March 2016 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH v2 0/7] topology: Add support for vendor tuples
by mengdong.lin＠linux.intel.com 05 Apr '16

05 Apr '16

From: Mengdong Lin <mengdong.lin(a)linux.intel.com> This series addes support for vendor tuples to topology, to avoid importing binary data blob from other files. Backward compatibility of ABI is not impacted. A kernel patch is also submitted "ASoC: topology: ABI - Define types for vendor tuples". The 1st patch is small code cleanup. The 2nd patch is a preparation, since tuples will need the type-specific free handler. History: v2: add check on string length, use strtol() to get hex value, and fix memory leak. Mengdong Lin (7): topology: Use the generic pointer to free an element's object topology: Define a free handler for the element topology: Add doc for vendor tuples topology: ABI - Define types for vendor tuples topology: Add support for vendor tokens topology: Add support for parsing vendor tuples topology: Build data objects with tuples include/sound/asoc.h | 42 +++- include/topology.h | 79 +++++++- src/topology/data.c | 496 +++++++++++++++++++++++++++++++++++++++++++++- src/topology/elem.c | 15 +- src/topology/parser.c | 24 +++ src/topology/tplg_local.h | 47 +++++ 6 files changed, 695 insertions(+), 8 deletions(-) -- 2.5.0

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[alsa-devel] Getting rid of inside_vm in intel8x0
by Andy Lutomirski 04 Apr '16

04 Apr '16

Would it be possible to revert: commit 228cf79376f13b98f2e1ac10586311312757675c Author: Konstantin Ozerkov <kozerkov(a)parallels.com> Date: Wed Oct 26 19:11:01 2011 +0400 ALSA: intel8x0: Improve performance in virtual environment Presumably one or more of the following is true: a) The inside_vm == true case is just an optimization and should apply unconditionally. b) The inside_vm == true case is incorrect and should be fixed or disabled. c) The inside_vm == true case is a special case that makes sense then IO is very very slow but doesn't make sense when IO is fast. If so, why not literally measure the time that the IO takes and switch over to the "inside VM" path when IO is slow? There are a pile of nonsensical "are we in a VM" checks of various sorts scattered throughout the kernel, they're all a mess to maintain (there are lots of kinds of VMs in the world, and Linux may not even know it's a guest), and, in most cases, it appears that the correct solution is to delete the checks. I just removed a nasty one in the x86_32 entry asm, and this one is written in C so it should be a piece of cake :) --Andy

5 11

[alsa-devel] [PATCH v2 0/2] ASoC: codecs: add support for TAS5720 digital amplifier
by Andreas Dannenberg 02 Apr '16

02 Apr '16

Driver for TI's TAS5720L/M digital audio amplifiers. The driver should be pretty standard except the optional interrupt-based fault reporting. Some background on the fault reporting since that might be a discussion of interest. The code should have that documented rather well but I wanted to bring it up right away. The TAS5720 devices can report conditions such as over-current (short), over temp, and others that can be important to know in a real-world system. However the device lacks a way to mask the generation of fault interrupts except by shutting it down, which can lead to cases where the device generates SAIF-fault host interrupts every 300us! This is clearly a lot of overhead to be thrown at the SoC (and my bench testing has demonstrated that) so after trying different things to tame this behavior I ended up implementing method to disable/enable the interrupts in the driver itself in combination with rate-limiting the processing of those events. The fact of processing interrupts in a threaded fashion (IRQF_ONESHOT) alone was not enough under certain conditions to prevent the SoC from not being able to get to all interrupts even with an empty threaded handler, and it would freeze and eventually the Kernel would disable the interrupt ("no one cared"). The scheme currently implemented seems to work reliably cycling different audio files over hours also forcing error conditions. But if anybody has a "cleaner" idea how to rate-limit IRQs on a very low level (currently done in the IRQ handler function) I'm absolutely open to suggestions. This all may not be an issue at all when using a faster/lower-latency SoC than what I'm using (BeagleBone Black) but the fact that IRQF_ONESHOT alone was not enough to prevent the system to be brought to its knees (again, even with an empty threaded handler) was a bit of a surprise to me. -- Andreas Dannenberg Texas Instruments Inc Changes since v1: - Simplified DT interrupt documentation (Thanks Rob Herring) - Fixed potential race condition during codec probe where deferred work that's used by the threaded IRQ handler was setup after the IRQ was initialized (would lead to an issue when the TAS5720 was already throwing interrupts at the time of probe) Andreas Dannenberg (2): ASoC: codecs: add TA5720 digital amplifier DT bindings ASoC: codecs: add support for TAS5720 digital amplifier .../devicetree/bindings/sound/tas5720.txt | 37 ++ sound/soc/codecs/Kconfig | 10 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/tas5720.c | 734 +++++++++++++++++++++ sound/soc/codecs/tas5720.h | 90 +++ 5 files changed, 873 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/tas5720.txt create mode 100644 sound/soc/codecs/tas5720.c create mode 100644 sound/soc/codecs/tas5720.h -- 2.6.4

3 8

[alsa-devel] [PATCH 1/2] ALSA: hda - Enable i915 binding for gen3/4 HDMI/DP
by Takashi Iwai 01 Apr '16

01 Apr '16

This patch fills the holes and now all i915 HDMI/DP codecs are managed over the audio ELD notifier, finally. The old gen3/gen4 chips have usually only a single pin/converter pair, and the digital port mapping is fixed. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/hda/hdac_i915.c | 11 +++++++++++ sound/pci/hda/patch_hdmi.c | 21 +++++++++++++++------ 2 files changed, 26 insertions(+), 6 deletions(-) diff --git a/sound/hda/hdac_i915.c b/sound/hda/hdac_i915.c index d0da2508823e..c62a9f830b84 100644 --- a/sound/hda/hdac_i915.c +++ b/sound/hda/hdac_i915.c @@ -128,6 +128,9 @@ EXPORT_SYMBOL_GPL(snd_hdac_get_display_clk); * Pin Widget 4 - PORT B (port = 1 in i915 driver) * Pin Widget 5 - PORT C (port = 2 in i915 driver) * Pin Widget 6 - PORT D (port = 3 in i915 driver) + * + * on earlier models: + * Pin Widget 3 - PORT B */ static int pin2port(struct hdac_device *codec, hda_nid_t pin_nid) { @@ -139,6 +142,14 @@ static int pin2port(struct hdac_device *codec, hda_nid_t pin_nid) case 0x80862882: /* VLV */ base_nid = 3; break; + case 0x80862801: /* Bearlake */ + case 0x80862802: /* Cantiga */ + case 0x80862803: /* Eaglelake */ + case 0x80862880: /* CedarTrail */ + case 0x808629fb: /* Crestline HDMI */ + if (WARN_ON(pin_nid != 0x03)) + return -1; + return 1; /* only port B */ default: base_nid = 4; break; diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 4833c7bdd1e8..2a3bf1584f3b 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -2286,6 +2286,15 @@ static void intel_pin_eld_notify(void *audio_ptr, int port) case 0x80862882: /* VLV */ pin_nid = port + 0x03; break; + case 0x80862801: /* Bearlake */ + case 0x80862802: /* Cantiga */ + case 0x80862803: /* Eaglelake */ + case 0x80862880: /* CedarTrail */ + case 0x808629fb: /* Crestline HDMI */ + if (WARN_ON(port != 1)) /* assume only port B */ + return; + pin_nid = 0x03; + break; default: pin_nid = port + 0x04; break; @@ -2424,7 +2433,7 @@ static int patch_i915_byt_hdmi(struct hda_codec *codec) return 0; } -/* Intel IronLake, SandyBridge and IvyBridge; with eld notifier */ +/* Intel Gen3/Gen4, IronLake, SandyBridge and IvyBridge; with eld notifier */ static int patch_i915_cpt_hdmi(struct hda_codec *codec) { struct hdmi_spec *spec; @@ -3627,9 +3636,9 @@ HDA_CODEC_ENTRY(0x11069f81, "VX900 HDMI/DP", patch_via_hdmi), HDA_CODEC_ENTRY(0x11069f84, "VX11 HDMI/DP", patch_generic_hdmi), HDA_CODEC_ENTRY(0x11069f85, "VX11 HDMI/DP", patch_generic_hdmi), HDA_CODEC_ENTRY(0x80860054, "IbexPeak HDMI", patch_i915_cpt_hdmi), -HDA_CODEC_ENTRY(0x80862801, "Bearlake HDMI", patch_generic_hdmi), -HDA_CODEC_ENTRY(0x80862802, "Cantiga HDMI", patch_generic_hdmi), -HDA_CODEC_ENTRY(0x80862803, "Eaglelake HDMI", patch_generic_hdmi), +HDA_CODEC_ENTRY(0x80862801, "Bearlake HDMI", patch_i915_cpt_hdmi), +HDA_CODEC_ENTRY(0x80862802, "Cantiga HDMI", patch_i915_cpt_hdmi), +HDA_CODEC_ENTRY(0x80862803, "Eaglelake HDMI", patch_i915_cpt_hdmi), HDA_CODEC_ENTRY(0x80862804, "IbexPeak HDMI", patch_i915_cpt_hdmi), HDA_CODEC_ENTRY(0x80862805, "CougarPoint HDMI", patch_i915_cpt_hdmi), HDA_CODEC_ENTRY(0x80862806, "PantherPoint HDMI", patch_i915_cpt_hdmi), @@ -3638,10 +3647,10 @@ HDA_CODEC_ENTRY(0x80862808, "Broadwell HDMI", patch_i915_hsw_hdmi), HDA_CODEC_ENTRY(0x80862809, "Skylake HDMI", patch_i915_hsw_hdmi), HDA_CODEC_ENTRY(0x8086280a, "Broxton HDMI", patch_i915_hsw_hdmi), HDA_CODEC_ENTRY(0x8086280b, "Kabylake HDMI", patch_i915_hsw_hdmi), -HDA_CODEC_ENTRY(0x80862880, "CedarTrail HDMI", patch_generic_hdmi), +HDA_CODEC_ENTRY(0x80862880, "CedarTrail HDMI", patch_i915_cpt_hdmi), HDA_CODEC_ENTRY(0x80862882, "Valleyview2 HDMI", patch_i915_byt_hdmi), HDA_CODEC_ENTRY(0x80862883, "Braswell HDMI", patch_i915_byt_hdmi), -HDA_CODEC_ENTRY(0x808629fb, "Crestline HDMI", patch_generic_hdmi), +HDA_CODEC_ENTRY(0x808629fb, "Crestline HDMI", patch_i915_cpt_hdmi), /* special ID for generic HDMI */ HDA_CODEC_ENTRY(HDA_CODEC_ID_GENERIC_HDMI, "Generic HDMI", patch_generic_hdmi), {} /* terminator */ -- 2.7.4

2 7

[alsa-devel] [PATCH] a52: Make compatible with ffmpeg 3.0
by Anatol Pomozov 01 Apr '16

01 Apr '16

audioconvert.h has been deprecated a while ago and got removed in ffmpeg 3.0 (commit 2d40968dd3ff17b12f7). Signed-off-by: Anatol Pomozov <anatol.pomozov(a)gmail.com> --- a52/pcm_a52.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/a52/pcm_a52.c b/a52/pcm_a52.c index 88529eb..6fad77d 100644 --- a/a52/pcm_a52.c +++ b/a52/pcm_a52.c @@ -39,7 +39,7 @@ #endif #if LIBAVCODEC_VERSION_INT >= AV_VERSION_INT(53, 34, 0) -#include <libavutil/audioconvert.h> +#include <libavutil/channel_layout.h> #include <libavutil/mem.h> #define USE_AVCODEC_FRAME #endif -- 2.7.1

3 3

[alsa-devel] [PATCH v3 1/5] ASoC: tas571x: added missing register literals
by Petr Kulhavy 01 Apr '16

01 Apr '16

The list of TAS571x registers was incomplete. Added the missing register definitions up to the register 0x25. Added volatile and read-only register tables into tas5711_regmap_config and tas5717_regmap_config. The chip has 256 registers in total. But from address 0x29 on (0x26 to 0x28 are reserved) the register width varies between 20, 12 and 8 bytes, which the register map is unable to represent. Signed-off-by: Petr Kulhavy <petr(a)barix.com> --- v1: <initial> v2: <no change> v3: added volatile and read-only register tables into regmap configs sound/soc/codecs/tas571x.c | 28 ++++++++++++++++++++++++++++ sound/soc/codecs/tas571x.h | 22 ++++++++++++++++++++++ 2 files changed, 50 insertions(+) diff --git a/sound/soc/codecs/tas571x.c b/sound/soc/codecs/tas571x.c index 39307ad..d255d65 100644 --- a/sound/soc/codecs/tas571x.c +++ b/sound/soc/codecs/tas571x.c @@ -57,6 +57,10 @@ static int tas571x_register_size(struct tas571x_private *priv, unsigned int reg) case TAS571X_CH1_VOL_REG: case TAS571X_CH2_VOL_REG: return priv->chip->vol_reg_size; + case TAS571X_INPUT_MUX_REG: + case TAS571X_CH4_SRC_SELECT_REG: + case TAS571X_PWM_MUX_REG: + return 4; default: return 1; } @@ -241,6 +245,26 @@ static const struct snd_kcontrol_new tas5711_controls[] = { 1, 1), }; +static const struct regmap_range tas571x_readonly_regs_range[] = { + regmap_reg_range(TAS571X_CLK_CTRL_REG, TAS571X_DEV_ID_REG), +}; + +static const struct regmap_range tas571x_volatile_regs_range[] = { + regmap_reg_range(TAS571X_CLK_CTRL_REG, TAS571X_ERR_STATUS_REG), + regmap_reg_range(TAS571X_OSC_TRIM_REG, TAS571X_OSC_TRIM_REG), +}; + +static const struct regmap_access_table tas571x_write_regs = { + .no_ranges = tas571x_readonly_regs_range, + .n_no_ranges = ARRAY_SIZE(tas571x_readonly_regs_range), +}; + +static const struct regmap_access_table tas571x_volatile_regs = { + .yes_ranges = tas571x_volatile_regs_range, + .n_yes_ranges = ARRAY_SIZE(tas571x_volatile_regs_range), + +}; + static const struct reg_default tas5711_reg_defaults[] = { { 0x04, 0x05 }, { 0x05, 0x40 }, @@ -260,6 +284,8 @@ static const struct regmap_config tas5711_regmap_config = { .reg_defaults = tas5711_reg_defaults, .num_reg_defaults = ARRAY_SIZE(tas5711_reg_defaults), .cache_type = REGCACHE_RBTREE, + .wr_table = &tas571x_write_regs, + .volatile_table = &tas571x_volatile_regs, }; static const struct tas571x_chip tas5711_chip = { @@ -314,6 +340,8 @@ static const struct regmap_config tas5717_regmap_config = { .reg_defaults = tas5717_reg_defaults, .num_reg_defaults = ARRAY_SIZE(tas5717_reg_defaults), .cache_type = REGCACHE_RBTREE, + .wr_table = &tas571x_write_regs, + .volatile_table = &tas571x_volatile_regs, }; /* This entry is reused for tas5719 as the software interface is identical. */ diff --git a/sound/soc/codecs/tas571x.h b/sound/soc/codecs/tas571x.h index 0aee471..cf800c3 100644 --- a/sound/soc/codecs/tas571x.h +++ b/sound/soc/codecs/tas571x.h @@ -13,6 +13,10 @@ #define _TAS571X_H /* device registers */ +#define TAS571X_CLK_CTRL_REG 0x00 +#define TAS571X_DEV_ID_REG 0x01 +#define TAS571X_ERR_STATUS_REG 0x02 +#define TAS571X_SYS_CTRL_1_REG 0x03 #define TAS571X_SDI_REG 0x04 #define TAS571X_SDI_FMT_MASK 0x0f @@ -27,7 +31,25 @@ #define TAS571X_MVOL_REG 0x07 #define TAS571X_CH1_VOL_REG 0x08 #define TAS571X_CH2_VOL_REG 0x09 +#define TAS571X_CH3_VOL_REG 0x0a +#define TAS571X_VOL_CFG_REG 0x0e +#define TAS571X_MODULATION_LIMIT_REG 0x10 +#define TAS571X_IC_DELAY_CH1_REG 0x11 +#define TAS571X_IC_DELAY_CH2_REG 0x12 +#define TAS571X_IC_DELAY_CH3_REG 0x13 +#define TAS571X_IC_DELAY_CH4_REG 0x14 +#define TAS571X_PWM_CH_SDN_GROUP_REG 0x19 /* N/A on TAS5717, TAS5719 */ +#define TAS571X_PWM_CH1_SDN_MASK (1<<0) +#define TAS571X_PWM_CH2_SDN_SHIFT (1<<1) +#define TAS571X_PWM_CH3_SDN_SHIFT (1<<2) +#define TAS571X_PWM_CH4_SDN_SHIFT (1<<3) + +#define TAS571X_START_STOP_PERIOD_REG 0x1a #define TAS571X_OSC_TRIM_REG 0x1b +#define TAS571X_BKND_ERR_REG 0x1c +#define TAS571X_INPUT_MUX_REG 0x20 +#define TAS571X_CH4_SRC_SELECT_REG 0x21 +#define TAS571X_PWM_MUX_REG 0x25 #endif /* _TAS571X_H */ -- 1.9.1

3 9

[alsa-devel] [PATCH v2 0/2] ASoC: qcom: fix static checker warnings.
by Srinivas Kandagatla 31 Mar '16

31 Mar '16

This patchset fixes 2 static checker warnings reported by Dan Carpenter. Thanks, srini Srinivas Kandagatla (2): ASoC: qcom: Fix uninitialized symbol warning. ASoC: qcom: remove IS_ERR_VALUE usage on int. sound/soc/qcom/lpass-platform.c | 8 +++++--- 1 file changed, 5 insertions(+), 3 deletions(-) -- 2.5.0

2 4

[alsa-devel] [PATCH 1/2] [media] Revert "[media] sound/usb: Use Media Controller API to share media resources"
by Mauro Carvalho Chehab 31 Mar '16

31 Mar '16

Unfortunately, this patch caused several regressions at au0828 and snd-usb-audio, like this one: https://bugzilla.kernel.org/show_bug.cgi?id=115561 It also showed several troubles at the MC core that handles pretty poorly the memory protections and data lifetime management. So, better to revert it and fix the core before reapplying this change. This reverts commit aebb2b89bff0 ("[media] sound/usb: Use Media Controller API to share media resources")' Signed-off-by: Mauro Carvalho Chehab <mchehab(a)osg.samsung.com> --- sound/usb/Kconfig | 4 - sound/usb/Makefile | 2 - sound/usb/card.c | 14 --- sound/usb/card.h | 3 - sound/usb/media.c | 318 ----------------------------------------------- sound/usb/media.h | 72 ----------- sound/usb/mixer.h | 3 - sound/usb/pcm.c | 28 +---- sound/usb/quirks-table.h | 1 - sound/usb/stream.c | 2 - sound/usb/usbaudio.h | 6 - 11 files changed, 5 insertions(+), 448 deletions(-) delete mode 100644 sound/usb/media.c delete mode 100644 sound/usb/media.h diff --git a/sound/usb/Kconfig b/sound/usb/Kconfig index d14bf411515b..a452ad7cec40 100644 --- a/sound/usb/Kconfig +++ b/sound/usb/Kconfig @@ -15,7 +15,6 @@ config SND_USB_AUDIO select SND_RAWMIDI select SND_PCM select BITREVERSE - select SND_USB_AUDIO_USE_MEDIA_CONTROLLER if MEDIA_CONTROLLER && (MEDIA_SUPPORT=y || MEDIA_SUPPORT=SND_USB_AUDIO) help Say Y here to include support for USB audio and USB MIDI devices. @@ -23,9 +22,6 @@ config SND_USB_AUDIO To compile this driver as a module, choose M here: the module will be called snd-usb-audio. -config SND_USB_AUDIO_USE_MEDIA_CONTROLLER - bool - config SND_USB_UA101 tristate "Edirol UA-101/UA-1000 driver" select SND_PCM diff --git a/sound/usb/Makefile b/sound/usb/Makefile index 8dca3c407f5a..2d2d122b069f 100644 --- a/sound/usb/Makefile +++ b/sound/usb/Makefile @@ -15,8 +15,6 @@ snd-usb-audio-objs := card.o \ quirks.o \ stream.o -snd-usb-audio-$(CONFIG_SND_USB_AUDIO_USE_MEDIA_CONTROLLER) += media.o - snd-usbmidi-lib-objs := midi.o # Toplevel Module Dependency diff --git a/sound/usb/card.c b/sound/usb/card.c index 63244bbba8c7..3fc63583a537 100644 --- a/sound/usb/card.c +++ b/sound/usb/card.c @@ -66,7 +66,6 @@ #include "format.h" #include "power.h" #include "stream.h" -#include "media.h" MODULE_AUTHOR("Takashi Iwai <tiwai(a)suse.de>"); MODULE_DESCRIPTION("USB Audio"); @@ -612,11 +611,6 @@ static int usb_audio_probe(struct usb_interface *intf, if (err < 0) goto __error; - if (quirk->media_device) { - /* don't want to fail when media_snd_device_create() fails */ - media_snd_device_create(chip, intf); - } - usb_chip[chip->index] = chip; chip->num_interfaces++; usb_set_intfdata(intf, chip); @@ -673,14 +667,6 @@ static void usb_audio_disconnect(struct usb_interface *intf) list_for_each(p, &chip->midi_list) { snd_usbmidi_disconnect(p); } - /* - * Nice to check quirk && quirk->media_device - * need some special handlings. Doesn't look like - * we have access to quirk here - * Acceses mixer_list - */ - media_snd_device_delete(chip); - /* release mixer resources */ list_for_each_entry(mixer, &chip->mixer_list, list) { snd_usb_mixer_disconnect(mixer); diff --git a/sound/usb/card.h b/sound/usb/card.h index 34a0898e2238..71778ca4b26a 100644 --- a/sound/usb/card.h +++ b/sound/usb/card.h @@ -105,8 +105,6 @@ struct snd_usb_endpoint { struct list_head list; }; -struct media_ctl; - struct snd_usb_substream { struct snd_usb_stream *stream; struct usb_device *dev; @@ -158,7 +156,6 @@ struct snd_usb_substream { } dsd_dop; bool trigger_tstamp_pending_update; /* trigger timestamp being updated from initial estimate */ - struct media_ctl *media_ctl; }; struct snd_usb_stream { diff --git a/sound/usb/media.c b/sound/usb/media.c deleted file mode 100644 index 93a50d01490c..000000000000 --- a/sound/usb/media.c +++ /dev/null @@ -1,318 +0,0 @@ -/* - * media.c - Media Controller specific ALSA driver code - * - * Copyright (c) 2016 Shuah Khan <shuahkh(a)osg.samsung.com> - * Copyright (c) 2016 Samsung Electronics Co., Ltd. - * - * This file is released under the GPLv2. - */ - -/* - * This file adds Media Controller support to ALSA driver - * to use the Media Controller API to share tuner with DVB - * and V4L2 drivers that control media device. Media device - * is created based on existing quirks framework. Using this - * approach, the media controller API usage can be added for - * a specific device. -*/ - -#include <linux/init.h> -#include <linux/list.h> -#include <linux/mutex.h> -#include <linux/slab.h> -#include <linux/usb.h> - -#include <sound/pcm.h> -#include <sound/core.h> - -#include "usbaudio.h" -#include "card.h" -#include "mixer.h" -#include "media.h" - -static int media_snd_enable_source(struct media_ctl *mctl) -{ - if (mctl && mctl->media_dev->enable_source) - return mctl->media_dev->enable_source(&mctl->media_entity, - &mctl->media_pipe); - return 0; -} - -static void media_snd_disable_source(struct media_ctl *mctl) -{ - if (mctl && mctl->media_dev->disable_source) - mctl->media_dev->disable_source(&mctl->media_entity); -} - -int media_snd_stream_init(struct snd_usb_substream *subs, struct snd_pcm *pcm, - int stream) -{ - struct media_device *mdev; - struct media_ctl *mctl; - struct device *pcm_dev = &pcm->streams[stream].dev; - u32 intf_type; - int ret = 0; - u16 mixer_pad; - struct media_entity *entity; - - mdev = subs->stream->chip->media_dev; - if (!mdev) - return -ENODEV; - - if (subs->media_ctl) - return 0; - - /* allocate media_ctl */ - mctl = kzalloc(sizeof(*mctl), GFP_KERNEL); - if (!mctl) - return -ENOMEM; - - mctl->media_dev = mdev; - if (stream == SNDRV_PCM_STREAM_PLAYBACK) { - intf_type = MEDIA_INTF_T_ALSA_PCM_PLAYBACK; - mctl->media_entity.function = MEDIA_ENT_F_AUDIO_PLAYBACK; - mctl->media_pad.flags = MEDIA_PAD_FL_SOURCE; - mixer_pad = 1; - } else { - intf_type = MEDIA_INTF_T_ALSA_PCM_CAPTURE; - mctl->media_entity.function = MEDIA_ENT_F_AUDIO_CAPTURE; - mctl->media_pad.flags = MEDIA_PAD_FL_SINK; - mixer_pad = 2; - } - mctl->media_entity.name = pcm->name; - media_entity_pads_init(&mctl->media_entity, 1, &mctl->media_pad); - ret = media_device_register_entity(mctl->media_dev, - &mctl->media_entity); - if (ret) - goto free_mctl; - - mctl->intf_devnode = media_devnode_create(mdev, intf_type, 0, - MAJOR(pcm_dev->devt), - MINOR(pcm_dev->devt)); - if (!mctl->intf_devnode) { - ret = -ENOMEM; - goto unregister_entity; - } - mctl->intf_link = media_create_intf_link(&mctl->media_entity, - &mctl->intf_devnode->intf, - MEDIA_LNK_FL_ENABLED); - if (!mctl->intf_link) { - ret = -ENOMEM; - goto devnode_remove; - } - - /* create link between mixer and audio */ - media_device_for_each_entity(entity, mdev) { - switch (entity->function) { - case MEDIA_ENT_F_AUDIO_MIXER: - ret = media_create_pad_link(entity, mixer_pad, - &mctl->media_entity, 0, - MEDIA_LNK_FL_ENABLED); - if (ret) - goto remove_intf_link; - break; - } - } - - subs->media_ctl = mctl; - return 0; - -remove_intf_link: - media_remove_intf_link(mctl->intf_link); -devnode_remove: - media_devnode_remove(mctl->intf_devnode); -unregister_entity: - media_device_unregister_entity(&mctl->media_entity); -free_mctl: - kfree(mctl); - return ret; -} - -void media_snd_stream_delete(struct snd_usb_substream *subs) -{ - struct media_ctl *mctl = subs->media_ctl; - - if (mctl && mctl->media_dev) { - struct media_device *mdev; - - mdev = subs->stream->chip->media_dev; - if (mdev && media_devnode_is_registered(&mdev->devnode)) { - media_devnode_remove(mctl->intf_devnode); - media_device_unregister_entity(&mctl->media_entity); - media_entity_cleanup(&mctl->media_entity); - } - kfree(mctl); - subs->media_ctl = NULL; - } -} - -int media_snd_start_pipeline(struct snd_usb_substream *subs) -{ - struct media_ctl *mctl = subs->media_ctl; - - if (mctl) - return media_snd_enable_source(mctl); - return 0; -} - -void media_snd_stop_pipeline(struct snd_usb_substream *subs) -{ - struct media_ctl *mctl = subs->media_ctl; - - if (mctl) - media_snd_disable_source(mctl); -} - -int media_snd_mixer_init(struct snd_usb_audio *chip) -{ - struct device *ctl_dev = &chip->card->ctl_dev; - struct media_intf_devnode *ctl_intf; - struct usb_mixer_interface *mixer; - struct media_device *mdev = chip->media_dev; - struct media_mixer_ctl *mctl; - u32 intf_type = MEDIA_INTF_T_ALSA_CONTROL; - int ret; - - if (!mdev) - return -ENODEV; - - ctl_intf = chip->ctl_intf_media_devnode; - if (!ctl_intf) { - ctl_intf = media_devnode_create(mdev, intf_type, 0, - MAJOR(ctl_dev->devt), - MINOR(ctl_dev->devt)); - if (!ctl_intf) - return -ENOMEM; - chip->ctl_intf_media_devnode = ctl_intf; - } - - list_for_each_entry(mixer, &chip->mixer_list, list) { - - if (mixer->media_mixer_ctl) - continue; - - /* allocate media_mixer_ctl */ - mctl = kzalloc(sizeof(*mctl), GFP_KERNEL); - if (!mctl) - return -ENOMEM; - - mctl->media_dev = mdev; - mctl->media_entity.function = MEDIA_ENT_F_AUDIO_MIXER; - mctl->media_entity.name = chip->card->mixername; - mctl->media_pad[0].flags = MEDIA_PAD_FL_SINK; - mctl->media_pad[1].flags = MEDIA_PAD_FL_SOURCE; - mctl->media_pad[2].flags = MEDIA_PAD_FL_SOURCE; - media_entity_pads_init(&mctl->media_entity, MEDIA_MIXER_PAD_MAX, - mctl->media_pad); - ret = media_device_register_entity(mctl->media_dev, - &mctl->media_entity); - if (ret) { - kfree(mctl); - return ret; - } - - mctl->intf_link = media_create_intf_link(&mctl->media_entity, - &ctl_intf->intf, - MEDIA_LNK_FL_ENABLED); - if (!mctl->intf_link) { - media_device_unregister_entity(&mctl->media_entity); - media_entity_cleanup(&mctl->media_entity); - kfree(mctl); - return -ENOMEM; - } - mctl->intf_devnode = ctl_intf; - mixer->media_mixer_ctl = mctl; - } - return 0; -} - -static void media_snd_mixer_delete(struct snd_usb_audio *chip) -{ - struct usb_mixer_interface *mixer; - struct media_device *mdev = chip->media_dev; - - if (!mdev) - return; - - list_for_each_entry(mixer, &chip->mixer_list, list) { - struct media_mixer_ctl *mctl; - - mctl = mixer->media_mixer_ctl; - if (!mixer->media_mixer_ctl) - continue; - - if (media_devnode_is_registered(&mdev->devnode)) { - media_device_unregister_entity(&mctl->media_entity); - media_entity_cleanup(&mctl->media_entity); - } - kfree(mctl); - mixer->media_mixer_ctl = NULL; - } - if (media_devnode_is_registered(&mdev->devnode)) - media_devnode_remove(chip->ctl_intf_media_devnode); - chip->ctl_intf_media_devnode = NULL; -} - -int media_snd_device_create(struct snd_usb_audio *chip, - struct usb_interface *iface) -{ - struct media_device *mdev; - struct usb_device *usbdev = interface_to_usbdev(iface); - int ret; - - mdev = media_device_get_devres(&usbdev->dev); - if (!mdev) - return -ENOMEM; - if (!mdev->dev) { - /* register media device */ - mdev->dev = &usbdev->dev; - if (usbdev->product) - strlcpy(mdev->model, usbdev->product, - sizeof(mdev->model)); - if (usbdev->serial) - strlcpy(mdev->serial, usbdev->serial, - sizeof(mdev->serial)); - strcpy(mdev->bus_info, usbdev->devpath); - mdev->hw_revision = le16_to_cpu(usbdev->descriptor.bcdDevice); - media_device_init(mdev); - } - if (!media_devnode_is_registered(&mdev->devnode)) { - ret = media_device_register(mdev); - if (ret) { - dev_err(&usbdev->dev, - "Couldn't register media device. Error: %d\n", - ret); - return ret; - } - } - - /* save media device - avoid lookups */ - chip->media_dev = mdev; - - /* Create media entities for mixer and control dev */ - ret = media_snd_mixer_init(chip); - if (ret) { - dev_err(&usbdev->dev, - "Couldn't create media mixer entities. Error: %d\n", - ret); - - /* clear saved media_dev */ - chip->media_dev = NULL; - - return ret; - } - return 0; -} - -void media_snd_device_delete(struct snd_usb_audio *chip) -{ - struct media_device *mdev = chip->media_dev; - - media_snd_mixer_delete(chip); - - if (mdev) { - if (media_devnode_is_registered(&mdev->devnode)) - media_device_unregister(mdev); - chip->media_dev = NULL; - } -} diff --git a/sound/usb/media.h b/sound/usb/media.h deleted file mode 100644 index 1dcdcdc5f7aa..000000000000 --- a/sound/usb/media.h +++ /dev/null @@ -1,72 +0,0 @@ -/* - * media.h - Media Controller specific ALSA driver code - * - * Copyright (c) 2016 Shuah Khan <shuahkh(a)osg.samsung.com> - * Copyright (c) 2016 Samsung Electronics Co., Ltd. - * - * This file is released under the GPLv2. - */ - -/* - * This file adds Media Controller support to ALSA driver - * to use the Media Controller API to share tuner with DVB - * and V4L2 drivers that control media device. Media device - * is created based on existing quirks framework. Using this - * approach, the media controller API usage can be added for - * a specific device. -*/ -#ifndef __MEDIA_H - -#ifdef CONFIG_SND_USB_AUDIO_USE_MEDIA_CONTROLLER - -#include <media/media-device.h> -#include <media/media-entity.h> -#include <sound/asound.h> - -struct media_ctl { - struct media_device *media_dev; - struct media_entity media_entity; - struct media_intf_devnode *intf_devnode; - struct media_link *intf_link; - struct media_pad media_pad; - struct media_pipeline media_pipe; -}; - -/* - * One source pad each for SNDRV_PCM_STREAM_CAPTURE and - * SNDRV_PCM_STREAM_PLAYBACK. One for sink pad to link - * to AUDIO Source -*/ -#define MEDIA_MIXER_PAD_MAX (SNDRV_PCM_STREAM_LAST + 2) - -struct media_mixer_ctl { - struct media_device *media_dev; - struct media_entity media_entity; - struct media_intf_devnode *intf_devnode; - struct media_link *intf_link; - struct media_pad media_pad[MEDIA_MIXER_PAD_MAX]; - struct media_pipeline media_pipe; -}; - -int media_snd_device_create(struct snd_usb_audio *chip, - struct usb_interface *iface); -void media_snd_device_delete(struct snd_usb_audio *chip); -int media_snd_stream_init(struct snd_usb_substream *subs, struct snd_pcm *pcm, - int stream); -void media_snd_stream_delete(struct snd_usb_substream *subs); -int media_snd_start_pipeline(struct snd_usb_substream *subs); -void media_snd_stop_pipeline(struct snd_usb_substream *subs); -#else -static inline int media_snd_device_create(struct snd_usb_audio *chip, - struct usb_interface *iface) - { return 0; } -static inline void media_snd_device_delete(struct snd_usb_audio *chip) { } -static inline int media_snd_stream_init(struct snd_usb_substream *subs, - struct snd_pcm *pcm, int stream) - { return 0; } -static inline void media_snd_stream_delete(struct snd_usb_substream *subs) { } -static inline int media_snd_start_pipeline(struct snd_usb_substream *subs) - { return 0; } -static inline void media_snd_stop_pipeline(struct snd_usb_substream *subs) { } -#endif -#endif /* __MEDIA_H */ diff --git a/sound/usb/mixer.h b/sound/usb/mixer.h index f3789446ab9c..3417ef347e40 100644 --- a/sound/usb/mixer.h +++ b/sound/usb/mixer.h @@ -3,8 +3,6 @@ #include <sound/info.h> -struct media_mixer_ctl; - struct usb_mixer_interface { struct snd_usb_audio *chip; struct usb_host_interface *hostif; @@ -24,7 +22,6 @@ struct usb_mixer_interface { struct urb *rc_urb; struct usb_ctrlrequest *rc_setup_packet; u8 rc_buffer[6]; - struct media_mixer_ctl *media_mixer_ctl; }; #define MAX_CHANNELS 16 /* max logical channels */ diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index 0e4e0640c504..44d178ee9177 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -35,7 +35,6 @@ #include "pcm.h" #include "clock.h" #include "power.h" -#include "media.h" #define SUBSTREAM_FLAG_DATA_EP_STARTED 0 #define SUBSTREAM_FLAG_SYNC_EP_STARTED 1 @@ -718,14 +717,10 @@ static int snd_usb_hw_params(struct snd_pcm_substream *substream, struct audioformat *fmt; int ret; - ret = media_snd_start_pipeline(subs); - if (ret) - return ret; - ret = snd_pcm_lib_alloc_vmalloc_buffer(substream, params_buffer_bytes(hw_params)); if (ret < 0) - goto err_ret; + return ret; subs->pcm_format = params_format(hw_params); subs->period_bytes = params_period_bytes(hw_params); @@ -739,27 +734,22 @@ static int snd_usb_hw_params(struct snd_pcm_substream *substream, dev_dbg(&subs->dev->dev, "cannot set format: format = %#x, rate = %d, channels = %d\n", subs->pcm_format, subs->cur_rate, subs->channels); - ret = -EINVAL; - goto err_ret; + return -EINVAL; } ret = snd_usb_lock_shutdown(subs->stream->chip); if (ret < 0) - goto err_ret; + return ret; ret = set_format(subs, fmt); snd_usb_unlock_shutdown(subs->stream->chip); if (ret < 0) - goto err_ret; + return ret; subs->interface = fmt->iface; subs->altset_idx = fmt->altset_idx; subs->need_setup_ep = true; return 0; - -err_ret: - media_snd_stop_pipeline(subs); - return ret; } /* @@ -771,7 +761,6 @@ static int snd_usb_hw_free(struct snd_pcm_substream *substream) { struct snd_usb_substream *subs = substream->runtime->private_data; - media_snd_stop_pipeline(subs); subs->cur_audiofmt = NULL; subs->cur_rate = 0; subs->period_bytes = 0; @@ -1232,7 +1221,6 @@ static int snd_usb_pcm_open(struct snd_pcm_substream *substream, int direction) struct snd_usb_stream *as = snd_pcm_substream_chip(substream); struct snd_pcm_runtime *runtime = substream->runtime; struct snd_usb_substream *subs = &as->substream[direction]; - int ret; subs->interface = -1; subs->altset_idx = 0; @@ -1246,12 +1234,7 @@ static int snd_usb_pcm_open(struct snd_pcm_substream *substream, int direction) subs->dsd_dop.channel = 0; subs->dsd_dop.marker = 1; - ret = setup_hw_info(runtime, subs); - if (ret == 0) - ret = media_snd_stream_init(subs, as->pcm, direction); - if (ret) - snd_usb_autosuspend(subs->stream->chip); - return ret; + return setup_hw_info(runtime, subs); } static int snd_usb_pcm_close(struct snd_pcm_substream *substream, int direction) @@ -1260,7 +1243,6 @@ static int snd_usb_pcm_close(struct snd_pcm_substream *substream, int direction) struct snd_usb_substream *subs = &as->substream[direction]; stop_endpoints(subs, true); - media_snd_stop_pipeline(subs); if (subs->interface >= 0 && !snd_usb_lock_shutdown(subs->stream->chip)) { diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index 9d087b19c70c..c60a776e815d 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -2886,7 +2886,6 @@ YAMAHA_DEVICE(0x7010, "UB99"), .product_name = pname, \ .ifnum = QUIRK_ANY_INTERFACE, \ .type = QUIRK_AUDIO_ALIGN_TRANSFER, \ - .media_device = 1, \ } \ } diff --git a/sound/usb/stream.c b/sound/usb/stream.c index 51258a15f653..c4dc577ab1bd 100644 --- a/sound/usb/stream.c +++ b/sound/usb/stream.c @@ -36,7 +36,6 @@ #include "format.h" #include "clock.h" #include "stream.h" -#include "media.h" /* * free a substream @@ -53,7 +52,6 @@ static void free_substream(struct snd_usb_substream *subs) kfree(fp); } kfree(subs->rate_list.list); - media_snd_stream_delete(subs); } diff --git a/sound/usb/usbaudio.h b/sound/usb/usbaudio.h index a161c7c1b126..b665d85555cb 100644 --- a/sound/usb/usbaudio.h +++ b/sound/usb/usbaudio.h @@ -30,9 +30,6 @@ * */ -struct media_device; -struct media_intf_devnode; - struct snd_usb_audio { int index; struct usb_device *dev; @@ -63,8 +60,6 @@ struct snd_usb_audio { bool autoclock; /* from the 'autoclock' module param */ struct usb_host_interface *ctrl_intf; /* the audio control interface */ - struct media_device *media_dev; - struct media_intf_devnode *ctl_intf_media_devnode; }; #define usb_audio_err(chip, fmt, args...) \ @@ -115,7 +110,6 @@ struct snd_usb_audio_quirk { const char *product_name; int16_t ifnum; uint16_t type; - bool media_device; const void *data; }; -- 2.5.5

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[alsa-devel] [PATCH] ASoC: qcom: Fix some static checker warnings
by Srinivas Kandagatla 31 Mar '16

31 Mar '16

This patch fixes below static checker warning plus suggestions from Dan. sound/soc/qcom/lpass-platform.c:555 lpass_platform_pcm_new() error: uninitialized symbol 'ret'. sound/soc/qcom/lpass-platform.c 515 csubstream = pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream; 516 if (csubstream) { 517 if (v->alloc_dma_channel) 518 data->wrdma_ch = v->alloc_dma_channel(drvdata, 519 SNDRV_PCM_STREAM_CAPTURE); 520 521 if (IS_ERR_VALUE(data->wrdma_ch)) 522 goto capture_alloc_err; wrdma_ch is an int so this should just be: if (data->wrdma_ch < 0) { ret = data->wrdma_ch; goto capture_alloc_err; } Reported-by: Dan Carpenter <dan.carpenter(a)oracle.com> Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> --- sound/soc/qcom/lpass-platform.c | 8 +++++--- 1 file changed, 5 insertions(+), 3 deletions(-) diff --git a/sound/soc/qcom/lpass-platform.c b/sound/soc/qcom/lpass-platform.c index 6e86654..db000c6 100644 --- a/sound/soc/qcom/lpass-platform.c +++ b/sound/soc/qcom/lpass-platform.c @@ -474,7 +474,7 @@ static int lpass_platform_pcm_new(struct snd_soc_pcm_runtime *soc_runtime) struct lpass_data *drvdata = snd_soc_platform_get_drvdata(soc_runtime->platform); struct lpass_variant *v = drvdata->variant; - int ret; + int ret = -EINVAL; struct lpass_pcm_data *data; size_t size = lpass_platform_pcm_hardware.buffer_bytes_max; @@ -491,7 +491,7 @@ static int lpass_platform_pcm_new(struct snd_soc_pcm_runtime *soc_runtime) data->rdma_ch = v->alloc_dma_channel(drvdata, SNDRV_PCM_STREAM_PLAYBACK); - if (IS_ERR_VALUE(data->rdma_ch)) + if (data->rdma_ch < 0) return data->rdma_ch; drvdata->substream[data->rdma_ch] = psubstream; @@ -518,8 +518,10 @@ static int lpass_platform_pcm_new(struct snd_soc_pcm_runtime *soc_runtime) data->wrdma_ch = v->alloc_dma_channel(drvdata, SNDRV_PCM_STREAM_CAPTURE); - if (IS_ERR_VALUE(data->wrdma_ch)) + if (data->wrdma_ch < 0) { + ret = data->wrdma_ch; goto capture_alloc_err; + } drvdata->substream[data->wrdma_ch] = csubstream; -- 2.5.0

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[alsa-devel] [PATCH v2 1/5] Sound: SOC: TAS571x: added missing register literals
by Petr Kulhavy 31 Mar '16

31 Mar '16

The list of TAS571x registers was incomplete. Added the missing register definitions up to register 0x25 Signed-off-by: Petr Kulhavy <petr(a)barix.com> --- v1: <initial> v2: <no change> sound/soc/codecs/tas571x.c | 4 ++++ sound/soc/codecs/tas571x.h | 22 ++++++++++++++++++++++ 2 files changed, 26 insertions(+) diff --git a/sound/soc/codecs/tas571x.c b/sound/soc/codecs/tas571x.c index 39307ad..346d3da 100644 --- a/sound/soc/codecs/tas571x.c +++ b/sound/soc/codecs/tas571x.c @@ -57,6 +57,10 @@ static int tas571x_register_size(struct tas571x_private *priv, unsigned int reg) case TAS571X_CH1_VOL_REG: case TAS571X_CH2_VOL_REG: return priv->chip->vol_reg_size; + case TAS571X_INPUT_MUX_REG: + case TAS571X_CH4_SRC_SELECT_REG: + case TAS571X_PWM_MUX_REG: + return 4; default: return 1; } diff --git a/sound/soc/codecs/tas571x.h b/sound/soc/codecs/tas571x.h index 0aee471..cf800c3 100644 --- a/sound/soc/codecs/tas571x.h +++ b/sound/soc/codecs/tas571x.h @@ -13,6 +13,10 @@ #define _TAS571X_H /* device registers */ +#define TAS571X_CLK_CTRL_REG 0x00 +#define TAS571X_DEV_ID_REG 0x01 +#define TAS571X_ERR_STATUS_REG 0x02 +#define TAS571X_SYS_CTRL_1_REG 0x03 #define TAS571X_SDI_REG 0x04 #define TAS571X_SDI_FMT_MASK 0x0f @@ -27,7 +31,25 @@ #define TAS571X_MVOL_REG 0x07 #define TAS571X_CH1_VOL_REG 0x08 #define TAS571X_CH2_VOL_REG 0x09 +#define TAS571X_CH3_VOL_REG 0x0a +#define TAS571X_VOL_CFG_REG 0x0e +#define TAS571X_MODULATION_LIMIT_REG 0x10 +#define TAS571X_IC_DELAY_CH1_REG 0x11 +#define TAS571X_IC_DELAY_CH2_REG 0x12 +#define TAS571X_IC_DELAY_CH3_REG 0x13 +#define TAS571X_IC_DELAY_CH4_REG 0x14 +#define TAS571X_PWM_CH_SDN_GROUP_REG 0x19 /* N/A on TAS5717, TAS5719 */ +#define TAS571X_PWM_CH1_SDN_MASK (1<<0) +#define TAS571X_PWM_CH2_SDN_SHIFT (1<<1) +#define TAS571X_PWM_CH3_SDN_SHIFT (1<<2) +#define TAS571X_PWM_CH4_SDN_SHIFT (1<<3) + +#define TAS571X_START_STOP_PERIOD_REG 0x1a #define TAS571X_OSC_TRIM_REG 0x1b +#define TAS571X_BKND_ERR_REG 0x1c +#define TAS571X_INPUT_MUX_REG 0x20 +#define TAS571X_CH4_SRC_SELECT_REG 0x21 +#define TAS571X_PWM_MUX_REG 0x25 #endif /* _TAS571X_H */ -- 1.9.1

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