August 2018 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] Possibility of routing at the kernel level
by Mark Farrugia 04 Sep '18

04 Sep '18

Hi all, I am currently using the ALSA loopback driver to create a "virtual" sound card to abstract underlying real sound cards. I capture from the loopback in user-space, and then subsequently playback to a real sound card. The issue with this is that there is a significant latency introduced (~100ms) due to the number of transitions of the stream from user to kernel space. I am seeking a way to avoid this. My question is this: is there any way of routing PCM substreams at the kernel level, so that we can avoid the need to pipe audio back up to user-space with the loopback capturer? If there is no current solution that would achieve this, I would proposition to design and implement a system that uses a top-level ALSA driver, which then routes dynamically to one or many subscribed lower-level "audio path" drivers. The system would also have the option to loopback upon itself. Does anyone see any reason that I couldn't (or shouldn't) attempt such a solution? All feedback is welcome. Regards, Mark

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[alsa-devel] [PATCH v2] sound: Convert to using %pOFn instead of device_node.name
by Rob Herring 03 Sep '18

03 Sep '18

In preparation to remove the node name pointer from struct device_node, convert printf users to use the %pOFn format specifier. Cc: Johannes Berg <johannes(a)sipsolutions.net> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: linuxppc-dev(a)lists.ozlabs.org Cc: alsa-devel(a)alsa-project.org Signed-off-by: Rob Herring <robh(a)kernel.org> --- v2: - Fix error in strncmp sound/aoa/soundbus/i2sbus/core.c | 15 ++++++++------- 1 file changed, 8 insertions(+), 7 deletions(-) diff --git a/sound/aoa/soundbus/i2sbus/core.c b/sound/aoa/soundbus/i2sbus/core.c index 000b58522106..bd7c5029fc59 100644 --- a/sound/aoa/soundbus/i2sbus/core.c +++ b/sound/aoa/soundbus/i2sbus/core.c @@ -157,18 +157,19 @@ static int i2sbus_add_dev(struct macio_dev *macio, struct device_node *child = NULL, *sound = NULL; struct resource *r; int i, layout = 0, rlen, ok = force; - static const char *rnames[] = { "i2sbus: %s (control)", - "i2sbus: %s (tx)", - "i2sbus: %s (rx)" }; + char node_name[6]; + static const char *rnames[] = { "i2sbus: %pOFn (control)", + "i2sbus: %pOFn (tx)", + "i2sbus: %pOFn (rx)" }; static irq_handler_t ints[] = { i2sbus_bus_intr, i2sbus_tx_intr, i2sbus_rx_intr }; - if (strlen(np->name) != 5) + if (snprintf(node_name, sizeof(node_name), "%pOFn", np) != 5) return 0; - if (strncmp(np->name, "i2s-", 4)) + if (strncmp(node_name, "i2s-", 4)) return 0; dev = kzalloc(sizeof(struct i2sbus_dev), GFP_KERNEL); @@ -228,13 +229,13 @@ static int i2sbus_add_dev(struct macio_dev *macio, dev->sound.pcmid = -1; dev->macio = macio; dev->control = control; - dev->bus_number = np->name[4] - 'a'; + dev->bus_number = node_name[4] - 'a'; INIT_LIST_HEAD(&dev->sound.codec_list); for (i = aoa_resource_i2smmio; i <= aoa_resource_rxdbdma; i++) { dev->interrupts[i] = -1; snprintf(dev->rnames[i], sizeof(dev->rnames[i]), - rnames[i], np->name); + rnames[i], np); } for (i = aoa_resource_i2smmio; i <= aoa_resource_rxdbdma; i++) { int irq = irq_of_parse_and_map(np, i); -- 2.17.1

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[alsa-devel] [PATCH] ASoC: core: Don't schedule DAPM work if already in target state
by Jon Hunter 03 Sep '18

03 Sep '18

When dapm_power_widgets() is called, the dapm_pre_sequence_async() and dapm_post_sequence_async() functions are scheduled for all DAPM contexts (apart from the card DAPM context) regardless of whether the DAPM context is already in the desired state. The overhead of this is not insignificant and the more DAPM contexts there are the more overhead there is. For example, on the Tegra124 Jetson TK1, when profiling the time taken to execute the dapm_power_widgets() the following times were observed. Times for function dapm_power_widgets() are (us): Min 23, Ave 190, Max 434, Count 39 Here 'Count' is the number of times that dapm_power_widgets() has been called. Please note that the above time were measured using ktime_get() to log the time on entry and exit from dapm_power_widgets(). So it should be noted that these times may not be purely the time take to execute this function if it is preempted. However, after applying this patch and measuring the time taken to execute dapm_power_widgets() again a significant improvement is seen as shown below. Times for function dapm_power_widgets() are (us): Min 4, Ave 16, Max 82, Count 39 Therefore, optimise the dapm_power_widgets() function by only scheduling the dapm_pre/post_sequence_async() work if the DAPM context is not in the desired state. Signed-off-by: Jon Hunter <jonathanh(a)nvidia.com> --- sound/soc/soc-dapm.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 461d951917c0..2e9231aeabb9 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1953,7 +1953,7 @@ static int dapm_power_widgets(struct snd_soc_card *card, int event) dapm_pre_sequence_async(&card->dapm, 0); /* Run other bias changes in parallel */ list_for_each_entry(d, &card->dapm_list, list) { - if (d != &card->dapm) + if (d != &card->dapm && d->bias_level != d->target_bias_level) async_schedule_domain(dapm_pre_sequence_async, d, &async_domain); } @@ -1977,7 +1977,7 @@ static int dapm_power_widgets(struct snd_soc_card *card, int event) /* Run all the bias changes in parallel */ list_for_each_entry(d, &card->dapm_list, list) { - if (d != &card->dapm) + if (d != &card->dapm && d->bias_level != d->target_bias_level) async_schedule_domain(dapm_post_sequence_async, d, &async_domain); } -- 2.7.4

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[alsa-devel] [PATCH v2 0/2] Add support for MikroElektronika PROTO codec
by Codrin Ciubotariu 03 Sep '18

03 Sep '18

This patchset is based on Florian's initial PROTO driver: https://lkml.org/lkml/2013/5/22/342 My changes made the driver more generic, so that we could connect other I2S CPU DAIs, by using device-tree. Tested with sama5d2 I2S driver. Changes in v2: - removed unnecessary code that added constraints for the fixed SYSLK; - fixed errors reported by kbuild; - adjusted "mikroe" in vendor prefixes; - updated description in Kconfig; Codrin Ciubotariu (2): ASoC: Add driver for PROTO Audio CODEC (with a WM8731) ASoC: mikroe-proto: dt-bindings: add DT bindings for PROTO board .../bindings/sound/mikroe,mikroe-proto.txt | 23 +++ .../devicetree/bindings/vendor-prefixes.txt | 1 + sound/soc/atmel/Kconfig | 11 ++ sound/soc/atmel/Makefile | 2 + sound/soc/atmel/mikroe-proto.c | 165 ++++++++++++++++++ 5 files changed, 202 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/mikroe,mikroe-proto.txt create mode 100644 sound/soc/atmel/mikroe-proto.c -- 2.17.1

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[alsa-devel] ASoC: pxa: remove pxa2xx-pcm driver, which caused regression
by petrcvekcz＠gmail.com 02 Sep '18

02 Sep '18

From: Petr Cvek <petrcvekcz(a)gmail.com> The switch of PXA to dmaengine caused a regression for devices which are using SSP/I2S and probably AC97. The problem is in multiple allocation of DMA by function pxa2xx_soc_pcm_new() called by pxa-pcm-audio and (SSP/I2S/AC97). The regression was found on HTC Magician and after playback kernel generates an oops: WARNING: CPU: 0 PID: 892 at drivers/dma/dmaengine.c:788 dma_release_channel+0x44/0xa0 chan reference count 0 != 1 Modules linked in: ... snd_soc_magician snd_soc_uda1380 snd_soc_pxa2xx_i2s spi_pxa2xx_platform snd_soc_pxa2xx snd_soc_pxa_ssp snd_pxa2xx_lib snd_pcm_dmaengine i2c_pxa snd_soc_core snd_pcm snd_timer i2c_core snd ssp soundcore [last unloaded: pwm_bl] CPU: 0 PID: 892 Comm: aplay Tainted: G C 4.18.0-rc6-next-20180726-magician+ #16 Hardware name: HTC Magician [<c0107d90>] (unwind_backtrace) from [<c010594c>] (show_stack+0x10/0x14) [<c010594c>] (show_stack) from [<c0111680>] (__warn+0xd4/0xec) [<c0111680>] (__warn) from [<c01112d0>] (warn_slowpath_fmt+0x44/0x6c) [<c01112d0>] (warn_slowpath_fmt) from [<c034908c>] (dma_release_channel+0x44/0xa0) [<c034908c>] (dma_release_channel) from [<bf330528>] (snd_dmaengine_pcm_close_release_chan+0x44/0xb1c [snd_pcm_dmaengine]) [<bf330528>] (snd_dmaengine_pcm_close_release_chan [snd_pcm_dmaengine]) from [<bf26c29c>] (soc_pcm_components_close+0x8c/0xac [snd_soc_core]) [<bf26c29c>] (soc_pcm_components_close [snd_soc_core]) from [<bf26c3a0>] (soc_pcm_close+0xe4/0xd44 [snd_soc_core]) [<bf26c3a0>] (soc_pcm_close [snd_soc_core]) from [<bf167254>] (snd_pcm_release_substream+0x88/0x108 [snd_pcm]) [<bf167254>] (snd_pcm_release_substream [snd_pcm]) from [<bf167330>] (snd_pcm_release+0x5c/0xd2c [snd_pcm]) [<bf167330>] (snd_pcm_release [snd_pcm]) from [<c01c39e4>] (__fput+0xdc/0x1e8) [<c01c39e4>] (__fput) from [<c0129e7c>] (task_work_run+0xc4/0xd4) [<c0129e7c>] (task_work_run) from [<c01053ac>] (do_work_pending+0xc0/0xc8) [<c01053ac>] (do_work_pending) from [<c0101068>] (slow_work_pending+0xc/0x20) Exception stack(0xc2fdffb0 to 0xc2fdfff8) ffa0: 00000000 0154d1a0 01542a3c 00000000 ffc0: 0154d150 00000000 0009283c 00000006 000003f0 0154d4c8 00000000 0154d4c8 ffe0: b6ee7d70 bef13960 b6e547b8 b6cfe018 60000010 00000004 Removing bindings for snd_soc_magician fixes the problem (tested). Other machines should have the same problem if they use at least one of pxa-ssp-dai or pxa2xx-i2s. Considering pxa2xx-ac97 uses the function pxa2xx_soc_pcm_new() too it should produce the same problem (untested). After removal from I2S/SSP/AC97 machines there seems to be no user of pxa2xx-pcm, so remove the driver completely, change platform device in arch/arm/mach-pxa/devices.c to snd-soc-dummy and an example in the documentation. Signed-off-by: Petr Cvek <petrcvekcz(a)gmail.com> --- Documentation/sound/soc/machine.rst | 2 +- arch/arm/mach-pxa/devices.c | 2 +- sound/soc/pxa/Kconfig | 43 ++++++++++---------------- sound/soc/pxa/Makefile | 2 -- sound/soc/pxa/corgi.c | 1 - sound/soc/pxa/e740_wm9705.c | 2 -- sound/soc/pxa/e750_wm9705.c | 2 -- sound/soc/pxa/e800_wm9712.c | 2 -- sound/soc/pxa/em-x270.c | 2 -- sound/soc/pxa/hx4700.c | 1 - sound/soc/pxa/imote2.c | 1 - sound/soc/pxa/magician.c | 2 -- sound/soc/pxa/mioa701_wm9713.c | 2 -- sound/soc/pxa/palm27x.c | 2 -- sound/soc/pxa/poodle.c | 1 - sound/soc/pxa/pxa2xx-pcm.c | 48 ----------------------------- sound/soc/pxa/raumfeld.c | 2 -- sound/soc/pxa/spitz.c | 1 - sound/soc/pxa/tosa.c | 2 -- sound/soc/pxa/z2.c | 1 - sound/soc/pxa/zylonite.c | 3 -- 21 files changed, 18 insertions(+), 106 deletions(-) delete mode 100644 sound/soc/pxa/pxa2xx-pcm.c diff --git a/Documentation/sound/soc/machine.rst b/Documentation/sound/soc/machine.rst index 515c9444deaf..a65f41ade2ca 100644 --- a/Documentation/sound/soc/machine.rst +++ b/Documentation/sound/soc/machine.rst @@ -65,7 +65,7 @@ struct snd_soc_dai_link is used to set up each DAI in your machine. e.g. .stream_name = "WM8731", .cpu_dai_name = "pxa-is2-dai", .codec_dai_name = "wm8731-hifi", - .platform_name = "pxa-pcm-audio", + .platform_name = "snd-soc-dummy", .codec_name = "wm8713-codec.0-001a", .init = corgi_wm8731_init, .ops = &corgi_ops, diff --git a/arch/arm/mach-pxa/devices.c b/arch/arm/mach-pxa/devices.c index a24783a03827..82ee7f6c1dcf 100644 --- a/arch/arm/mach-pxa/devices.c +++ b/arch/arm/mach-pxa/devices.c @@ -371,7 +371,7 @@ struct platform_device pxa_device_asoc_ssp4 = { }; struct platform_device pxa_device_asoc_platform = { - .name = "pxa-pcm-audio", + .name = "snd-soc-dummy", .id = -1, }; diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig index 776e148b0aa2..d865c1041c6b 100644 --- a/sound/soc/pxa/Kconfig +++ b/sound/soc/pxa/Kconfig @@ -1,12 +1,3 @@ -config SND_PXA2XX_SOC - tristate "SoC Audio for the Intel PXA2xx chip" - depends on ARCH_PXA || COMPILE_TEST - select SND_PXA2XX_LIB - help - Say Y or M if you want to add support for codecs attached to - the PXA2xx AC97, I2S or SSP interface. You will also need - to select the audio interfaces to support below. - config SND_MMP_SOC bool "Soc Audio for Marvell MMP chips" depends on ARCH_MMP @@ -43,7 +34,7 @@ config SND_MMP_SOC_SSPA config SND_PXA2XX_SOC_CORGI tristate "SoC Audio support for Sharp Zaurus SL-C7x0" - depends on SND_PXA2XX_SOC && PXA_SHARP_C7xx && I2C + depends on PXA_SHARP_C7xx && I2C select SND_PXA2XX_SOC_I2S select SND_SOC_WM8731 help @@ -52,7 +43,7 @@ config SND_PXA2XX_SOC_CORGI config SND_PXA2XX_SOC_SPITZ tristate "SoC Audio support for Sharp Zaurus SL-Cxx00" - depends on SND_PXA2XX_SOC && PXA_SHARP_Cxx00 && I2C + depends on PXA_SHARP_Cxx00 && I2C select SND_PXA2XX_SOC_I2S select SND_SOC_WM8750 help @@ -61,7 +52,7 @@ config SND_PXA2XX_SOC_SPITZ config SND_PXA2XX_SOC_Z2 tristate "SoC Audio support for Zipit Z2" - depends on SND_PXA2XX_SOC && MACH_ZIPIT2 && I2C + depends on MACH_ZIPIT2 && I2C select SND_PXA2XX_SOC_I2S select SND_SOC_WM8750 help @@ -69,7 +60,7 @@ config SND_PXA2XX_SOC_Z2 config SND_PXA2XX_SOC_POODLE tristate "SoC Audio support for Poodle" - depends on SND_PXA2XX_SOC && MACH_POODLE && I2C + depends on MACH_POODLE && I2C select SND_PXA2XX_SOC_I2S select SND_SOC_WM8731 help @@ -78,7 +69,7 @@ config SND_PXA2XX_SOC_POODLE config SND_PXA2XX_SOC_TOSA tristate "SoC AC97 Audio support for Tosa" - depends on SND_PXA2XX_SOC && MACH_TOSA + depends on MACH_TOSA depends on MFD_TC6393XB select SND_PXA2XX_SOC_AC97 select SND_SOC_WM9712 @@ -88,7 +79,7 @@ config SND_PXA2XX_SOC_TOSA config SND_PXA2XX_SOC_E740 tristate "SoC AC97 Audio support for e740" - depends on SND_PXA2XX_SOC && MACH_E740 + depends on MACH_E740 select SND_SOC_WM9705 select SND_PXA2XX_SOC_AC97 help @@ -97,7 +88,7 @@ config SND_PXA2XX_SOC_E740 config SND_PXA2XX_SOC_E750 tristate "SoC AC97 Audio support for e750" - depends on SND_PXA2XX_SOC && MACH_E750 + depends on MACH_E750 select SND_SOC_WM9705 select SND_PXA2XX_SOC_AC97 help @@ -106,7 +97,7 @@ config SND_PXA2XX_SOC_E750 config SND_PXA2XX_SOC_E800 tristate "SoC AC97 Audio support for e800" - depends on SND_PXA2XX_SOC && MACH_E800 + depends on MACH_E800 select SND_SOC_WM9712 select SND_PXA2XX_SOC_AC97 help @@ -115,8 +106,7 @@ config SND_PXA2XX_SOC_E800 config SND_PXA2XX_SOC_EM_X270 tristate "SoC Audio support for CompuLab EM-x270, eXeda and CM-X300" - depends on SND_PXA2XX_SOC && (MACH_EM_X270 || MACH_EXEDA || \ - MACH_CM_X300) + depends on MACH_EM_X270 || MACH_EXEDA || MACH_CM_X300 select SND_PXA2XX_SOC_AC97 select SND_SOC_WM9712 help @@ -125,8 +115,7 @@ config SND_PXA2XX_SOC_EM_X270 config SND_PXA2XX_SOC_PALM27X bool "SoC Audio support for Palm T|X, T5, E2 and LifeDrive" - depends on SND_PXA2XX_SOC && (MACH_PALMLD || MACH_PALMTX || \ - MACH_PALMT5 || MACH_PALMTE2) + depends on MACH_PALMLD || MACH_PALMTX || MACH_PALMT5 || MACH_PALMTE2 select SND_PXA2XX_SOC_AC97 select SND_SOC_WM9712 help @@ -155,7 +144,7 @@ config SND_SOC_TTC_DKB config SND_SOC_ZYLONITE tristate "SoC Audio support for Marvell Zylonite" - depends on SND_PXA2XX_SOC && MACH_ZYLONITE + depends on MACH_ZYLONITE select SND_PXA2XX_SOC_AC97 select SND_PXA_SOC_SSP select SND_SOC_WM9713 @@ -165,7 +154,7 @@ config SND_SOC_ZYLONITE config SND_SOC_RAUMFELD tristate "SoC Audio support Raumfeld audio adapter" - depends on SND_PXA2XX_SOC && (MACH_RAUMFELD_SPEAKER || MACH_RAUMFELD_CONNECTOR) + depends on MACH_RAUMFELD_SPEAKER || MACH_RAUMFELD_CONNECTOR depends on I2C && SPI_MASTER select SND_PXA_SOC_SSP select SND_SOC_CS4270 @@ -175,7 +164,7 @@ config SND_SOC_RAUMFELD config SND_PXA2XX_SOC_HX4700 tristate "SoC Audio support for HP iPAQ hx4700" - depends on SND_PXA2XX_SOC && MACH_H4700 && I2C + depends on MACH_H4700 && I2C select SND_PXA2XX_SOC_I2S select SND_SOC_AK4641 help @@ -184,7 +173,7 @@ config SND_PXA2XX_SOC_HX4700 config SND_PXA2XX_SOC_MAGICIAN tristate "SoC Audio support for HTC Magician" - depends on SND_PXA2XX_SOC && MACH_MAGICIAN && I2C + depends on MACH_MAGICIAN && I2C select SND_PXA2XX_SOC_I2S select SND_PXA_SOC_SSP select SND_SOC_UDA1380 @@ -194,7 +183,7 @@ config SND_PXA2XX_SOC_MAGICIAN config SND_PXA2XX_SOC_MIOA701 tristate "SoC Audio support for MIO A701" - depends on SND_PXA2XX_SOC && MACH_MIOA701 + depends on MACH_MIOA701 select SND_PXA2XX_SOC_AC97 select SND_SOC_WM9713 help @@ -203,7 +192,7 @@ config SND_PXA2XX_SOC_MIOA701 config SND_PXA2XX_SOC_IMOTE2 tristate "SoC Audio support for IMote 2" - depends on SND_PXA2XX_SOC && MACH_INTELMOTE2 && I2C + depends on MACH_INTELMOTE2 && I2C select SND_PXA2XX_SOC_I2S select SND_SOC_WM8940 help diff --git a/sound/soc/pxa/Makefile b/sound/soc/pxa/Makefile index 5b265662f04f..93c0fdd041ed 100644 --- a/sound/soc/pxa/Makefile +++ b/sound/soc/pxa/Makefile @@ -1,13 +1,11 @@ # SPDX-License-Identifier: GPL-2.0 # PXA Platform Support -snd-soc-pxa2xx-objs := pxa2xx-pcm.o snd-soc-pxa2xx-ac97-objs := pxa2xx-ac97.o snd-soc-pxa2xx-i2s-objs := pxa2xx-i2s.o snd-soc-pxa-ssp-objs := pxa-ssp.o snd-soc-mmp-objs := mmp-pcm.o snd-soc-mmp-sspa-objs := mmp-sspa.o -obj-$(CONFIG_SND_PXA2XX_SOC) += snd-soc-pxa2xx.o obj-$(CONFIG_SND_PXA2XX_SOC_AC97) += snd-soc-pxa2xx-ac97.o obj-$(CONFIG_SND_PXA2XX_SOC_I2S) += snd-soc-pxa2xx-i2s.o obj-$(CONFIG_SND_PXA_SOC_SSP) += snd-soc-pxa-ssp.o diff --git a/sound/soc/pxa/corgi.c b/sound/soc/pxa/corgi.c index 054e0d65db9d..d36193e45be9 100644 --- a/sound/soc/pxa/corgi.c +++ b/sound/soc/pxa/corgi.c @@ -265,7 +265,6 @@ static struct snd_soc_dai_link corgi_dai = { .stream_name = "WM8731", .cpu_dai_name = "pxa2xx-i2s", .codec_dai_name = "wm8731-hifi", - .platform_name = "pxa-pcm-audio", .codec_name = "wm8731.0-001b", .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS, diff --git a/sound/soc/pxa/e740_wm9705.c b/sound/soc/pxa/e740_wm9705.c index 8ab7032631b7..cfae28571ec7 100644 --- a/sound/soc/pxa/e740_wm9705.c +++ b/sound/soc/pxa/e740_wm9705.c @@ -90,7 +90,6 @@ static struct snd_soc_dai_link e740_dai[] = { .stream_name = "AC97 HiFi", .cpu_dai_name = "pxa2xx-ac97", .codec_dai_name = "wm9705-hifi", - .platform_name = "pxa-pcm-audio", .codec_name = "wm9705-codec", }, { @@ -98,7 +97,6 @@ static struct snd_soc_dai_link e740_dai[] = { .stream_name = "AC97 Aux", .cpu_dai_name = "pxa2xx-ac97-aux", .codec_dai_name = "wm9705-aux", - .platform_name = "pxa-pcm-audio", .codec_name = "wm9705-codec", }, }; diff --git a/sound/soc/pxa/e750_wm9705.c b/sound/soc/pxa/e750_wm9705.c index 82bcbbb1841b..21c1f504a141 100644 --- a/sound/soc/pxa/e750_wm9705.c +++ b/sound/soc/pxa/e750_wm9705.c @@ -73,7 +73,6 @@ static struct snd_soc_dai_link e750_dai[] = { .stream_name = "AC97 HiFi", .cpu_dai_name = "pxa2xx-ac97", .codec_dai_name = "wm9705-hifi", - .platform_name = "pxa-pcm-audio", .codec_name = "wm9705-codec", /* use ops to check startup state */ }, @@ -82,7 +81,6 @@ static struct snd_soc_dai_link e750_dai[] = { .stream_name = "AC97 Aux", .cpu_dai_name = "pxa2xx-ac97-aux", .codec_dai_name = "wm9705-aux", - .platform_name = "pxa-pcm-audio", .codec_name = "wm9705-codec", }, }; diff --git a/sound/soc/pxa/e800_wm9712.c b/sound/soc/pxa/e800_wm9712.c index 1ed8aa2348f1..2475366f0e0b 100644 --- a/sound/soc/pxa/e800_wm9712.c +++ b/sound/soc/pxa/e800_wm9712.c @@ -74,7 +74,6 @@ static struct snd_soc_dai_link e800_dai[] = { .stream_name = "AC97 HiFi", .cpu_dai_name = "pxa2xx-ac97", .codec_dai_name = "wm9712-hifi", - .platform_name = "pxa-pcm-audio", .codec_name = "wm9712-codec", }, { @@ -82,7 +81,6 @@ static struct snd_soc_dai_link e800_dai[] = { .stream_name = "AC97 Aux", .cpu_dai_name = "pxa2xx-ac97-aux", .codec_dai_name = "wm9712-aux", - .platform_name = "pxa-pcm-audio", .codec_name = "wm9712-codec", }, }; diff --git a/sound/soc/pxa/em-x270.c b/sound/soc/pxa/em-x270.c index e046770ce70e..76b466b3eaee 100644 --- a/sound/soc/pxa/em-x270.c +++ b/sound/soc/pxa/em-x270.c @@ -36,7 +36,6 @@ static struct snd_soc_dai_link em_x270_dai[] = { .stream_name = "AC97 HiFi", .cpu_dai_name = "pxa2xx-ac97", .codec_dai_name = "wm9712-hifi", - .platform_name = "pxa-pcm-audio", .codec_name = "wm9712-codec", }, { @@ -44,7 +43,6 @@ static struct snd_soc_dai_link em_x270_dai[] = { .stream_name = "AC97 Aux", .cpu_dai_name = "pxa2xx-ac97-aux", .codec_dai_name = "wm9712-aux", - .platform_name = "pxa-pcm-audio", .codec_name = "wm9712-codec", }, }; diff --git a/sound/soc/pxa/hx4700.c b/sound/soc/pxa/hx4700.c index 6cdef5d4954e..79ffeab616f7 100644 --- a/sound/soc/pxa/hx4700.c +++ b/sound/soc/pxa/hx4700.c @@ -144,7 +144,6 @@ static struct snd_soc_dai_link hx4700_dai = { .stream_name = "AK4641", .cpu_dai_name = "pxa2xx-i2s", .codec_dai_name = "ak4641-hifi", - .platform_name = "pxa-pcm-audio", .codec_name = "ak4641.0-0012", .init = hx4700_ak4641_init, .dai_fmt = SND_SOC_DAIFMT_MSB | SND_SOC_DAIFMT_NB_NF | diff --git a/sound/soc/pxa/imote2.c b/sound/soc/pxa/imote2.c index 78475376f971..84dff8d34472 100644 --- a/sound/soc/pxa/imote2.c +++ b/sound/soc/pxa/imote2.c @@ -51,7 +51,6 @@ static struct snd_soc_dai_link imote2_dai = { .stream_name = "WM8940", .cpu_dai_name = "pxa2xx-i2s", .codec_dai_name = "wm8940-hifi", - .platform_name = "pxa-pcm-audio", .codec_name = "wm8940-codec.0-0034", .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS, diff --git a/sound/soc/pxa/magician.c b/sound/soc/pxa/magician.c index 935a248e5bf6..a7e7fa08e0a4 100644 --- a/sound/soc/pxa/magician.c +++ b/sound/soc/pxa/magician.c @@ -296,7 +296,6 @@ static struct snd_soc_dai_link magician_dai[] = { .stream_name = "UDA1380 Playback", .cpu_dai_name = "pxa-ssp-dai.0", .codec_dai_name = "uda1380-hifi-playback", - .platform_name = "pxa-pcm-audio", .codec_name = "uda1380-codec.0-0018", .ops = &magician_playback_ops, }, @@ -305,7 +304,6 @@ static struct snd_soc_dai_link magician_dai[] = { .stream_name = "UDA1380 Capture", .cpu_dai_name = "pxa2xx-i2s", .codec_dai_name = "uda1380-hifi-capture", - .platform_name = "pxa-pcm-audio", .codec_name = "uda1380-codec.0-0018", .ops = &magician_capture_ops, } diff --git a/sound/soc/pxa/mioa701_wm9713.c b/sound/soc/pxa/mioa701_wm9713.c index 47052fe3f5f4..121dcb1def3c 100644 --- a/sound/soc/pxa/mioa701_wm9713.c +++ b/sound/soc/pxa/mioa701_wm9713.c @@ -150,7 +150,6 @@ static struct snd_soc_dai_link mioa701_dai[] = { .codec_dai_name = "wm9713-hifi", .codec_name = "wm9713-codec", .init = mioa701_wm9713_init, - .platform_name = "pxa-pcm-audio", .ops = &mioa701_ops, }, { @@ -159,7 +158,6 @@ static struct snd_soc_dai_link mioa701_dai[] = { .cpu_dai_name = "pxa2xx-ac97-aux", .codec_dai_name = "wm9713-aux", .codec_name = "wm9713-codec", - .platform_name = "pxa-pcm-audio", .ops = &mioa701_ops, }, }; diff --git a/sound/soc/pxa/palm27x.c b/sound/soc/pxa/palm27x.c index 97167048572d..490717c7a5ce 100644 --- a/sound/soc/pxa/palm27x.c +++ b/sound/soc/pxa/palm27x.c @@ -94,7 +94,6 @@ static struct snd_soc_dai_link palm27x_dai[] = { .cpu_dai_name = "pxa2xx-ac97", .codec_dai_name = "wm9712-hifi", .codec_name = "wm9712-codec", - .platform_name = "pxa-pcm-audio", .init = palm27x_ac97_init, }, { @@ -103,7 +102,6 @@ static struct snd_soc_dai_link palm27x_dai[] = { .cpu_dai_name = "pxa2xx-ac97-aux", .codec_dai_name = "wm9712-aux", .codec_name = "wm9712-codec", - .platform_name = "pxa-pcm-audio", }, }; diff --git a/sound/soc/pxa/poodle.c b/sound/soc/pxa/poodle.c index b6693f32fc02..728904266930 100644 --- a/sound/soc/pxa/poodle.c +++ b/sound/soc/pxa/poodle.c @@ -229,7 +229,6 @@ static struct snd_soc_dai_link poodle_dai = { .stream_name = "WM8731", .cpu_dai_name = "pxa2xx-i2s", .codec_dai_name = "wm8731-hifi", - .platform_name = "pxa-pcm-audio", .codec_name = "wm8731.0-001b", .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS, diff --git a/sound/soc/pxa/pxa2xx-pcm.c b/sound/soc/pxa/pxa2xx-pcm.c deleted file mode 100644 index 72eaaef1b426..000000000000 --- a/sound/soc/pxa/pxa2xx-pcm.c +++ /dev/null @@ -1,48 +0,0 @@ -/* - * linux/sound/arm/pxa2xx-pcm.c -- ALSA PCM interface for the Intel PXA2xx chip - * - * Author: Nicolas Pitre - * Created: Nov 30, 2004 - * Copyright: (C) 2004 MontaVista Software, 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. - */ - -#include <linux/dma-mapping.h> -#include <linux/module.h> -#include <linux/dmaengine.h> -#include <linux/of.h> - -#include <sound/core.h> -#include <sound/soc.h> -#include <sound/pxa2xx-lib.h> -#include <sound/dmaengine_pcm.h> - -static const struct snd_soc_component_driver pxa2xx_soc_platform = { - .ops = &pxa2xx_pcm_ops, - .pcm_new = pxa2xx_soc_pcm_new, - .pcm_free = pxa2xx_pcm_free_dma_buffers, -}; - -static int pxa2xx_soc_platform_probe(struct platform_device *pdev) -{ - return devm_snd_soc_register_component(&pdev->dev, &pxa2xx_soc_platform, - NULL, 0); -} - -static struct platform_driver pxa_pcm_driver = { - .driver = { - .name = "pxa-pcm-audio", - }, - - .probe = pxa2xx_soc_platform_probe, -}; - -module_platform_driver(pxa_pcm_driver); - -MODULE_AUTHOR("Nicolas Pitre"); -MODULE_DESCRIPTION("Intel PXA2xx PCM DMA module"); -MODULE_LICENSE("GPL"); -MODULE_ALIAS("platform:pxa-pcm-audio"); diff --git a/sound/soc/pxa/raumfeld.c b/sound/soc/pxa/raumfeld.c index 111a907c4eb9..b2ae5aec0b83 100644 --- a/sound/soc/pxa/raumfeld.c +++ b/sound/soc/pxa/raumfeld.c @@ -205,7 +205,6 @@ static struct snd_soc_ops raumfeld_ak4104_ops = { .name = "CS4270", \ .stream_name = "CS4270", \ .cpu_dai_name = "pxa-ssp-dai.0", \ - .platform_name = "pxa-pcm-audio", \ .codec_dai_name = "cs4270-hifi", \ .codec_name = "cs4270.0-0048", \ .dai_fmt = SND_SOC_DAIFMT_I2S | \ @@ -220,7 +219,6 @@ static struct snd_soc_ops raumfeld_ak4104_ops = { .stream_name = "Playback", \ .cpu_dai_name = "pxa-ssp-dai.1", \ .codec_dai_name = "ak4104-hifi", \ - .platform_name = "pxa-pcm-audio", \ .dai_fmt = SND_SOC_DAIFMT_I2S | \ SND_SOC_DAIFMT_NB_NF | \ SND_SOC_DAIFMT_CBS_CFS, \ diff --git a/sound/soc/pxa/spitz.c b/sound/soc/pxa/spitz.c index 1671da648e95..57e4ecb9cbe2 100644 --- a/sound/soc/pxa/spitz.c +++ b/sound/soc/pxa/spitz.c @@ -262,7 +262,6 @@ static struct snd_soc_dai_link spitz_dai = { .stream_name = "WM8750", .cpu_dai_name = "pxa2xx-i2s", .codec_dai_name = "wm8750-hifi", - .platform_name = "pxa-pcm-audio", .codec_name = "wm8750.0-001b", .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS, diff --git a/sound/soc/pxa/tosa.c b/sound/soc/pxa/tosa.c index ae9c12e1ea2a..b2ddfeea6b85 100644 --- a/sound/soc/pxa/tosa.c +++ b/sound/soc/pxa/tosa.c @@ -188,7 +188,6 @@ static struct snd_soc_dai_link tosa_dai[] = { .stream_name = "AC97 HiFi", .cpu_dai_name = "pxa2xx-ac97", .codec_dai_name = "wm9712-hifi", - .platform_name = "pxa-pcm-audio", .codec_name = "wm9712-codec", .ops = &tosa_ops, }, @@ -197,7 +196,6 @@ static struct snd_soc_dai_link tosa_dai[] = { .stream_name = "AC97 Aux", .cpu_dai_name = "pxa2xx-ac97-aux", .codec_dai_name = "wm9712-aux", - .platform_name = "pxa-pcm-audio", .codec_name = "wm9712-codec", .ops = &tosa_ops, }, diff --git a/sound/soc/pxa/z2.c b/sound/soc/pxa/z2.c index 5b0eccd2b4dd..5e2307959b7b 100644 --- a/sound/soc/pxa/z2.c +++ b/sound/soc/pxa/z2.c @@ -162,7 +162,6 @@ static struct snd_soc_dai_link z2_dai = { .stream_name = "WM8750", .cpu_dai_name = "pxa2xx-i2s", .codec_dai_name = "wm8750-hifi", - .platform_name = "pxa-pcm-audio", .codec_name = "wm8750.0-001b", .init = z2_wm8750_init, .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | diff --git a/sound/soc/pxa/zylonite.c b/sound/soc/pxa/zylonite.c index 230eee450f45..753c828ade37 100644 --- a/sound/soc/pxa/zylonite.c +++ b/sound/soc/pxa/zylonite.c @@ -132,7 +132,6 @@ static struct snd_soc_dai_link zylonite_dai[] = { .name = "AC97", .stream_name = "AC97 HiFi", .codec_name = "wm9713-codec", - .platform_name = "pxa-pcm-audio", .cpu_dai_name = "pxa2xx-ac97", .codec_dai_name = "wm9713-hifi", .init = zylonite_wm9713_init, @@ -141,7 +140,6 @@ static struct snd_soc_dai_link zylonite_dai[] = { .name = "AC97 Aux", .stream_name = "AC97 Aux", .codec_name = "wm9713-codec", - .platform_name = "pxa-pcm-audio", .cpu_dai_name = "pxa2xx-ac97-aux", .codec_dai_name = "wm9713-aux", }, @@ -149,7 +147,6 @@ static struct snd_soc_dai_link zylonite_dai[] = { .name = "WM9713 Voice", .stream_name = "WM9713 Voice", .codec_name = "wm9713-codec", - .platform_name = "pxa-pcm-audio", .cpu_dai_name = "pxa-ssp-dai.2", .codec_dai_name = "wm9713-voice", .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | -- 2.18.0

3 10

[alsa-devel] [PATCH] ALSA: control: Make sure that id->index does not overflow in function snd_ctl_replace
by Xiao Yang 01 Sep '18

01 Sep '18

From: Young_X <YangX92(a)hotmail.com> The ALSA control code expects that the range of assigned indices to a control is continuous and does not overflow. Currently there are no checks to enforce this. If a control with a overflowing index range is created that control becomes effectively inaccessible and unremovable since snd_ctl_find_id() will not be able to find it. This patch adds a check that makes sure that controls with a overflowing index range can not be created. (same issue as CVE-2014-4656) Signed-off-by: Young_X <YangX92(a)hotmail.com> --- sound/core/control.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/sound/core/control.c b/sound/core/control.c index 9aa15bf..6435772 100644 --- a/sound/core/control.c +++ b/sound/core/control.c @@ -441,6 +441,11 @@ int snd_ctl_replace(struct snd_card *card, struct snd_kcontrol *kcontrol, goto error; } id = kcontrol->id; + if (id.index > UINT_MAX - kcontrol->count) { + ret = -EINVAL; + goto error; + } + down_write(&card->controls_rwsem); old = snd_ctl_find_id(card, &id); if (!old) { -- 2.7.4

2 1

[alsa-devel] Applied "ALSA: hda: move hda_codec.h to include/sound" to the asoc tree
by Mark Brown 31 Aug '18

31 Aug '18

The patch ALSA: hda: move hda_codec.h to include/sound has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From be57bfffb7b5ba72e1293643053f8861fcaa5163 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Date: Wed, 22 Aug 2018 15:24:57 -0500 Subject: [PATCH] ALSA: hda: move hda_codec.h to include/sound As suggested by Takashi, move this header file to make it easier to include from e.g. the Intel Skylake driver in follow-up patches Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- {sound/pci/hda => include/sound}/hda_codec.h | 0 sound/pci/hda/hda_auto_parser.c | 2 +- sound/pci/hda/hda_beep.h | 2 +- sound/pci/hda/hda_bind.c | 2 +- sound/pci/hda/hda_codec.c | 2 +- sound/pci/hda/hda_controller.h | 2 +- sound/pci/hda/hda_eld.c | 2 +- sound/pci/hda/hda_generic.c | 2 +- sound/pci/hda/hda_hwdep.c | 2 +- sound/pci/hda/hda_intel.c | 2 +- sound/pci/hda/hda_jack.c | 2 +- sound/pci/hda/hda_proc.c | 2 +- sound/pci/hda/hda_sysfs.c | 2 +- sound/pci/hda/hda_tegra.c | 2 +- sound/pci/hda/patch_analog.c | 2 +- sound/pci/hda/patch_ca0110.c | 2 +- sound/pci/hda/patch_ca0132.c | 2 +- sound/pci/hda/patch_cirrus.c | 2 +- sound/pci/hda/patch_cmedia.c | 2 +- sound/pci/hda/patch_conexant.c | 2 +- sound/pci/hda/patch_hdmi.c | 2 +- sound/pci/hda/patch_realtek.c | 2 +- sound/pci/hda/patch_si3054.c | 2 +- sound/pci/hda/patch_sigmatel.c | 2 +- sound/pci/hda/patch_via.c | 2 +- 25 files changed, 24 insertions(+), 24 deletions(-) rename {sound/pci/hda => include/sound}/hda_codec.h (100%) diff --git a/sound/pci/hda/hda_codec.h b/include/sound/hda_codec.h similarity index 100% rename from sound/pci/hda/hda_codec.h rename to include/sound/hda_codec.h diff --git a/sound/pci/hda/hda_auto_parser.c b/sound/pci/hda/hda_auto_parser.c index b9a6b66aeb0e..df0d636145f8 100644 --- a/sound/pci/hda/hda_auto_parser.c +++ b/sound/pci/hda/hda_auto_parser.c @@ -13,7 +13,7 @@ #include <linux/export.h> #include <linux/sort.h> #include <sound/core.h> -#include "hda_codec.h" +#include <sound/hda_codec.h> #include "hda_local.h" #include "hda_auto_parser.h" diff --git a/sound/pci/hda/hda_beep.h b/sound/pci/hda/hda_beep.h index d1a6a9c1329a..f1457c6b3969 100644 --- a/sound/pci/hda/hda_beep.h +++ b/sound/pci/hda/hda_beep.h @@ -9,7 +9,7 @@ #ifndef __SOUND_HDA_BEEP_H #define __SOUND_HDA_BEEP_H -#include "hda_codec.h" +#include <sound/hda_codec.h> #define HDA_BEEP_MODE_OFF 0 #define HDA_BEEP_MODE_ON 1 diff --git a/sound/pci/hda/hda_bind.c b/sound/pci/hda/hda_bind.c index d361bb77ca00..2222b47d4ec4 100644 --- a/sound/pci/hda/hda_bind.c +++ b/sound/pci/hda/hda_bind.c @@ -11,7 +11,7 @@ #include <linux/pm.h> #include <linux/pm_runtime.h> #include <sound/core.h> -#include "hda_codec.h" +#include <sound/hda_codec.h> #include "hda_local.h" /* diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 0a5085537034..ccbb0d12b8cc 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -27,7 +27,7 @@ #include <linux/pm.h> #include <linux/pm_runtime.h> #include <sound/core.h> -#include "hda_codec.h" +#include <sound/hda_codec.h> #include <sound/asoundef.h> #include <sound/tlv.h> #include <sound/initval.h> diff --git a/sound/pci/hda/hda_controller.h b/sound/pci/hda/hda_controller.h index a68e75b00ea3..55760e5231e6 100644 --- a/sound/pci/hda/hda_controller.h +++ b/sound/pci/hda/hda_controller.h @@ -20,7 +20,7 @@ #include <sound/core.h> #include <sound/pcm.h> #include <sound/initval.h> -#include "hda_codec.h" +#include <sound/hda_codec.h> #include <sound/hda_register.h> #define AZX_MAX_CODECS HDA_MAX_CODECS diff --git a/sound/pci/hda/hda_eld.c b/sound/pci/hda/hda_eld.c index ba7fe9b6655c..806b12ed44a2 100644 --- a/sound/pci/hda/hda_eld.c +++ b/sound/pci/hda/hda_eld.c @@ -27,7 +27,7 @@ #include <sound/core.h> #include <asm/unaligned.h> #include <sound/hda_chmap.h> -#include "hda_codec.h" +#include <sound/hda_codec.h> #include "hda_local.h" enum eld_versions { diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index 579984ecdec3..276150f29cda 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -32,7 +32,7 @@ #include <sound/core.h> #include <sound/jack.h> #include <sound/tlv.h> -#include "hda_codec.h" +#include <sound/hda_codec.h> #include "hda_local.h" #include "hda_auto_parser.h" #include "hda_jack.h" diff --git a/sound/pci/hda/hda_hwdep.c b/sound/pci/hda/hda_hwdep.c index cc009a4a3d1d..268bba6ec985 100644 --- a/sound/pci/hda/hda_hwdep.c +++ b/sound/pci/hda/hda_hwdep.c @@ -23,7 +23,7 @@ #include <linux/compat.h> #include <linux/nospec.h> #include <sound/core.h> -#include "hda_codec.h" +#include <sound/hda_codec.h> #include "hda_local.h" #include <sound/hda_hwdep.h> #include <sound/minors.h> diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 1b2ce304152a..c8fde95e2393 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -63,7 +63,7 @@ #include <linux/vgaarb.h> #include <linux/vga_switcheroo.h> #include <linux/firmware.h> -#include "hda_codec.h" +#include <sound/hda_codec.h> #include "hda_controller.h" #include "hda_intel.h" diff --git a/sound/pci/hda/hda_jack.c b/sound/pci/hda/hda_jack.c index a33234e04d4f..c499727920e6 100644 --- a/sound/pci/hda/hda_jack.c +++ b/sound/pci/hda/hda_jack.c @@ -15,7 +15,7 @@ #include <sound/core.h> #include <sound/control.h> #include <sound/jack.h> -#include "hda_codec.h" +#include <sound/hda_codec.h> #include "hda_local.h" #include "hda_auto_parser.h" #include "hda_jack.h" diff --git a/sound/pci/hda/hda_proc.c b/sound/pci/hda/hda_proc.c index c6b778b2580c..a65740419650 100644 --- a/sound/pci/hda/hda_proc.c +++ b/sound/pci/hda/hda_proc.c @@ -25,7 +25,7 @@ #include <linux/slab.h> #include <sound/core.h> #include <linux/module.h> -#include "hda_codec.h" +#include <sound/hda_codec.h> #include "hda_local.h" static int dump_coef = -1; diff --git a/sound/pci/hda/hda_sysfs.c b/sound/pci/hda/hda_sysfs.c index 6ec79c58d48d..c154b19a0c45 100644 --- a/sound/pci/hda/hda_sysfs.c +++ b/sound/pci/hda/hda_sysfs.c @@ -14,7 +14,7 @@ #include <linux/string.h> #include <linux/export.h> #include <sound/core.h> -#include "hda_codec.h" +#include <sound/hda_codec.h> #include "hda_local.h" #include <sound/hda_hwdep.h> #include <sound/minors.h> diff --git a/sound/pci/hda/hda_tegra.c b/sound/pci/hda/hda_tegra.c index 0621920f7617..4bc5232eac1c 100644 --- a/sound/pci/hda/hda_tegra.c +++ b/sound/pci/hda/hda_tegra.c @@ -35,7 +35,7 @@ #include <sound/core.h> #include <sound/initval.h> -#include "hda_codec.h" +#include <sound/hda_codec.h> #include "hda_controller.h" /* Defines for Nvidia Tegra HDA support */ diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index fd476fb40e1b..ebfd0be885b3 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -24,7 +24,7 @@ #include <linux/module.h> #include <sound/core.h> -#include "hda_codec.h" +#include <sound/hda_codec.h> #include "hda_local.h" #include "hda_auto_parser.h" #include "hda_beep.h" diff --git a/sound/pci/hda/patch_ca0110.c b/sound/pci/hda/patch_ca0110.c index c2d9ee9cfdc0..21d0f0610913 100644 --- a/sound/pci/hda/patch_ca0110.c +++ b/sound/pci/hda/patch_ca0110.c @@ -22,7 +22,7 @@ #include <linux/slab.h> #include <linux/module.h> #include <sound/core.h> -#include "hda_codec.h" +#include <sound/hda_codec.h> #include "hda_local.h" #include "hda_auto_parser.h" #include "hda_jack.h" diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c index 0166a3d7cd55..a585d0ec6d77 100644 --- a/sound/pci/hda/patch_ca0132.c +++ b/sound/pci/hda/patch_ca0132.c @@ -32,7 +32,7 @@ #include <linux/io.h> #include <linux/pci.h> #include <sound/core.h> -#include "hda_codec.h" +#include <sound/hda_codec.h> #include "hda_local.h" #include "hda_auto_parser.h" #include "hda_jack.h" diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c index a7f91be45194..64fa5a82bb9f 100644 --- a/sound/pci/hda/patch_cirrus.c +++ b/sound/pci/hda/patch_cirrus.c @@ -23,7 +23,7 @@ #include <linux/module.h> #include <sound/core.h> #include <sound/tlv.h> -#include "hda_codec.h" +#include <sound/hda_codec.h> #include "hda_local.h" #include "hda_auto_parser.h" #include "hda_jack.h" diff --git a/sound/pci/hda/patch_cmedia.c b/sound/pci/hda/patch_cmedia.c index 1b2195dd2b26..52642ba3e2c0 100644 --- a/sound/pci/hda/patch_cmedia.c +++ b/sound/pci/hda/patch_cmedia.c @@ -25,7 +25,7 @@ #include <linux/slab.h> #include <linux/module.h> #include <sound/core.h> -#include "hda_codec.h" +#include <sound/hda_codec.h> #include "hda_local.h" #include "hda_auto_parser.h" #include "hda_jack.h" diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index cfd4e4f97f8f..5592557fe50e 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -27,7 +27,7 @@ #include <sound/core.h> #include <sound/jack.h> -#include "hda_codec.h" +#include <sound/hda_codec.h> #include "hda_local.h" #include "hda_auto_parser.h" #include "hda_beep.h" diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index cb587dce67a9..67099cbb6be2 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -41,7 +41,7 @@ #include <sound/hdaudio.h> #include <sound/hda_i915.h> #include <sound/hda_chmap.h> -#include "hda_codec.h" +#include <sound/hda_codec.h> #include "hda_local.h" #include "hda_jack.h" diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 1d117f00d04d..6f3c8e888c2a 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -32,7 +32,7 @@ #include <linux/input.h> #include <sound/core.h> #include <sound/jack.h> -#include "hda_codec.h" +#include <sound/hda_codec.h> #include "hda_local.h" #include "hda_auto_parser.h" #include "hda_jack.h" diff --git a/sound/pci/hda/patch_si3054.c b/sound/pci/hda/patch_si3054.c index f63acb1b965c..c49d25bcd7f2 100644 --- a/sound/pci/hda/patch_si3054.c +++ b/sound/pci/hda/patch_si3054.c @@ -27,7 +27,7 @@ #include <linux/slab.h> #include <linux/module.h> #include <sound/core.h> -#include "hda_codec.h" +#include <sound/hda_codec.h> #include "hda_local.h" /* si3054 verbs */ diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 046705b4691a..d16a25a395c9 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -32,7 +32,7 @@ #include <linux/module.h> #include <sound/core.h> #include <sound/jack.h> -#include "hda_codec.h" +#include <sound/hda_codec.h> #include "hda_local.h" #include "hda_auto_parser.h" #include "hda_beep.h" diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index 6b9617aee0e6..9f6f13e25145 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -52,7 +52,7 @@ #include <linux/module.h> #include <sound/core.h> #include <sound/asoundef.h> -#include "hda_codec.h" +#include <sound/hda_codec.h> #include "hda_local.h" #include "hda_auto_parser.h" #include "hda_jack.h" -- 2.19.0.rc1

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[alsa-devel] [PATCH] ASoC: davinci-mcasp: Add support for FIFO usage caused delay reporting
by Peter Ujfalusi 31 Aug '18

31 Aug '18

McASP have write and read FIFO, each 64 words deep. >From the WFIFOS/RFIFOS registers we can read the amount of data currently in the FIFO which can be directly reported as delay. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- sound/soc/davinci/davinci-mcasp.c | 37 +++++++++++++++++++++++++++++++ 1 file changed, 37 insertions(+) diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index f7f546bdd312..1f8f4f3929f2 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -1062,6 +1062,42 @@ static int davinci_mcasp_calc_clk_div(struct davinci_mcasp *mcasp, return error_ppm; } +static inline u32 davinci_mcasp_tx_delay(struct davinci_mcasp *mcasp) +{ + if (!mcasp->txnumevt) + return 0; + + return mcasp_get_reg(mcasp, mcasp->fifo_base + MCASP_WFIFOSTS_OFFSET); +} + +static inline u32 davinci_mcasp_rx_delay(struct davinci_mcasp *mcasp) +{ + if (!mcasp->rxnumevt) + return 0; + + return mcasp_get_reg(mcasp, mcasp->fifo_base + MCASP_RFIFOSTS_OFFSET); +} + +static snd_pcm_sframes_t davinci_mcasp_delay( + struct snd_pcm_substream *substream, + struct snd_soc_dai *cpu_dai) +{ + struct davinci_mcasp *mcasp = snd_soc_dai_get_drvdata(cpu_dai); + u32 fifo_use; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + fifo_use = davinci_mcasp_tx_delay(mcasp); + else + fifo_use = davinci_mcasp_rx_delay(mcasp); + + /* + * Divide the used locations with the channel count to get the + * FIFO usage in samples (don't care about partial samples in the + * buffer). + */ + return fifo_use / substream->runtime->channels; +} + static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *cpu_dai) @@ -1386,6 +1422,7 @@ static const struct snd_soc_dai_ops davinci_mcasp_dai_ops = { .startup = davinci_mcasp_startup, .shutdown = davinci_mcasp_shutdown, .trigger = davinci_mcasp_trigger, + .delay = davinci_mcasp_delay, .hw_params = davinci_mcasp_hw_params, .set_fmt = davinci_mcasp_set_dai_fmt, .set_clkdiv = davinci_mcasp_set_clkdiv, -- Peter Texas Instruments Finland Oy, Porkkalankatu 22, 00180 Helsinki. Y-tunnus/Business ID: 0615521-4. Kotipaikka/Domicile: Helsinki

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[alsa-devel] [PATCH v2 1/2] ASoC: codecs: tas5720: add TAS5722 specific volume control
by Andrew F. Davis 31 Aug '18

31 Aug '18

From: Andreas Dannenberg <dannenberg(a)ti.com> The TAS5722 supports modifying volume in 0.25dB steps (as opposed to 0.5dB steps on the TAS5720). Introduce a custom mixer control that allows taking advantage of this finer output volume granularity. Also add custom getters/setters for access as the TAS5722 digital volume controls are split over two registers. Signed-off-by: Andreas Dannenberg <dannenberg(a)ti.com> Signed-off-by: Andrew F. Davis <afd(a)ti.com> --- Changes from v1: - Revise commit message - codec -> component sound/soc/codecs/tas5720.c | 88 ++++++++++++++++++++++++++++++++++---- 1 file changed, 80 insertions(+), 8 deletions(-) diff --git a/sound/soc/codecs/tas5720.c b/sound/soc/codecs/tas5720.c index ae3d032ac35a..3c469112477a 100644 --- a/sound/soc/codecs/tas5720.c +++ b/sound/soc/codecs/tas5720.c @@ -485,15 +485,56 @@ static const DECLARE_TLV_DB_RANGE(dac_analog_tlv, ); /* - * DAC digital volumes. From -103.5 to 24 dB in 0.5 dB steps. Note that - * setting the gain below -100 dB (register value <0x7) is effectively a MUTE - * as per device datasheet. + * DAC digital volumes. From -103.5 to 24 dB in 0.5 dB or 0.25 dB steps + * depending on the device. Note that setting the gain below -100 dB + * (register value <0x7) is effectively a MUTE as per device datasheet. + * + * Note that for the TAS5722 the digital volume controls are actually split + * over two registers, so we need custom getters/setters for access. */ -static DECLARE_TLV_DB_SCALE(dac_tlv, -10350, 50, 0); +static DECLARE_TLV_DB_SCALE(tas5720_dac_tlv, -10350, 50, 0); +static DECLARE_TLV_DB_SCALE(tas5722_dac_tlv, -10350, 25, 0); + +static int tas5722_volume_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol); + unsigned int val; + + snd_soc_component_read(component, TAS5720_VOLUME_CTRL_REG, &val); + ucontrol->value.integer.value[0] = val << 1; + + snd_soc_component_read(component, TAS5722_DIGITAL_CTRL2_REG, &val); + ucontrol->value.integer.value[0] |= val & TAS5722_VOL_CONTROL_LSB; + + return 0; +} + +static int tas5722_volume_set(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol); + unsigned int sel = ucontrol->value.integer.value[0]; + + snd_soc_component_write(component, TAS5720_VOLUME_CTRL_REG, sel >> 1); + snd_soc_component_update_bits(component, TAS5722_DIGITAL_CTRL2_REG, + TAS5722_VOL_CONTROL_LSB, sel); + + return 0; +} static const struct snd_kcontrol_new tas5720_snd_controls[] = { SOC_SINGLE_TLV("Speaker Driver Playback Volume", - TAS5720_VOLUME_CTRL_REG, 0, 0xff, 0, dac_tlv), + TAS5720_VOLUME_CTRL_REG, 0, 0xff, 0, tas5720_dac_tlv), + SOC_SINGLE_TLV("Speaker Driver Analog Gain", TAS5720_ANALOG_CTRL_REG, + TAS5720_ANALOG_GAIN_SHIFT, 3, 0, dac_analog_tlv), +}; + +static const struct snd_kcontrol_new tas5722_snd_controls[] = { + SOC_SINGLE_EXT_TLV("Speaker Driver Playback Volume", + 0, 0, 511, 0, + tas5722_volume_get, tas5722_volume_set, + tas5722_dac_tlv), SOC_SINGLE_TLV("Speaker Driver Analog Gain", TAS5720_ANALOG_CTRL_REG, TAS5720_ANALOG_GAIN_SHIFT, 3, 0, dac_analog_tlv), }; @@ -527,6 +568,23 @@ static const struct snd_soc_component_driver soc_component_dev_tas5720 = { .non_legacy_dai_naming = 1, }; +static const struct snd_soc_component_driver soc_component_dev_tas5722 = { + .probe = tas5720_codec_probe, + .remove = tas5720_codec_remove, + .suspend = tas5720_suspend, + .resume = tas5720_resume, + .controls = tas5722_snd_controls, + .num_controls = ARRAY_SIZE(tas5722_snd_controls), + .dapm_widgets = tas5720_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(tas5720_dapm_widgets), + .dapm_routes = tas5720_audio_map, + .num_dapm_routes = ARRAY_SIZE(tas5720_audio_map), + .idle_bias_on = 1, + .use_pmdown_time = 1, + .endianness = 1, + .non_legacy_dai_naming = 1, +}; + /* PCM rates supported by the TAS5720 driver */ #define TAS5720_RATES (SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 |\ SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000) @@ -613,9 +671,23 @@ static int tas5720_probe(struct i2c_client *client, dev_set_drvdata(dev, data); - ret = devm_snd_soc_register_component(&client->dev, - &soc_component_dev_tas5720, - tas5720_dai, ARRAY_SIZE(tas5720_dai)); + switch (id->driver_data) { + case TAS5720: + ret = devm_snd_soc_register_component(&client->dev, + &soc_component_dev_tas5720, + tas5720_dai, + ARRAY_SIZE(tas5720_dai)); + break; + case TAS5722: + ret = devm_snd_soc_register_component(&client->dev, + &soc_component_dev_tas5722, + tas5720_dai, + ARRAY_SIZE(tas5720_dai)); + break; + default: + dev_err(dev, "unexpected private driver data\n"); + return -EINVAL; + } if (ret < 0) { dev_err(dev, "failed to register component: %d\n", ret); return ret; -- 2.18.0

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[alsa-devel] Applied "ASoC: cs4265: SOC_SINGLE register value error fix" to the asoc tree
by Mark Brown 31 Aug '18

31 Aug '18

The patch ASoC: cs4265: SOC_SINGLE register value error fix has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 6f18bcdaa24bae39c746b57b95af19ff3c41b17f Mon Sep 17 00:00:00 2001 From: Matt Flax <flatmax(a)flatmax.org> Date: Thu, 30 Aug 2018 09:38:00 +1000 Subject: [PATCH] ASoC: cs4265: SOC_SINGLE register value error fix The cs4265 driver declares the "MMTLR Data Switch" register setting with a 0 register value rather then the 0x12 register (CS4265_SPDIF_CTL2). This incorrect value causes alsamixer to fault with the output : cannot load mixer controls: Input/output error This patch corrects the register value. alsamixer now runs. Signed-off-by: Matt Flax <flatmax(a)flatmax.org> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/cs4265.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/codecs/cs4265.c b/sound/soc/codecs/cs4265.c index 275677de669f..15b4ae04870f 100644 --- a/sound/soc/codecs/cs4265.c +++ b/sound/soc/codecs/cs4265.c @@ -157,8 +157,7 @@ static const struct snd_kcontrol_new cs4265_snd_controls[] = { SOC_SINGLE("Validity Bit Control Switch", CS4265_SPDIF_CTL2, 3, 1, 0), SOC_ENUM("SPDIF Mono/Stereo", spdif_mono_stereo_enum), - SOC_SINGLE("MMTLR Data Switch", 0, - 1, 1, 0), + SOC_SINGLE("MMTLR Data Switch", CS4265_SPDIF_CTL2, 0, 1, 0), SOC_ENUM("Mono Channel Select", spdif_mono_select_enum), SND_SOC_BYTES("C Data Buffer", CS4265_C_DATA_BUFF, 24), }; -- 2.19.0.rc1

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