January 2019 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [RFC PATCH] ASoC: pcm512x: Implement the set_bclk_ratio interface
by Dimitris Papavasiliou 24 Jan '19

24 Jan '19

>From 5e6ad135fd063d4b22cd962de43499a161bbc7db Mon Sep 17 00:00:00 2001 From: Dimitris Papavasiliou <dpapavas(a)gmail.com> Date: Fri, 11 Jan 2019 22:13:27 +0200 Subject: [RFC PATCH] ASoC: pcm512x: Implement the set_bclk_ratio interface Some boards, such as the HiFiBerry DAC+ Pro, use a pair of external oscillators, to generate 44.1 or 48kHz multiples and are forced to resort to hacks[1] in order to support 24-bit data without ending up with fractional dividers. This patch allows the machine driver to use 32-bit frames for 24-bit data to avoid such issues. [1] http://mailman.alsa-project.org/pipermail/alsa-devel/2018-December/143442.h… Signed-off-by: Dimitris Papavasiliou <dpapavas(a)gmail.com> --- Notes: The patch also includes a related change in the calculation of the bclk divider. The current calculation is not very clear to me, as I can't find much concrete information on what this sample rate fraction is, how it's calculated and how, therefore, its denominator would fit into the calculation. It also yields slightly wrong dividers in certain cases, which the machine driver for my HiFiBerry DAC+ Pro board seemingly tries to circumvent, by updating the rate fraction so as to suit this calculation. The updated calculation seems to work fine in my tests and, as far as I can see, should correctly yield the smallest bit clock rate that would fit the frame. Please comment if anything looks off. sound/soc/codecs/pcm512x.c | 29 +++++++++++++++++++++++------ 1 file changed, 23 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/pcm512x.c b/sound/soc/codecs/pcm512x.c index 4cc24a5d5c31..8cd728f9a1eb 100644 --- a/sound/soc/codecs/pcm512x.c +++ b/sound/soc/codecs/pcm512x.c @@ -55,6 +55,7 @@ struct pcm512x_priv { unsigned long overclock_dsp; int mute; struct mutex mutex; + unsigned int bclk_ratio; }; /* @@ -915,16 +916,21 @@ static int pcm512x_set_dividers(struct snd_soc_dai *dai, int fssp; int gpio; - lrclk_div = snd_soc_params_to_frame_size(params); - if (lrclk_div == 0) { - dev_err(dev, "No LRCLK?\n"); - return -EINVAL; + if (pcm512x->bclk_ratio > 0) { + lrclk_div = pcm512x->bclk_ratio; + } else { + lrclk_div = snd_soc_params_to_frame_size(params); + + if (lrclk_div == 0) { + dev_err(dev, "No LRCLK?\n"); + return -EINVAL; + } } if (!pcm512x->pll_out) { sck_rate = clk_get_rate(pcm512x->sclk); - bclk_div = params->rate_den * 64 / lrclk_div; - bclk_rate = DIV_ROUND_CLOSEST(sck_rate, bclk_div); + bclk_rate = params_rate(params) * lrclk_div; + bclk_div = DIV_ROUND_CLOSEST(sck_rate, bclk_rate); mck_rate = sck_rate; } else { @@ -1383,6 +1389,16 @@ static int pcm512x_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) return 0; } +static int pcm512x_set_bclk_ratio(struct snd_soc_dai *dai, unsigned int ratio) +{ + struct snd_soc_component *component = dai->component; + struct pcm512x_priv *pcm512x = snd_soc_component_get_drvdata(component); + + pcm512x->bclk_ratio = ratio; + + return 0; +} + static int pcm512x_digital_mute(struct snd_soc_dai *dai, int mute) { struct snd_soc_component *component = dai->component; @@ -1435,6 +1451,7 @@ static const struct snd_soc_dai_ops pcm512x_dai_ops = { .hw_params = pcm512x_hw_params, .set_fmt = pcm512x_set_fmt, .digital_mute = pcm512x_digital_mute, + .set_bclk_ratio = pcm512x_set_bclk_ratio, }; static struct snd_soc_dai_driver pcm512x_dai = { -- 2.11.0

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[alsa-devel] [PATCH][next] ASoC: Intel: bytcht_es8316: make mic_name static, reduces object code size
by Colin King 24 Jan '19

24 Jan '19

From: Colin Ian King <colin.king(a)canonical.com> Don't populate the const array mic_name on the stack but instead make it static. Makes the object code smaller by 28 bytes: Before: text data bss dec hex filename 14107 8832 224 23163 5a7b bytcht_es8316.o After: text data bss dec hex filename 14015 8896 224 23135 5a5f bytcht_es8316.o (gcc version 8.2.0 x86_64) Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/intel/boards/bytcht_es8316.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/boards/bytcht_es8316.c b/sound/soc/intel/boards/bytcht_es8316.c index fa9c4cf97686..1364e4e601d8 100644 --- a/sound/soc/intel/boards/bytcht_es8316.c +++ b/sound/soc/intel/boards/bytcht_es8316.c @@ -437,7 +437,7 @@ static const struct acpi_gpio_mapping byt_cht_es8316_gpios[] = { static int snd_byt_cht_es8316_mc_probe(struct platform_device *pdev) { - const char * const mic_name[] = { "in1", "in2" }; + static const char * const mic_name[] = { "in1", "in2" }; struct byt_cht_es8316_private *priv; struct device *dev = &pdev->dev; struct snd_soc_acpi_mach *mach; -- 2.19.1

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[alsa-devel] [PATCH] ASoC: intel: cht_bsw_max98090_ti: Enable codec clock once and keep it enabled
by Hans de Goede 24 Jan '19

24 Jan '19

Users have been seeing sound stability issues with max98090 codecs since: commit 648e921888ad ("clk: x86: Stop marking clocks as CLK_IS_CRITICAL") At first that commit broke sound for Chromebook Swanky and Clapper models, the problem was that the machine-driver has been controlling the wrong clock on those models since support for them was added. This was hidden by clk-pmc-atom.c keeping the actual clk on unconditionally. With the machine-driver controlling the proper clock, sound works again but we are seeing bug reports describing it as: low volume, "sounds like played at 10x speed" and instable. When these issues are hit the following message is seen in dmesg: "max98090 i2c-193C9890:00: PLL unlocked". Attempts have been made to fix this by inserting a delay between enabling the clk and enabling and checking the pll, but this has not helped. It seems that if we ever disable the clk, the pll looses its lock and then we get various issues. This commit fixes this by enabling the clock once at probe time, in essence restoring the old behavior of clk-pmc-atom.c always keeping the clk on, but then only on machines with a max98090 codec. Fixes: 648e921888ad ("clk: x86: Stop marking clocks as CLK_IS_CRITICAL") Reported-and-tested-by: Mogens Jensen <mogens-jensen(a)protonmail.com> Reported-and-tested-by: Dean Wallace <duffydack73(a)gmail.com> Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> --- sound/soc/intel/boards/cht_bsw_max98090_ti.c | 75 ++++++-------------- 1 file changed, 20 insertions(+), 55 deletions(-) diff --git a/sound/soc/intel/boards/cht_bsw_max98090_ti.c b/sound/soc/intel/boards/cht_bsw_max98090_ti.c index 08a5152e635a..549d6ce12ec4 100644 --- a/sound/soc/intel/boards/cht_bsw_max98090_ti.c +++ b/sound/soc/intel/boards/cht_bsw_max98090_ti.c @@ -44,43 +44,11 @@ struct cht_mc_private { bool ts3a227e_present; }; -static int platform_clock_control(struct snd_soc_dapm_widget *w, - struct snd_kcontrol *k, int event) -{ - struct snd_soc_dapm_context *dapm = w->dapm; - struct snd_soc_card *card = dapm->card; - struct snd_soc_dai *codec_dai; - struct cht_mc_private *ctx = snd_soc_card_get_drvdata(card); - int ret; - - codec_dai = snd_soc_card_get_codec_dai(card, CHT_CODEC_DAI); - if (!codec_dai) { - dev_err(card->dev, "Codec dai not found; Unable to set platform clock\n"); - return -EIO; - } - - if (SND_SOC_DAPM_EVENT_ON(event)) { - ret = clk_prepare_enable(ctx->mclk); - if (ret < 0) { - dev_err(card->dev, - "could not configure MCLK state"); - return ret; - } - } else { - clk_disable_unprepare(ctx->mclk); - } - - return 0; -} - static const struct snd_soc_dapm_widget cht_dapm_widgets[] = { SND_SOC_DAPM_HP("Headphone", NULL), SND_SOC_DAPM_MIC("Headset Mic", NULL), SND_SOC_DAPM_MIC("Int Mic", NULL), SND_SOC_DAPM_SPK("Ext Spk", NULL), - SND_SOC_DAPM_SUPPLY("Platform Clock", SND_SOC_NOPM, 0, 0, - platform_clock_control, SND_SOC_DAPM_PRE_PMU | - SND_SOC_DAPM_POST_PMD), }; static const struct snd_soc_dapm_route cht_audio_map[] = { @@ -97,10 +65,6 @@ static const struct snd_soc_dapm_route cht_audio_map[] = { {"codec_in0", NULL, "ssp2 Rx" }, {"codec_in1", NULL, "ssp2 Rx" }, {"ssp2 Rx", NULL, "HiFi Capture"}, - {"Headphone", NULL, "Platform Clock"}, - {"Headset Mic", NULL, "Platform Clock"}, - {"Int Mic", NULL, "Platform Clock"}, - {"Ext Spk", NULL, "Platform Clock"}, }; static const struct snd_kcontrol_new cht_mc_controls[] = { @@ -222,25 +186,6 @@ static int cht_codec_init(struct snd_soc_pcm_runtime *runtime) "jack detection gpios not added, error %d\n", ret); } - /* - * The firmware might enable the clock at - * boot (this information may or may not - * be reflected in the enable clock register). - * To change the rate we must disable the clock - * first to cover these cases. Due to common - * clock framework restrictions that do not allow - * to disable a clock that has not been enabled, - * we need to enable the clock first. - */ - ret = clk_prepare_enable(ctx->mclk); - if (!ret) - clk_disable_unprepare(ctx->mclk); - - ret = clk_set_rate(ctx->mclk, CHT_PLAT_CLK_3_HZ); - - if (ret) - dev_err(runtime->dev, "unable to set MCLK rate\n"); - return ret; } @@ -459,6 +404,16 @@ static int snd_cht_mc_probe(struct platform_device *pdev) return PTR_ERR(drv->mclk); } + /* + * The MAX98090 does not seem to like it if we muck with its clock, + * so we enable it here once and leave it at that. + */ + ret_val = clk_prepare_enable(drv->mclk); + if (ret_val < 0) { + dev_err(&pdev->dev, "Failed to enable MCLK: %d\n", ret_val); + return ret_val; + } + ret_val = devm_snd_soc_register_card(&pdev->dev, &snd_soc_card_cht); if (ret_val) { dev_err(&pdev->dev, @@ -469,11 +424,21 @@ static int snd_cht_mc_probe(struct platform_device *pdev) return ret_val; } +static int snd_cht_mc_remove(struct platform_device *pdev) +{ + struct snd_soc_card *card = platform_get_drvdata(pdev); + struct cht_mc_private *ctx = snd_soc_card_get_drvdata(card); + + clk_disable_unprepare(ctx->mclk); + return 0; +} + static struct platform_driver snd_cht_mc_driver = { .driver = { .name = "cht-bsw-max98090", }, .probe = snd_cht_mc_probe, + .remove = snd_cht_mc_remove, }; module_platform_driver(snd_cht_mc_driver) -- 2.20.1

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[alsa-devel] [PATCH] ALSA: fireface: comment cleanup about destination address of async transactions for MIDI messages
by Takashi Sakamoto 24 Jan '19

24 Jan '19

In Fireface series, registration of higher 4 bytes of destination address for asynchronous transaction of MIDI messages is done by a write transaction to model-specific register. On the other hand, registration of lower 4 bytes of the address is selectable from 4 options. A register for this registration includes the other purpose options such as input attenuation. Thus this driver expects userspace applications to configure the register. Actual behaviour for the asynchronous transaction is different depending on protocols. In former protocol, destination offset of each transaction is the same as the registered address even if it is block request. In latter models, destination offset of each transaction is the offset of previous transaction plus 4 byte and the transaction is quadlet request. This commit cleanups comments about the above mechanism. Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/fireface/ff-protocol-former.c | 23 +++++++++++++ sound/firewire/fireface/ff-protocol-latter.c | 18 ++++++++++ sound/firewire/fireface/ff-transaction.c | 36 ++++---------------- 3 files changed, 48 insertions(+), 29 deletions(-) diff --git a/sound/firewire/fireface/ff-protocol-former.c b/sound/firewire/fireface/ff-protocol-former.c index e0acf40a02ee..8d1c2c6e907b 100644 --- a/sound/firewire/fireface/ff-protocol-former.c +++ b/sound/firewire/fireface/ff-protocol-former.c @@ -396,6 +396,10 @@ static void ff800_finish_session(struct snd_ff *ff) FF800_ISOC_COMM_STOP, &reg, sizeof(reg), 0); } +// Fireface 800 doesn't allow drivers to register lower 4 bytes of destination +// address. +// A write transaction to clear registered higher 4 bytes of destination address +// has an effect to suppress asynchronous transaction from device. static void ff800_handle_midi_msg(struct snd_ff *ff, unsigned int offset, __le32 *buf, size_t length) { @@ -525,6 +529,25 @@ static void ff400_finish_session(struct snd_ff *ff) FF400_ISOC_COMM_STOP, &reg, sizeof(reg), 0); } +// For Fireface 400, lower 4 bytes of destination address is configured by bit +// flag in quadlet register (little endian) at 0x'0000'801'0051c. Drivers can +// select one of 4 options: +// +// bit flags: offset of destination address +// - 0x04000000: 0x'....'....'0000'0000 +// - 0x08000000: 0x'....'....'0000'0080 +// - 0x10000000: 0x'....'....'0000'0100 +// - 0x20000000: 0x'....'....'0000'0180 +// +// Drivers can suppress the device to transfer asynchronous transactions by +// using below 2 bits. +// - 0x01000000: suppress transmission +// - 0x02000000: suppress transmission +// +// Actually, the register is write-only and includes the other options such as +// input attenuation. This driver allocates destination address with '0000'0000 +// in its lower offset and expects userspace application to configure the +// register for it. static void ff400_handle_midi_msg(struct snd_ff *ff, unsigned int offset, __le32 *buf, size_t length) { diff --git a/sound/firewire/fireface/ff-protocol-latter.c b/sound/firewire/fireface/ff-protocol-latter.c index 817af4447349..0fbc1950327f 100644 --- a/sound/firewire/fireface/ff-protocol-latter.c +++ b/sound/firewire/fireface/ff-protocol-latter.c @@ -266,6 +266,24 @@ static void latter_dump_status(struct snd_ff *ff, struct snd_info_buffer *buffer // NOTE: transactions are transferred within 0x00-0x7f in allocated range of // address. This seems to be for check of discontinuity in receiver side. +// +// Like Fireface 400, drivers can select one of 4 options for lower 4 bytes of +// destination address by bit flags in quadlet register (little endian) at +// 0x'ffff'0000'0014: +// +// bit flags: offset of destination address +// - 0x00002000: 0x'....'....'0000'0000 +// - 0x00004000: 0x'....'....'0000'0080 +// - 0x00008000: 0x'....'....'0000'0100 +// - 0x00010000: 0x'....'....'0000'0180 +// +// Drivers can suppress the device to transfer asynchronous transactions by +// clear these bit flags. +// +// Actually, the register is write-only and includes the other settings such as +// input attenuation. This driver allocates for the first option +// (0x'....'....'0000'0000) and expects userspace application to configure the +// register for it. static void latter_handle_midi_msg(struct snd_ff *ff, unsigned int offset, __le32 *buf, size_t length) { diff --git a/sound/firewire/fireface/ff-transaction.c b/sound/firewire/fireface/ff-transaction.c index d8a8b01b39a1..0d6ad19363b8 100644 --- a/sound/firewire/fireface/ff-transaction.c +++ b/sound/firewire/fireface/ff-transaction.c @@ -165,35 +165,13 @@ static int allocate_own_address(struct snd_ff *ff, int i) return err; } -/* - * Controllers are allowed to register higher 4 bytes of address to receive - * the transactions. Different models have different registers for this purpose; - * e.g. 0x'0000'8010'03f4 for Fireface 400. - * The controllers are not allowed to register lower 4 bytes of the address. - * They are forced to select one of 4 options for the part of address by writing - * corresponding bits to 0x'0000'8010'051f. - * - * The 3rd-6th bits of this register are flags to indicate lower 4 bytes of - * address to which the device transferrs the transactions. In short: - * - 0x20: 0x'....'....'0000'0180 - * - 0x10: 0x'....'....'0000'0100 - * - 0x08: 0x'....'....'0000'0080 - * - 0x04: 0x'....'....'0000'0000 - * - * This driver configure 0x'....'....'0000'0000 to receive MIDI messages from - * units. The 3rd bit of the register should be configured, however this driver - * deligates this task to userspace applications due to a restriction that this - * register is write-only and the other bits have own effects. - * - * Unlike Fireface 800, Fireface 400 cancels transferring asynchronous - * transactions when the 1st and 2nd of the register stand. These two bits have - * the same effect. - * - 0x02, 0x01: cancel transferring - * - * On the other hand, the bits have no effect on Fireface 800. This model - * cancels asynchronous transactions when the higher 4 bytes of address is - * overwritten with zero. - */ +// Controllers are allowed to register higher 4 bytes of destination address to +// receive asynchronous transactions for MIDI messages, while the way to +// register lower 4 bytes of address is different depending on protocols. For +// details, please refer to comments in protocol implementations. +// +// This driver expects userspace applications to configure registers for the +// lower address because in most cases such registers has the other settings. int snd_ff_transaction_reregister(struct snd_ff *ff) { struct fw_card *fw_card = fw_parent_device(ff->unit)->card; -- 2.19.1

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[alsa-devel] [PATCH v9 0/4] Media Device Allocator API
by shuah＠kernel.org 24 Jan '19

24 Jan '19

From: Shuah Khan <shuah(a)kernel.org> Media Device Allocator API to allows multiple drivers share a media device. This API solves a very common use-case for media devices where one physical device (an USB stick) provides both audio and video. When such media device exposes a standard USB Audio class, a proprietary Video class, two or more independent drivers will share a single physical USB bridge. In such cases, it is necessary to coordinate access to the shared resource. Using this API, drivers can allocate a media device with the shared struct device as the key. Once the media device is allocated by a driver, other drivers can get a reference to it. The media device is released when all the references are released. - Tested sharing resources with kaffeine, vlc, xawtv, tvtime, and arecord. When analog is streaming, digital and audio user-space applications detect that the tuner is busy and exit. When digital is streaming, analog and audio applications detect that the tuner is busy and exit. When arecord is owns the tuner, digital and analog detect that the tuner is busy and exit. - Tested media device allocator API with bind/unbind testing on snd-usb-audio and au0828 drivers to make sure /dev/mediaX is released only when the last driver is unbound. - This patch series is tested on 4.20-rc6 - Addressed review comments from Hans on the RFC v8 (rebased on 4.19) - Updated change log to describe the use-case more clearly. - No changes to 0001,0002 code since the v7 referenced below. - 0003 is a new patch to enable ALSA defines that have been disabled for kernel between 4.9 and 4.19. - Minor merge conflict resolution in 0004. - Added SPDX to new files. References: https://lkml.org/lkml/2018/11/2/169 https://www.mail-archive.com/linux-media@vger.kernel.org/msg105854.html Shuah Khan (4): media: Media Device Allocator API media: change au0828 to use Media Device Allocator API media: media.h: Enable ALSA MEDIA_INTF_T* interface types sound/usb: Use Media Controller API to share media resources Documentation/media/kapi/mc-core.rst | 41 ++++ drivers/media/Makefile | 4 + drivers/media/media-dev-allocator.c | 132 ++++++++++ drivers/media/usb/au0828/au0828-core.c | 12 +- drivers/media/usb/au0828/au0828.h | 1 + include/media/media-dev-allocator.h | 53 ++++ include/uapi/linux/media.h | 25 +- sound/usb/Kconfig | 4 + sound/usb/Makefile | 2 + sound/usb/card.c | 14 ++ sound/usb/card.h | 3 + sound/usb/media.c | 321 +++++++++++++++++++++++++ sound/usb/media.h | 74 ++++++ sound/usb/mixer.h | 3 + sound/usb/pcm.c | 29 ++- sound/usb/quirks-table.h | 1 + sound/usb/stream.c | 2 + sound/usb/usbaudio.h | 6 + 18 files changed, 705 insertions(+), 22 deletions(-) create mode 100644 drivers/media/media-dev-allocator.c create mode 100644 include/media/media-dev-allocator.h create mode 100644 sound/usb/media.c create mode 100644 sound/usb/media.h -- 2.17.1

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[alsa-devel] Possible ALSA loopback bug?
by Timothy Pearson 23 Jan '19

23 Jan '19

I've been running a dedicated machine with ALSA loopback, feeding darkice on the capture side and (currently) using mplayer to provide audio on the playback side. I've seen sporadic hangs for a couple of years on this setup, but recently with the switch to mplayer I have additional information that points more to ALSA than the other applications. When the audio hangs, mplayer prints a continuous stream of: "Audio device got stuck!" The only way to "unstick" the audio stream is to terminate and restart mplayer. Restarting darkice does not fix the stuck stream. There is no pulseaudio on this sytem. What would be the next steps to debug this setup? It's shown up on two entirely different installs at this point (Ubuntu 14.04 and Debian 9). Thanks! -- Timothy Pearson Raptor Engineering +1 (415) 727-8645 (direct line) +1 (512) 690-0200 (switchboard) https://www.raptorengineering.com

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[alsa-devel] [PATCH v3 0/3] ASoC: Intel: Update broxton machine driver
by Yong Zhi 23 Jan '19

23 Jan '19

Hello, This patchset is an update to the following v2 posted by Naveen last year: https://patchwork.kernel.org/patch/10335295/ https://patchwork.kernel.org/patch/10335293/ https://patchwork.kernel.org/patch/10335297/ 1/5 and 5/5 of original series were merged. The changes have been tested for both bxt_da7219_max98357a and glk_da7219_max98357a. Yong Zhi (3): ASoC: Intel: Headset button support in broxton machine driver ASoC: Intel: Add Geminilake Dialog Maxim machine driver ASoC: Intel: glk: Add DAI links for Multi-Playback sound/soc/intel/boards/bxt_da7219_max98357a.c | 101 +++++++++++++++++++++++--- 1 file changed, 89 insertions(+), 12 deletions(-) -- 2.7.4

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[alsa-devel] [PATCH] ALSA: proc: Avoid possible leaks of snd_info_entry objects
by Takashi Iwai 23 Jan '19

23 Jan '19

This patch changes the parent pointer assignment of snd_info_entry object to be always non-NULL. More specifically,check the parent argument in snd_info_create_module_entry() & co, and assign snd_proc_root if NULL is passed there. This assures that the proc object is always freed when the root is freed, so avoid possible memory leaks. For example, some error paths (e.g. snd_info_register() error at snd_minor_info_init()) may leave snd_info_entry object although the proc file itself is freed. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/info.c | 12 ++++++++++-- 1 file changed, 10 insertions(+), 2 deletions(-) diff --git a/sound/core/info.c b/sound/core/info.c index fe502bc5e6d2..2dfb6389c084 100644 --- a/sound/core/info.c +++ b/sound/core/info.c @@ -741,7 +741,11 @@ struct snd_info_entry *snd_info_create_module_entry(struct module * module, const char *name, struct snd_info_entry *parent) { - struct snd_info_entry *entry = snd_info_create_entry(name, parent); + struct snd_info_entry *entry; + + if (!parent) + parent = snd_proc_root; + entry = snd_info_create_entry(name, parent); if (entry) entry->module = module; return entry; @@ -762,7 +766,11 @@ struct snd_info_entry *snd_info_create_card_entry(struct snd_card *card, const char *name, struct snd_info_entry * parent) { - struct snd_info_entry *entry = snd_info_create_entry(name, parent); + struct snd_info_entry *entry; + + if (!parent) + parent = card->proc_root; + entry = snd_info_create_entry(name, parent); if (entry) { entry->module = card->module; entry->card = card; -- 2.16.4

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[alsa-devel] [PATCH] ALSA: pcm: Simplify proc file destruction
by Takashi Iwai 23 Jan '19

23 Jan '19

The proc files are recursively freed by calling with the root snd_info_entry object, so we don't have to keep each object for releasing one by one. Move the release of the PCM stream proc root at the beginning, so that we can remove the redundant code and resource. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- include/sound/pcm.h | 11 --------- sound/core/pcm.c | 66 +++++++++---------------------------------------- sound/core/pcm_memory.c | 16 ++---------- 3 files changed, 13 insertions(+), 80 deletions(-) diff --git a/include/sound/pcm.h b/include/sound/pcm.h index a20d3a48df00..eae6d2b82d7a 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -481,15 +481,6 @@ struct snd_pcm_substream { #endif #ifdef CONFIG_SND_VERBOSE_PROCFS struct snd_info_entry *proc_root; - struct snd_info_entry *proc_info_entry; - struct snd_info_entry *proc_hw_params_entry; - struct snd_info_entry *proc_sw_params_entry; - struct snd_info_entry *proc_status_entry; - struct snd_info_entry *proc_prealloc_entry; - struct snd_info_entry *proc_prealloc_max_entry; -#ifdef CONFIG_SND_PCM_XRUN_DEBUG - struct snd_info_entry *proc_xrun_injection_entry; -#endif #endif /* CONFIG_SND_VERBOSE_PROCFS */ /* misc flags */ unsigned int hw_opened: 1; @@ -511,10 +502,8 @@ struct snd_pcm_str { #endif #ifdef CONFIG_SND_VERBOSE_PROCFS struct snd_info_entry *proc_root; - struct snd_info_entry *proc_info_entry; #ifdef CONFIG_SND_PCM_XRUN_DEBUG unsigned int xrun_debug; /* 0 = disabled, 1 = verbose, 2 = stacktrace */ - struct snd_info_entry *proc_xrun_debug_entry; #endif #endif struct snd_kcontrol *chmap_kctl; /* channel-mapping controls */ diff --git a/sound/core/pcm.c b/sound/core/pcm.c index ca1ea3cf9350..bca0bdf3e33c 100644 --- a/sound/core/pcm.c +++ b/sound/core/pcm.c @@ -536,12 +536,9 @@ static int snd_pcm_stream_proc_init(struct snd_pcm_str *pstr) entry = snd_info_create_card_entry(pcm->card, "info", pstr->proc_root); if (entry) { snd_info_set_text_ops(entry, pstr, snd_pcm_stream_proc_info_read); - if (snd_info_register(entry) < 0) { + if (snd_info_register(entry) < 0) snd_info_free_entry(entry); - entry = NULL; - } } - pstr->proc_info_entry = entry; #ifdef CONFIG_SND_PCM_XRUN_DEBUG entry = snd_info_create_card_entry(pcm->card, "xrun_debug", @@ -551,24 +548,15 @@ static int snd_pcm_stream_proc_init(struct snd_pcm_str *pstr) entry->c.text.write = snd_pcm_xrun_debug_write; entry->mode |= 0200; entry->private_data = pstr; - if (snd_info_register(entry) < 0) { + if (snd_info_register(entry) < 0) snd_info_free_entry(entry); - entry = NULL; - } } - pstr->proc_xrun_debug_entry = entry; #endif return 0; } static int snd_pcm_stream_proc_done(struct snd_pcm_str *pstr) { -#ifdef CONFIG_SND_PCM_XRUN_DEBUG - snd_info_free_entry(pstr->proc_xrun_debug_entry); - pstr->proc_xrun_debug_entry = NULL; -#endif - snd_info_free_entry(pstr->proc_info_entry); - pstr->proc_info_entry = NULL; snd_info_free_entry(pstr->proc_root); pstr->proc_root = NULL; return 0; @@ -597,45 +585,33 @@ static int snd_pcm_substream_proc_init(struct snd_pcm_substream *substream) if (entry) { snd_info_set_text_ops(entry, substream, snd_pcm_substream_proc_info_read); - if (snd_info_register(entry) < 0) { + if (snd_info_register(entry) < 0) snd_info_free_entry(entry); - entry = NULL; - } } - substream->proc_info_entry = entry; entry = snd_info_create_card_entry(card, "hw_params", substream->proc_root); if (entry) { snd_info_set_text_ops(entry, substream, snd_pcm_substream_proc_hw_params_read); - if (snd_info_register(entry) < 0) { + if (snd_info_register(entry) < 0) snd_info_free_entry(entry); - entry = NULL; - } } - substream->proc_hw_params_entry = entry; entry = snd_info_create_card_entry(card, "sw_params", substream->proc_root); if (entry) { snd_info_set_text_ops(entry, substream, snd_pcm_substream_proc_sw_params_read); - if (snd_info_register(entry) < 0) { + if (snd_info_register(entry) < 0) snd_info_free_entry(entry); - entry = NULL; - } } - substream->proc_sw_params_entry = entry; entry = snd_info_create_card_entry(card, "status", substream->proc_root); if (entry) { snd_info_set_text_ops(entry, substream, snd_pcm_substream_proc_status_read); - if (snd_info_register(entry) < 0) { + if (snd_info_register(entry) < 0) snd_info_free_entry(entry); - entry = NULL; - } } - substream->proc_status_entry = entry; #ifdef CONFIG_SND_PCM_XRUN_DEBUG entry = snd_info_create_card_entry(card, "xrun_injection", @@ -645,40 +621,18 @@ static int snd_pcm_substream_proc_init(struct snd_pcm_substream *substream) entry->c.text.read = NULL; entry->c.text.write = snd_pcm_xrun_injection_write; entry->mode = S_IFREG | 0200; - if (snd_info_register(entry) < 0) { + if (snd_info_register(entry) < 0) snd_info_free_entry(entry); - entry = NULL; - } } - substream->proc_xrun_injection_entry = entry; #endif /* CONFIG_SND_PCM_XRUN_DEBUG */ return 0; } -static int snd_pcm_substream_proc_done(struct snd_pcm_substream *substream) -{ - snd_info_free_entry(substream->proc_info_entry); - substream->proc_info_entry = NULL; - snd_info_free_entry(substream->proc_hw_params_entry); - substream->proc_hw_params_entry = NULL; - snd_info_free_entry(substream->proc_sw_params_entry); - substream->proc_sw_params_entry = NULL; - snd_info_free_entry(substream->proc_status_entry); - substream->proc_status_entry = NULL; -#ifdef CONFIG_SND_PCM_XRUN_DEBUG - snd_info_free_entry(substream->proc_xrun_injection_entry); - substream->proc_xrun_injection_entry = NULL; -#endif - snd_info_free_entry(substream->proc_root); - substream->proc_root = NULL; - return 0; -} #else /* !CONFIG_SND_VERBOSE_PROCFS */ static inline int snd_pcm_stream_proc_init(struct snd_pcm_str *pstr) { return 0; } static inline int snd_pcm_stream_proc_done(struct snd_pcm_str *pstr) { return 0; } static inline int snd_pcm_substream_proc_init(struct snd_pcm_substream *substream) { return 0; } -static inline int snd_pcm_substream_proc_done(struct snd_pcm_substream *substream) { return 0; } #endif /* CONFIG_SND_VERBOSE_PROCFS */ static const struct attribute_group *pcm_dev_attr_groups[]; @@ -911,15 +865,17 @@ static void snd_pcm_free_stream(struct snd_pcm_str * pstr) #if IS_ENABLED(CONFIG_SND_PCM_OSS) struct snd_pcm_oss_setup *setup, *setupn; #endif + + /* free all proc files under the stream */ + snd_pcm_stream_proc_done(pstr); + substream = pstr->substream; while (substream) { substream_next = substream->next; snd_pcm_timer_done(substream); - snd_pcm_substream_proc_done(substream); kfree(substream); substream = substream_next; } - snd_pcm_stream_proc_done(pstr); #if IS_ENABLED(CONFIG_SND_PCM_OSS) for (setup = pstr->oss.setup_list; setup; setup = setupn) { setupn = setup->next; diff --git a/sound/core/pcm_memory.c b/sound/core/pcm_memory.c index 4b5356a10315..9a98bc61461f 100644 --- a/sound/core/pcm_memory.c +++ b/sound/core/pcm_memory.c @@ -93,12 +93,6 @@ static void snd_pcm_lib_preallocate_dma_free(struct snd_pcm_substream *substream int snd_pcm_lib_preallocate_free(struct snd_pcm_substream *substream) { snd_pcm_lib_preallocate_dma_free(substream); -#ifdef CONFIG_SND_VERBOSE_PROCFS - snd_info_free_entry(substream->proc_prealloc_max_entry); - substream->proc_prealloc_max_entry = NULL; - snd_info_free_entry(substream->proc_prealloc_entry); - substream->proc_prealloc_entry = NULL; -#endif return 0; } @@ -203,21 +197,15 @@ static inline void preallocate_info_init(struct snd_pcm_substream *substream) entry->c.text.write = snd_pcm_lib_preallocate_proc_write; entry->mode |= 0200; entry->private_data = substream; - if (snd_info_register(entry) < 0) { + if (snd_info_register(entry) < 0) snd_info_free_entry(entry); - entry = NULL; - } } - substream->proc_prealloc_entry = entry; if ((entry = snd_info_create_card_entry(substream->pcm->card, "prealloc_max", substream->proc_root)) != NULL) { entry->c.text.read = snd_pcm_lib_preallocate_max_proc_read; entry->private_data = substream; - if (snd_info_register(entry) < 0) { + if (snd_info_register(entry) < 0) snd_info_free_entry(entry); - entry = NULL; - } } - substream->proc_prealloc_max_entry = entry; } #else /* !CONFIG_SND_VERBOSE_PROCFS */ -- 2.16.4

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[alsa-devel] [PATCH v5 0/8] ASoC: Add support to WCD9335 Audio Codec
by Srinivas Kandagatla 23 Jan '19

23 Jan '19

Thankyou for reviewing v4 patchset, here is v5 addressing comments from v4. Major change is moving the driver out of MFD as there are no immediate plans to add support to other drivers on this codec. Qualcomm WCD9335 Codec is a standalone Hi-Fi audio codec IC. It is integrated in multiple Qualcomm SoCs like: MSM8996, MSM8976, and MSM8956 chipsets. WCD9335 had multiple functional blocks, like: Soundwire controller, interrupt mux, pin controller, Audio codec, MBHC, MAD(Mic activity Detection), Ultrasonic proximity and pen detection, Battery-voltage monitoring and Codec processing engine. Currently this patchset has been only tested with SLIMbus interface due to hardware avaiablity, but it can be easily made to work with both SLIMbus and I2C/I2S. This patchset adds very basic playback and capture support without much fancy features. New features will be added once the basic support is in. This patchset is tested on top of linux-next on DB820c for both playback , capture paths and MBHC. Some parts of the code has been inherited from Qualcomm andriod kernel, so credits to various authors. WCD9335 can be interfaced via I2S/I2C or SLIMbus. Here is our integration branch incase someone want to try this out: https://git.linaro.org/landing-teams/working/qualcomm/kernel.git/log/?h=int… Thanks, Srini Changes since v4 (https://lkml.org/lkml/2018/9/17/21/) - Move the driver out of MFD as discussed with Lee - defer support to MBHC for more testing! Srinivas Kandagatla (8): ASoC: dt-bindings: update wcd9335 bindings. ASoC: wcd9335: add support to wcd9335 codec ASoC: wcd9335: add CLASS-H Controller support ASoC: wcd9335: add basic controls ASoC: wcd9335: add playback dapm widgets ASoC: wcd9335: add capture dapm widgets ASoC: wcd9335: add audio routings ASoC: apq8096: add slim support .../bindings/sound/qcom,wcd9335.txt | 10 +- sound/soc/codecs/Kconfig | 6 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/wcd-clsh-v2.c | 576 ++ sound/soc/codecs/wcd-clsh-v2.h | 49 + sound/soc/codecs/wcd9335.c | 5275 +++++++++++++++++ sound/soc/codecs/wcd9335.h | 640 ++ sound/soc/qcom/apq8096.c | 71 +- 8 files changed, 6623 insertions(+), 6 deletions(-) create mode 100644 sound/soc/codecs/wcd-clsh-v2.c create mode 100644 sound/soc/codecs/wcd-clsh-v2.h create mode 100644 sound/soc/codecs/wcd9335.c create mode 100644 sound/soc/codecs/wcd9335.h -- 2.20.1

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