define BA0_SERBAD 0x00000448 -#define BA0_SERBCF 0x0000044C -#define BA0_SERBWP 0x00000450 -#define BA0_SERBRP 0x00000454 -#ifndef NO_CS4612 -#define BA0_ASER_FADDR 0x00000458 -#endif -#define BA0_ACCTL 0x00000460 -#define BA0_ACSTS 0x00000464 -#define BA0_ACOSV 0x00000468 -#define BA0_ACCAD 0x0000046C -#define BA0_ACCDA 0x00000470 -#define BA0_ACISV 0x00000474 -#define BA0_ACSAD 0x00000478 -#define BA0_ACSDA 0x0000047C -#define BA0_JSPT 0x00000480 -#define BA0_JSCTL 0x00000484 -#define BA0_JSC1 0x00000488 -#define BA0_JSC2 0x0000048C -#define BA0_MIDCR 0x00000490 -#define BA0_MIDSR 0x00000494 -#define BA0_MIDWP 0x00000498 -#define BA0_MIDRP 0x0000049C -#define BA0_JSIO 0x000004A0 -#ifndef NO_CS4612 -#define BA0_ASER_MASTER 0x000004A4 -#endif -#define BA0_CFGI 0x000004B0 -#define BA0_SSVID 0x000004B4 -#define BA0_GPIOR 0x000004B8 -#ifndef NO_CS4612 -#define BA0_EGPIODR 0x000004BC -#define BA0_EGPIOPTR 0x000004C0 -#define BA0_EGPIOTR 0x000004C4 -#define BA0_EGPIOWR 0x000004C8 -#define BA0_EGPIOSR 0x000004CC -#define BA0_SERC6 0x000004D0 -#define BA0_SERC7 0x000004D4 -#define BA0_SERACC 0x000004D8 -#define BA0_ACCTL2 0x000004E0 -#define BA0_ACSTS2 0x000004E4 -#define BA0_ACOSV2 0x000004E8 -#define BA0_ACCAD2 0x000004EC -#define BA0_ACCDA2 0x000004F0 -#define BA0_ACISV2 0x000004F4 -#define BA0_ACSAD2 0x000004F8 -#define BA0_ACSDA2 0x000004FC -#define BA0_IOTAC0 0x00000500 -#define BA0_IOTAC1 0x00000504 -#define BA0_IOTAC2 0x00000508 -#define BA0_IOTAC3 0x0000050C -#define BA0_IOTAC4 0x00000510 -#define BA0_IOTAC5 0x00000514 -#define BA0_IOTAC6 0x00000518 -#define BA0_IOTAC7 0x0000051C -#define BA0_IOTAC8 0x00000520 -#define BA0_IOTAC9 0x00000524 -#define BA0_IOTAC10 0x00000528 -#define BA0_IOTAC11 0x0000052C -#define BA0_IOTFR0 0x00000540 -#define BA0_IOTFR1 0x00000544 -#define BA0_IOTFR2 0x00000548 -#define BA0_IOTFR3 0x0000054C -#define BA0_IOTFR4 0x00000550 -#define BA0_IOTFR5 0x00000554 -#define BA0_IOTFR6 0x00000558 -#define BA0_IOTFR7 0x0000055C -#define BA0_IOTFIFO 0x00000580 -#define BA0_IOTRRD 0x00000584 -#define BA0_IOTFP 0x00000588 -#define BA0_IOTCR 0x0000058C -#define BA0_DPCID 0x00000590 -#define BA0_DPCIA 0x00000594 -#define BA0_DPCIC 0x00000598 -#define BA0_PCPCIR 0x00000600 -#define BA0_PCPCIG 0x00000604 -#define BA0_PCPCIEN 0x00000608 -#define BA0_EPCIPMC 0x00000610 -#endif - -/* - * The following define the offsets of the registers and memories accessed via - * base address register one on the CS46xx part. - */ -#define BA1_SP_DMEM0 0x00000000 -#define BA1_SP_DMEM1 0x00010000 -#define BA1_SP_PMEM 0x00020000 -#define BA1_SP_REG 0x00030000 -#define BA1_SPCR 0x00030000 -#define BA1_DREG 0x00030004 -#define BA1_DSRWP 0x00030008 -#define BA1_TWPR 0x0003000C -#define BA1_SPWR 0x00030010 -#define BA1_SPIR 0x00030014 -#define BA1_FGR1 0x00030020 -#define BA1_SPCS 0x00030028 -#define BA1_SDSR 0x0003002C -#define BA1_FRMT 0x00030030 -#define BA1_FRCC 0x00030034 -#define BA1_FRSC 0x00030038 -#define BA1_OMNI_MEM 0x000E0000 - - -/* - * The following defines are for the flags in the host interrupt status - * register. - */ -#define HISR_VC_MASK 0x0000FFFF -#define HISR_VC0 0x00000001 -#define HISR_VC1 0x00000002 -#define HISR_VC2 0x00000004 -#define HISR_VC3 0x00000008 -#define HISR_VC4 0x00000010 -#define HISR_VC5 0x00000020 -#define HISR_VC6 0x00000040 -#define HISR_VC7 0x00000080 -#define HISR_VC8 0x00000100 -#define HISR_VC9 0x00000200 -#define HISR_VC10 0x00000400 -#define HISR_VC11 0x00000800 -#define HISR_VC12 0x00001000 -#define HISR_VC13 0x00002000 -#define HISR_VC14 0x00004000 -#define HISR_VC15 0x00008000 -#define HISR_INT0 0x00010000 -#define HISR_INT1 0x00020000 -#define HISR_DMAI 0x00040000 -#define HISR_FROVR 0x00080000 -#define HISR_MIDI 0x00100000 -#ifdef NO_CS4612 -#define HISR_RESERVED 0x0FE00000 -#else -#define HISR_SBINT 0x00200000 -#define HISR_RESERVED 0x0FC00000 -#endif -#define HISR_H0P 0x40000000 -#define HISR_INTENA 0x80000000 - -/* - * The following defines are for the flags in the host signal register 0. - */ -#define HSR0_VC_MASK 0xFFFFFFFF -#define HSR0_VC16 0x00000001 -#define HSR0_VC17 0x00000002 -#define HSR0_VC18 0x00000004 -#define HSR0_VC19 0x00000008 -#define HSR0_VC20 0x00000010 -#define HSR0_VC21 0x00000020 -#define HSR0_VC22 0x00000040 -#define HSR0_VC23 0x00000080 -#define HSR0_VC24 0x00000100 -#define HSR0_VC