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

[alsa-devel] [PATCH] ASoC: nau8825: erase pop noise by soft mute
by John Hsu 28 Nov '16

28 Nov '16

The patch is a solution to fix the following issue. Chrome OS Issue 54078: Chell_headphone pop back from S3 When the system resume from S3 and playing music, there is a pop noise happened. It looks like system changes the sound immediately from maximum volume to the volume configured by user in playback beginning. The sound signal changes sharply, and there is a DC offset in the signal to make pop noise. Therefore, the driver introduces the soft mute to avoid the side effect of the unstable signal. Signed-off-by: John Hsu <KCHSU0(a)nuvoton.com> --- sound/soc/codecs/nau8825.c | 52 +++++++++++++++++++++++++++++++++++++++++++++- sound/soc/codecs/nau8825.h | 3 +++ 2 files changed, 54 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/nau8825.c b/sound/soc/codecs/nau8825.c index e643be9..3bcba14 100644 --- a/sound/soc/codecs/nau8825.c +++ b/sound/soc/codecs/nau8825.c @@ -754,8 +754,9 @@ static void nau8825_xtalk_measure(struct nau8825 *nau8825) nau8825->imp_rms[NAU8825_XTALK_HPR_R2L], nau8825->imp_rms[NAU8825_XTALK_HPL_R2L]); dev_dbg(nau8825->dev, "cross talk sidetone: %x\n", sidetone); + nau8825->sidetone = (sidetone << 8) | sidetone; regmap_write(nau8825->regmap, NAU8825_REG_DAC_DGAIN_CTRL, - (sidetone << 8) | sidetone); + nau8825->sidetone); nau8825_xtalk_clean(nau8825); nau8825->xtalk_state = NAU8825_XTALK_DONE; break; @@ -1334,6 +1335,50 @@ int nau8825_enable_jack_detect(struct snd_soc_codec *codec, } EXPORT_SYMBOL_GPL(nau8825_enable_jack_detect); +/** + * nau8825_soft_mute - provide soft mute function + * + * @component: codec component + * @mute: 1 for mute; 0 for unmute + * + * Enable soft mute to gradually lower DAC volume to zero; + * Soft unmute will gradually increase DAC volume to volume setting. + * We found that signal at beginning is not stable in some application. + * That signal changes sharply will make pop noise; but the steps of soft + * mute is not enough in codec to cover the change period. Thus, the driver + * deffers 20ms to do soft unmute for the pop noise issue. + */ +int nau8825_soft_mute(struct snd_soc_codec *codec, int mute) +{ + struct nau8825 *nau8825 = snd_soc_codec_get_drvdata(codec); + + if (mute) { + cancel_delayed_work(&nau8825->softmute_work); + /* Disable sidetone to avoid it affect the mute function */ + regmap_write(nau8825->regmap, NAU8825_REG_DAC_DGAIN_CTRL, 0); + regmap_update_bits(nau8825->regmap, NAU8825_REG_MUTE_CTRL, + NAU8825_DAC_SOFT_MUTE, NAU8825_DAC_SOFT_MUTE); + } else { + /* Defer 20ms to do soft unmute to skip the unstable signal */ + schedule_delayed_work(&nau8825->softmute_work, + msecs_to_jiffies(20)); + } + + return 0; +} +EXPORT_SYMBOL_GPL(nau8825_soft_mute); + +static void nau8825_unmute_deferred(struct work_struct *work) +{ + struct nau8825 *nau8825 = container_of(work, + struct nau8825, softmute_work.work); + + /* Restore sidetone */ + regmap_write(nau8825->regmap, NAU8825_REG_DAC_DGAIN_CTRL, + nau8825->sidetone); + regmap_update_bits(nau8825->regmap, NAU8825_REG_MUTE_CTRL, + NAU8825_DAC_SOFT_MUTE, 0); +} static bool nau8825_is_jack_inserted(struct regmap *regmap) { @@ -2150,6 +2195,7 @@ static int nau8825_set_sysclk(struct snd_soc_codec *codec, int clk_id, static int nau8825_resume_setup(struct nau8825 *nau8825) { + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(nau8825->dapm); struct regmap *regmap = nau8825->regmap; /* Close clock when jack type detection at manual mode */ @@ -2169,6 +2215,8 @@ static int nau8825_resume_setup(struct nau8825 *nau8825) NAU8825_JACK_DET_DB_BYPASS, NAU8825_JACK_DET_DB_BYPASS); regmap_update_bits(regmap, NAU8825_REG_INTERRUPT_DIS_CTRL, NAU8825_IRQ_INSERT_DIS | NAU8825_IRQ_EJECT_DIS, 0); + /* Enable codec soft mute */ + nau8825_soft_mute(codec, 1); return 0; } @@ -2393,8 +2441,10 @@ static int nau8825_i2c_probe(struct i2c_client *i2c, */ nau8825->xtalk_state = NAU8825_XTALK_DONE; nau8825->xtalk_protect = false; + nau8825->sidetone = 0; sema_init(&nau8825->xtalk_sem, 1); INIT_WORK(&nau8825->xtalk_work, nau8825_xtalk_work); + INIT_DELAYED_WORK(&nau8825->softmute_work, nau8825_unmute_deferred); nau8825_print_device_properties(nau8825); diff --git a/sound/soc/codecs/nau8825.h b/sound/soc/codecs/nau8825.h index 1c63e2a..e15ea5c 100644 --- a/sound/soc/codecs/nau8825.h +++ b/sound/soc/codecs/nau8825.h @@ -433,6 +433,7 @@ struct nau8825 { struct snd_soc_dapm_context *dapm; struct snd_soc_jack *jack; struct clk *mclk; + struct delayed_work softmute_work; struct work_struct xtalk_work; struct semaphore xtalk_sem; int irq; @@ -459,10 +460,12 @@ struct nau8825 { int xtalk_event_mask; bool xtalk_protect; int imp_rms[NAU8825_XTALK_IMM]; + int sidetone; }; int nau8825_enable_jack_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack); +int nau8825_soft_mute(struct snd_soc_codec *codec, int mute); #endif /* __NAU8825_H__ */ -- 2.6.4

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[alsa-devel] [RFC 00/17] clk: Add per-controller locks to fix deadlocks
by Krzysztof Kozlowski 12 Nov '16

12 Nov '16

Hi, RFC, please, do not apply, maybe except patch #1 which is harmless. Introduction ============ The patchset brings new entity: clock controller representing a hardware block. The clock controller comes with its own prepare lock which is used then in many places. The idea is to fix the deadlock mentioned in commit 10ff4c5239a1 ("i2c: exynos5: Fix possible ABBA deadlock by keeping I2C clock prepared") and commit 34e81ad5f0b6 ("i2c: s3c2410: fix ABBA deadlock by keeping clock prepared"). Disclaimer ========== Request for comments, so: 1. Only exynos_defconfig builds, 2. A lot of FIXME/TODO note still, 3. Checkpatch not run, lines not aligned, 4. Other (non-exynos) drivers not converted, 5. Probably not yet bisectable, 6. Locking became quite complex. The previous one lock was simple. Inefficient and dead-lock prone but simple. Because of clock hierarchy spanning through controllers, the new locking became quite complicated. I don't like it but... Details ======= In Exynos-based boards case the deadlock occurs between clock's prepare_lock and regmap-i2c's lock: CPU #0: CPU #1: lock(regmap) s2mps11-clk: clk_prepare_lock() i2c-exynos: clk_prepare_lock() - wait lock(regmap) - wait The clk_prepare_lock() on both CPUs come from different clock drivers and components: 1. I2C clock is part of SoC block and is required by i2c-s3c2410/i2c-exynos5 driver, 2. S2MPS11 clock is separate device, however driver uses I2C regmap. The deadlock was reported by lockdep (always) and was happening in 20% of boots of Odroid XU3 with multi_v7 defconfig. Workaround for deadlock was implemented by removing prepare/unprepare calls from I2C transfers. However these are just workarounds... which after applying this patch can be reverted. Additionally Marek Szyprowski's work on domains/clocks/pinctrl exposed the deadlock again in different configuration. Comments as usual are welcomed. Best regards, Krzysztof Krzysztof Kozlowski (17): clk: bcm2835: Rename clk_register to avoid name conflict clk: Add clock controller to fine-grain the prepare lock clk: s2mps11: Switch to new clock controller API clk: samsung: Allocate a clock controller in context clk: fixed-rate: Switch to new clock controller API clk: gate: Switch to new clock controller API clk: mux: Switch to new clock controller API clk: fixed-factor: Switch to new clock controller API clk: divider: Switch to new clock controller API clk: composite: Switch to new clock controller API clk: gpio: Switch to new clock controller API ASoC: samsung: Switch to new clock controller API clk: samsung: audss: samsung: Switch to new clock controller API clk: samsung: clkout: samsung: Switch to new clock controller API clk: Use per-controller locking Revert "i2c: exynos5: Fix possible ABBA deadlock by keeping I2C clock prepared" Revert "i2c: s3c2410: fix ABBA deadlock by keeping clock prepared" drivers/clk/bcm/clk-bcm2835.c | 8 +- drivers/clk/clk-composite.c | 8 +- drivers/clk/clk-divider.c | 10 +- drivers/clk/clk-fixed-factor.c | 11 +- drivers/clk/clk-fixed-rate.c | 28 +- drivers/clk/clk-fractional-divider.c | 5 +- drivers/clk/clk-gate.c | 8 +- drivers/clk/clk-gpio.c | 29 +- drivers/clk/clk-mux.c | 32 ++- drivers/clk/clk-s2mps11.c | 10 +- drivers/clk/clk.c | 456 +++++++++++++++++++++++++++----- drivers/clk/samsung/clk-exynos-audss.c | 30 ++- drivers/clk/samsung/clk-exynos-clkout.c | 11 +- drivers/clk/samsung/clk.c | 25 +- drivers/clk/samsung/clk.h | 1 + drivers/i2c/busses/i2c-exynos5.c | 24 +- drivers/i2c/busses/i2c-s3c2410.c | 23 +- include/linux/clk-provider.h | 88 ++++-- include/linux/clk.h | 1 + sound/soc/samsung/i2s.c | 13 +- 20 files changed, 612 insertions(+), 209 deletions(-) -- 1.9.1

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[alsa-devel] [PATCH 00/10] ASoC: Intel: Skylake: Add support for D0i3
by Vinod Koul 09 Nov '16

09 Nov '16

This series adds support for low power D0i3 mode which lets audio controller perform stream and non streaming usage scenarios while being in lower power D0i3 state. In this series, first we modify the sst_ipc_tx_message() API to check if the DSP is in lower powered D0i3 mode or not and if so bring it to active before sending any messages. This is followed by a new API sst_ipc_tx_message_nopm() to send message without checking the new PM method. Then we start adding D0i3 refcounting which decides the entry to D0i3 mode based on capability. This is derived from topology data. The new IPC to enter D0i3 is also required. Finally we remove unwarranted pci device enabling calls on suspend. Jayachandran B (3): ASoC: Intel: Skylake: Add D0iX callbacks ASoC: Intel: Skylake: Flush pending D0i3 request on suspend ASoC: Intel: Skylake: remove pci device enabling calls on suspend Pardha Saradhi K (2): ASoC: Intel: Skylake: Add D0iX IPCs ASoC: Intel: Skylake: Add support for programming D0i3C Vinod Koul (5): ASoC: Intel: Add check_dsp_lp_on callback on IPC ASoC: Intel: Add sst_ipc_tx_message_nopm ASoC: Intel: Skylake: Add support for LPMode ASoC: Intel: Skylake: Add support for specifying D0i3 configuration ASoC: Intel: Skylake: Add D0i3 mode ref counting include/uapi/sound/snd_sst_tokens.h | 8 +- sound/soc/intel/common/sst-ipc.c | 27 +++++- sound/soc/intel/common/sst-ipc.h | 4 + sound/soc/intel/skylake/bxt-sst.c | 145 +++++++++++++++++++++++++++++++++ sound/soc/intel/skylake/skl-messages.c | 30 ++++++- sound/soc/intel/skylake/skl-pcm.c | 10 +++ sound/soc/intel/skylake/skl-sst-dsp.h | 12 +++ sound/soc/intel/skylake/skl-sst-ipc.c | 69 +++++++++++++++- sound/soc/intel/skylake/skl-sst-ipc.h | 36 +++++++- sound/soc/intel/skylake/skl-topology.c | 47 +++++++++++ sound/soc/intel/skylake/skl-topology.h | 11 +++ sound/soc/intel/skylake/skl.c | 59 +++++++++++++- sound/soc/intel/skylake/skl.h | 6 ++ 13 files changed, 456 insertions(+), 8 deletions(-) -- 1.9.1

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[alsa-devel] [PATCH v4 0/3] ASoC/ARM: tegra: apalis t30/tk1/colibri t30: sgtl5000 audio
by Marcel Ziswiler 07 Nov '16

07 Nov '16

This series adds/integrates Freescale SGTL5000 analogue audio codec support. Changes in v4: - simple-audio-card does still not allow for more advanced use cases like Tegra SoCs - further platform drivers have been accepted since my last attempt (e.g. rt5677 one) - relevance for one further board the new Toradex Apalis TK1 - drop unused sound/jack.h include - in tegra_sgtl5000_driver_remove() pass return value of snd_soc_unregister_card() on to caller - no longer set owner property in platform_driver to THIS_MODULE - re-based/resend Changes in v3: - revert to not using simple-audio-card being incompatible with tegra - rebased to for-next Changes in v2: - using simple-audio-card as suggested by Fabio Marcel Ziswiler (3): ASoC: tegra: add tegra sgtl5000 machine driver ARM: tegra: apalis/colibri t30: integrate audio ARM: tegra: enable sgtl5000 audio .../bindings/sound/nvidia,tegra-audio-sgtl5000.txt | 42 ++++ arch/arm/boot/dts/tegra30-apalis.dtsi | 49 +++++ arch/arm/boot/dts/tegra30-colibri.dtsi | 49 +++++ arch/arm/configs/tegra_defconfig | 1 + sound/soc/tegra/Kconfig | 11 ++ sound/soc/tegra/Makefile | 2 + sound/soc/tegra/tegra_sgtl5000.c | 212 +++++++++++++++++++++ 7 files changed, 366 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra-audio-sgtl5000.txt create mode 100644 sound/soc/tegra/tegra_sgtl5000.c -- 2.5.5

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Re: [alsa-devel] Intel Cherry Trail platform sound issue
by Zhihe Tu 05 Nov '16

05 Nov '16

Can anyone give some suggestions? Thanks Tu 发件人: 涂治河 [mailto:tuzhihe@pcasl.com] 发送时间: 2015年11月5日 19:37 收件人: 'alsa-devel(a)alsa-project.org' 主题: Intel Cherry Trail platform sound issue Hi All, I get cherry trail tablet, and install ubuntu14.04.3, Kernel verison is 3.19. The table using rt5672 codec with no sound output in ubuntu, but it works in win8. I get intel sst firmware fw_sst_22a8.bin from https://git.kernel.org/cgit/linux/kernel/git/firmware/linux-firmware.git/tr… The driver load the firmware successfully and I change some alsa mixer items: amixer -c0 sset 'codec_out0 mix 0 pcm0_in' on amixer -c0 sset 'media0_out mix 0 media1_in' on amixer -c0 sset 'media1_in Gain 0' 80% amixer -c0 sset 'media1_in Gain 0 Ramp Delay' 50 amixer -c0 sset 'media1_in Gain 0' off amixer -c0 sset 'pcm0_in Gain 0' 80% amixer -c0 sset 'pcm0_in Gain 0 Ramp Delay' 50 amixer -c0 sset 'pcm0_in Gain 0' off amixer -c0 sset 'codec_out0 Gain 0' 80% amixer -c0 sset 'codec_out0 Gain 0 Ramp Delay' 50 amixer -c0 sset 'codec_out0 Gain 0' off But there are some errors with dmesg: [ 16.359646] Audio Port: ASoC: no backend DAIs enabled for Audio Port … [ 16.382156] sst-mfld-platform sst-mfld-platform: Enter: enable=0 port_name=ssp2-port [ 16.383984] sst-mfld-platform sst-mfld-platform: Enter: enable=1 port_name=ssp2-port [ 16.541809] intel_sst_acpi 808622A8:00: Alloc for str 1 pipe 0x90 [ 17.545512] intel_sst_acpi 808622A8:00: Wait timed-out condition:0x0, msg_id:0x1 fw_state 0x2 [ 17.555207] intel_sst_acpi 808622A8:00: fw returned err -16 [ 18.605859] intel_sst_acpi 808622A8:00: Wait timed-out condition:0x0, msg_id:0x1 fw_state 0x3 [ 18.615908] intel_sst_acpi 808622A8:00: fw returned err -16 [ 19.621943] intel_sst_acpi 808622A8:00: Wait timed-out condition:0x0, msg_id:0x2 fw_state 0x3 [ 19.631499] intel_sst_acpi 808622A8:00: fw returned err -16 [ 20.637527] intel_sst_acpi 808622A8:00: Wait timed-out condition:0x0, msg_id:0x2 fw_state 0x3 [ 20.647083] intel_sst_acpi 808622A8:00: fw returned err -16 [ 20.682979] sst-mfld-platform sst-mfld-platform: Enter: enable=0 port_name=ssp2-port [ 21.681280] intel_sst_acpi 808622A8:00: Wait timed-out condition:0x0, msg_id:0x2 fw_state 0x3 [ 21.690836] intel_sst_acpi 808622A8:00: fw returned err -16 [ 21.697087] intel_sst_acpi 808622A8:00: free stream returned err -19 [ 22.757082] intel_sst_acpi 808622A8:00: Wait timed-out condition:0x0, msg_id:0x1 fw_state 0x3 [ 22.766640] intel_sst_acpi 808622A8:00: fw returned err -16 [ 23.772867] intel_sst_acpi 808622A8:00: Wait timed-out condition:0x0, msg_id:0x1 fw_state 0x3 [ 23.782432] intel_sst_acpi 808622A8:00: fw returned err -16 [ 23.857938] sst-mfld-platform sst-mfld-platform: Enter: enable=1 port_name=ssp2-port [ 23.989254] intel_sst_acpi 808622A8:00: Alloc for str 1 pipe 0x90 [ 23.989491] intel_sst_acpi 808622A8:00: FW sent error response 0x40034 [ 23.996864] intel_sst_acpi 808622A8:00: FW alloc failed ret -4 [ 24.003399] intel_sst_acpi 808622A8:00: sst_get_stream returned err -5 [ 24.010737] sst-mfld-platform sst-mfld-platform: ASoC: DAI prepare error: -5 [ 24.018652] Audio Port: ASoC: prepare FE Audio Port failed [ 25.024622] intel_sst_acpi 808622A8:00: Wait timed-out condition:0x0, msg_id:0x1 fw_state 0x3 [ 25.034180] intel_sst_acpi 808622A8:00: fw returned err -16 [ 26.040399] intel_sst_acpi 808622A8:00: Wait timed-out condition:0x0, msg_id:0x1 fw_state 0x3 [ 26.049955] intel_sst_acpi 808622A8:00: fw returned err -16 [ 27.056333] intel_sst_acpi 808622A8:00: Wait timed-out condition:0x0, msg_id:0x2 fw_state 0x3 [ 27.065890] intel_sst_acpi 808622A8:00: fw returned err -16 [ 28.072032] intel_sst_acpi 808622A8:00: Wait timed-out condition:0x0, msg_id:0x2 fw_state 0x3 [ 28.081591] intel_sst_acpi 808622A8:00: fw returned err -16 [ 28.117427] Audio Port: sst: error code = -22 [ 28.122401] snd_pcm_update_hw_ptr0: 120 callbacks suppressed [ 28.122409] cht-bsw-rt5672 cht-bsw-rt5672: BUG: , pos = -22, buffer size = 96000, period size = 48000 [ 28.132784] Audio Port: sst: error code = -22 [ 28.137749] cht-bsw-rt5672 cht-bsw-rt5672: BUG: , pos = -22, buffer size = 96000, period size = 48000 [ 28.372967] Audio Port: sst: error code = -22 [ 28.377946] cht-bsw-rt5672 cht-bsw-rt5672: BUG: , pos = -22, buffer size = 96000, period size = 48000 [ 28.388317] Audio Port: sst: error code = -22 [ 28.393289] cht-bsw-rt5672 cht-bsw-rt5672: BUG: , pos = -22, buffer size = 96000, period size = 48000 [ 28.403687] Audio Port: sst: error code = -22 lsmod info: snd_soc_sst_cht_bsw_rt5672 16384 0 snd_intel_sst_acpi 16384 1 snd_intel_sst_core 73728 1 snd_intel_sst_acpi snd_soc_sst_mfld_platform 77824 3 snd_intel_sst_core snd_soc_rt5670 106496 1 snd_soc_rl6231 16384 1 snd_soc_rt5670 snd_soc_core 196608 4 snd_soc_rt5670,snd_soc_sst_cht_bsw_rt5672,snd_soc_sst_mfld_platform snd_seq_midi 16384 0 snd_seq_midi_event 16384 1 snd_seq_midi snd_compress 20480 1 snd_soc_core snd_pcm_dmaengine 16384 1 snd_soc_core snd_rawmidi 32768 1 snd_seq_midi snd_pcm 106496 5 snd_soc_rt5670,snd_soc_core,snd_soc_sst_cht_bsw_rt5672,snd_soc_sst_mfld_platform,snd_pcm_dmaengine snd_seq 65536 2 snd_seq_midi_event,snd_seq_midi snd_seq_device 16384 3 snd_seq,snd_rawmidi,snd_seq_midi snd_timer 32768 2 snd_pcm,snd_seq snd 86016 10 snd_soc_core,snd_timer,snd_pcm,snd_seq,snd_rawmidi,snd_soc_sst_mfld_platform,snd_seq_device,snd_compress soundcore 16384 1 snd I have no idea with the next step, so can anyone give a look? Thanks Tu __________ Information from ESET Smart Security, version of virus signature database 4468 (20090929) __________ The message was checked by ESET Smart Security. http://www.eset.com __________ Information from ESET Smart Security, version of virus signature database 4468 (20090929) __________ The message was checked by ESET Smart Security. http://www.eset.com

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[alsa-devel] [PATCH v3 0/8] ASoC: topology: Remaining kernel patches
by mengdong.lin＠linux.intel.com 04 Nov '16

04 Nov '16

From: Mengdong Lin <mengdong.lin(a)linux.intel.com> This series contains all remaining kernel patches of topology, including some ABI update to PCM (FrontEnds) and link (BackEnds) objects. User will be able to config existing BE DAI links, configure more for FE links. Code are verified and can cover reqeust of Intel pre-release platforms for next year, so ABI should be stable. Current kernel topology code does not really touch Codec-Codec links since there is no user requst atm. We can add support for CC links later by reusing code and data structures for BE links, and no need to revise ABI. History: v2: Add the reason for creating BE DAI & DAI links by topology to commit message. Drop support for configuring DPCM trigger ordering in topology. v3: Topology no longer creates BE DAI or BE DAI links, but only configure existing ones. The API to find a DAI link can also check the name and stream name in case a soc card doesn't use unique ID for DAI links. The user space code is also ready. Mengdong Lin (8): ASoC: topology: ABI - Add flags to PCM ASoC: topology: ABI - Add private data to PCM ASoC: topology: ABI - Define DAI physical PCM data formats ASoC: topology: ABI - Add HW configurations to BE DAI links ASoC: topology: ABI - Add flags and private data to BE DAI links ASoC: topology: ABI - Add name and stream name to BE DAI links ASoC: Define API to find a dai link ASoC: topology: Add support to configure existing BE DAI links include/sound/soc-dai.h | 15 +++-- include/sound/soc.h | 3 + include/uapi/sound/asoc.h | 67 +++++++++++++++++++ sound/soc/soc-core.c | 42 ++++++++++++ sound/soc/soc-topology.c | 166 ++++++++++++++++++++++++++++++++++++++++++++-- 5 files changed, 282 insertions(+), 11 deletions(-) -- 2.5.0

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[alsa-devel] [PATCH v1 1/2] sound: lpass-cpu: add module licence and description
by Srinivas Kandagatla 31 Oct '16

31 Oct '16

This patch adds module licence to lpass-cpu driver, without this patch lpass-cpu module would taint with below error, while insmod/modprobe. snd_soc_lpass_cpu: module license 'unspecified' taints kernel. Disabling lock debugging due to kernel taint snd_soc_lpass_cpu: Unknown symbol regmap_write (err 0) snd_soc_lpass_cpu: Unknown symbol devm_kmalloc (err 0) ... Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> --- sound/soc/qcom/lpass-cpu.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/qcom/lpass-cpu.c b/sound/soc/qcom/lpass-cpu.c index 3cde9fb..eff3f9a 100644 --- a/sound/soc/qcom/lpass-cpu.c +++ b/sound/soc/qcom/lpass-cpu.c @@ -586,3 +586,6 @@ int asoc_qcom_lpass_cpu_platform_remove(struct platform_device *pdev) return 0; } EXPORT_SYMBOL_GPL(asoc_qcom_lpass_cpu_platform_remove); + +MODULE_DESCRIPTION("QTi LPASS CPU Driver"); +MODULE_LICENSE("GPL v2"); -- 2.8.3

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[alsa-devel] [PATCH 0/13] ALSA: Deletion of some unnecessary checks
by SF Markus Elfring 26 Oct '16

26 Oct '16

From: Markus Elfring <elfring(a)users.sourceforge.net> Date: Sun, 4 Jan 2015 11:50:12 +0100 Further update suggestions were taken into account after several patches were applied from static source code analysis. Markus Elfring (3): seq: Deletion of unnecessary checks before the function call "snd_midi_event_free" oss: Deletion of unnecessary checks before the function call "vfree" emu10k1: Delete an unnecessary check before the function call "kfree" oxygen: Delete an unnecessary check before the function call "snd_pcm_suspend" emux: Delete an unnecessary check before the function call "snd_sf_free" ASoC: Intel: Delete an unnecessary check before the function call "sst_dma_free" ASoC: fsi: Deletion of unnecessary checks before the function call "clk_enable" ASoC: Intel: Delete an unnecessary check before the function call "release_firmware" i2sbus: Delete an unnecessary check before the function call "snd_pcm_suspend_all" sb: Delete an unnecessary check before the function call "snd_emux_free" Deletion of checks before the function call "iounmap" msnd: One function call less in snd_msnd_attach() after error detection msnd: Fix centralized exiting from snd_msnd_attach() sound/aoa/soundbus/i2sbus/core.c | 13 ++++++------- sound/arm/aaci.c | 4 ++-- sound/drivers/ml403-ac97cr.c | 3 +-- sound/isa/msnd/msnd_pinnacle.c | 31 ++++++++++++++++--------------- sound/parisc/harmony.c | 4 +--- sound/pci/ad1889.c | 5 +---- sound/pci/asihpi/hpioctl.c | 6 ++---- sound/pci/atiixp.c | 3 +-- sound/pci/atiixp_modem.c | 3 +-- sound/pci/aw2/aw2-alsa.c | 4 +--- sound/pci/bt87x.c | 3 +-- sound/pci/cs4281.c | 6 ++---- sound/pci/cs46xx/cs46xx_lib.c | 4 ++-- sound/pci/ctxfi/cthw20k1.c | 5 +---- sound/pci/ctxfi/cthw20k2.c | 5 +---- sound/pci/echoaudio/echoaudio.c | 6 +----- sound/pci/hda/hda_intel.c | 3 +-- sound/pci/lola/lola.c | 6 ++---- sound/pci/mixart/mixart.c | 7 +++---- sound/pci/nm256/nm256.c | 6 ++---- sound/pci/rme9652/hdsp.c | 4 +--- sound/pci/rme9652/hdspm.c | 4 +--- sound/pci/rme9652/rme9652.c | 3 +-- sound/pci/sis7019.c | 5 +---- sound/pci/ymfpci/ymfpci_main.c | 3 +-- sound/ppc/pmac.c | 15 +++++---------- 26 files changed, 58 insertions(+), 103 deletions(-) -- 2.2.1

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[alsa-devel] [PATCH] ASoC: tlv320aic31xx: add explicit support for tlv320dac31xx
by Nikita Yushchenko 25 Oct '16

25 Oct '16

tlv320dac31xx is a subset of tlv320aic31xx: - it does not have MIC inputs and ADC, thus capture is not supported, - it has analog inputs AIN1/AIN2 that can be mixed into output. Although tlv320dac31xx does work with tlv320aic31xx driver, this setup does register non-existent widgets and non-existent capture stream. Thus userspace lists non-existent objects in user interfaces, an can access these, causing operations with device registers that are declared as "reserved" in tlv320dac31xx datasheet. This patch fixes this situation by separating controls/widgets/routes into common, aic31xx-specific, and dac31xx-specific parts. Only parts that match actual hardware (as declared in "compatible" device tree property) are registered. Signed-off-by: Nikita Yushchenko <nikita.yoush(a)cogentembedded.com> --- sound/soc/codecs/tlv320aic31xx.c | 212 ++++++++++++++++++++++++++++----------- sound/soc/codecs/tlv320aic31xx.h | 2 + 2 files changed, 158 insertions(+), 56 deletions(-) diff --git a/sound/soc/codecs/tlv320aic31xx.c b/sound/soc/codecs/tlv320aic31xx.c index 3c5e1df..fb7d648 100644 --- a/sound/soc/codecs/tlv320aic31xx.c +++ b/sound/soc/codecs/tlv320aic31xx.c @@ -273,10 +273,20 @@ static const DECLARE_TLV_DB_SCALE(sp_vol_tlv, -6350, 50, 0); /* * controls to be exported to the user space */ -static const struct snd_kcontrol_new aic31xx_snd_controls[] = { +static const struct snd_kcontrol_new common31xx_snd_controls[] = { SOC_DOUBLE_R_S_TLV("DAC Playback Volume", AIC31XX_LDACVOL, AIC31XX_RDACVOL, 0, -127, 48, 7, 0, dac_vol_tlv), + SOC_DOUBLE_R("HP Driver Playback Switch", AIC31XX_HPLGAIN, + AIC31XX_HPRGAIN, 2, 1, 0), + SOC_DOUBLE_R_TLV("HP Driver Playback Volume", AIC31XX_HPLGAIN, + AIC31XX_HPRGAIN, 3, 0x09, 0, hp_drv_tlv), + + SOC_DOUBLE_R_TLV("HP Analog Playback Volume", AIC31XX_LANALOGHPL, + AIC31XX_RANALOGHPR, 0, 0x7F, 1, hp_vol_tlv), +}; + +static const struct snd_kcontrol_new aic31xx_snd_controls[] = { SOC_SINGLE_TLV("ADC Fine Capture Volume", AIC31XX_ADCFGA, 4, 4, 1, adc_fgain_tlv), @@ -286,14 +296,6 @@ static const struct snd_kcontrol_new aic31xx_snd_controls[] = { SOC_SINGLE_TLV("Mic PGA Capture Volume", AIC31XX_MICPGA, 0, 119, 0, mic_pga_tlv), - - SOC_DOUBLE_R("HP Driver Playback Switch", AIC31XX_HPLGAIN, - AIC31XX_HPRGAIN, 2, 1, 0), - SOC_DOUBLE_R_TLV("HP Driver Playback Volume", AIC31XX_HPLGAIN, - AIC31XX_HPRGAIN, 3, 0x09, 0, hp_drv_tlv), - - SOC_DOUBLE_R_TLV("HP Analog Playback Volume", AIC31XX_LANALOGHPL, - AIC31XX_RANALOGHPR, 0, 0x7F, 1, hp_vol_tlv), }; static const struct snd_kcontrol_new aic311x_snd_controls[] = { @@ -397,17 +399,28 @@ static int aic31xx_dapm_power_event(struct snd_soc_dapm_widget *w, return 0; } -static const struct snd_kcontrol_new left_output_switches[] = { +static const struct snd_kcontrol_new aic31xx_left_output_switches[] = { SOC_DAPM_SINGLE("From Left DAC", AIC31XX_DACMIXERROUTE, 6, 1, 0), SOC_DAPM_SINGLE("From MIC1LP", AIC31XX_DACMIXERROUTE, 5, 1, 0), SOC_DAPM_SINGLE("From MIC1RP", AIC31XX_DACMIXERROUTE, 4, 1, 0), }; -static const struct snd_kcontrol_new right_output_switches[] = { +static const struct snd_kcontrol_new aic31xx_right_output_switches[] = { SOC_DAPM_SINGLE("From Right DAC", AIC31XX_DACMIXERROUTE, 2, 1, 0), SOC_DAPM_SINGLE("From MIC1RP", AIC31XX_DACMIXERROUTE, 1, 1, 0), }; +static const struct snd_kcontrol_new dac31xx_left_output_switches[] = { + SOC_DAPM_SINGLE("From Left DAC", AIC31XX_DACMIXERROUTE, 6, 1, 0), + SOC_DAPM_SINGLE("From AIN1", AIC31XX_DACMIXERROUTE, 5, 1, 0), + SOC_DAPM_SINGLE("From AIN2", AIC31XX_DACMIXERROUTE, 4, 1, 0), +}; + +static const struct snd_kcontrol_new dac31xx_right_output_switches[] = { + SOC_DAPM_SINGLE("From Right DAC", AIC31XX_DACMIXERROUTE, 2, 1, 0), + SOC_DAPM_SINGLE("From AIN2", AIC31XX_DACMIXERROUTE, 1, 1, 0), +}; + static const struct snd_kcontrol_new p_term_mic1lp = SOC_DAPM_ENUM("MIC1LP P-Terminal", mic1lp_p_enum); @@ -457,7 +470,7 @@ static int mic_bias_event(struct snd_soc_dapm_widget *w, return 0; } -static const struct snd_soc_dapm_widget aic31xx_dapm_widgets[] = { +static const struct snd_soc_dapm_widget common31xx_dapm_widgets[] = { SND_SOC_DAPM_AIF_IN("DAC IN", "DAC Playback", 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_MUX("DAC Left Input", @@ -473,14 +486,7 @@ static const struct snd_soc_dapm_widget aic31xx_dapm_widgets[] = { AIC31XX_DACSETUP, 6, 0, aic31xx_dapm_power_event, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), - /* Output Mixers */ - SND_SOC_DAPM_MIXER("Output Left", SND_SOC_NOPM, 0, 0, - left_output_switches, - ARRAY_SIZE(left_output_switches)), - SND_SOC_DAPM_MIXER("Output Right", SND_SOC_NOPM, 0, 0, - right_output_switches, - ARRAY_SIZE(right_output_switches)), - + /* HP */ SND_SOC_DAPM_SWITCH("HP Left", SND_SOC_NOPM, 0, 0, &aic31xx_dapm_hpl_switch), SND_SOC_DAPM_SWITCH("HP Right", SND_SOC_NOPM, 0, 0, @@ -494,10 +500,34 @@ static const struct snd_soc_dapm_widget aic31xx_dapm_widgets[] = { NULL, 0, aic31xx_dapm_power_event, SND_SOC_DAPM_POST_PMD | SND_SOC_DAPM_POST_PMU), - /* ADC */ - SND_SOC_DAPM_ADC_E("ADC", "Capture", AIC31XX_ADCSETUP, 7, 0, - aic31xx_dapm_power_event, SND_SOC_DAPM_POST_PMU | - SND_SOC_DAPM_POST_PMD), + /* Mic Bias */ + SND_SOC_DAPM_SUPPLY("MICBIAS", SND_SOC_NOPM, 0, 0, mic_bias_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), + + /* Outputs */ + SND_SOC_DAPM_OUTPUT("HPL"), + SND_SOC_DAPM_OUTPUT("HPR"), +}; + +static const struct snd_soc_dapm_widget dac31xx_dapm_widgets[] = { + /* Inputs */ + SND_SOC_DAPM_INPUT("AIN1"), + SND_SOC_DAPM_INPUT("AIN2"), + + /* Output Mixers */ + SND_SOC_DAPM_MIXER("Output Left", SND_SOC_NOPM, 0, 0, + dac31xx_left_output_switches, + ARRAY_SIZE(dac31xx_left_output_switches)), + SND_SOC_DAPM_MIXER("Output Right", SND_SOC_NOPM, 0, 0, + dac31xx_right_output_switches, + ARRAY_SIZE(dac31xx_right_output_switches)), +}; + +static const struct snd_soc_dapm_widget aic31xx_dapm_widgets[] = { + /* Inputs */ + SND_SOC_DAPM_INPUT("MIC1LP"), + SND_SOC_DAPM_INPUT("MIC1RP"), + SND_SOC_DAPM_INPUT("MIC1LM"), /* Input Selection to MIC_PGA */ SND_SOC_DAPM_MUX("MIC1LP P-Terminal", SND_SOC_NOPM, 0, 0, @@ -507,24 +537,25 @@ static const struct snd_soc_dapm_widget aic31xx_dapm_widgets[] = { SND_SOC_DAPM_MUX("MIC1LM P-Terminal", SND_SOC_NOPM, 0, 0, &p_term_mic1lm), + /* ADC */ + SND_SOC_DAPM_ADC_E("ADC", "Capture", AIC31XX_ADCSETUP, 7, 0, + aic31xx_dapm_power_event, SND_SOC_DAPM_POST_PMU | + SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_MUX("MIC1LM M-Terminal", SND_SOC_NOPM, 0, 0, &m_term_mic1lm), + /* Enabling & Disabling MIC Gain Ctl */ SND_SOC_DAPM_PGA("MIC_GAIN_CTL", AIC31XX_MICPGA, 7, 1, NULL, 0), - /* Mic Bias */ - SND_SOC_DAPM_SUPPLY("MICBIAS", SND_SOC_NOPM, 0, 0, mic_bias_event, - SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), - - /* Outputs */ - SND_SOC_DAPM_OUTPUT("HPL"), - SND_SOC_DAPM_OUTPUT("HPR"), - - /* Inputs */ - SND_SOC_DAPM_INPUT("MIC1LP"), - SND_SOC_DAPM_INPUT("MIC1RP"), - SND_SOC_DAPM_INPUT("MIC1LM"), + /* Output Mixers */ + SND_SOC_DAPM_MIXER("Output Left", SND_SOC_NOPM, 0, 0, + aic31xx_left_output_switches, + ARRAY_SIZE(aic31xx_left_output_switches)), + SND_SOC_DAPM_MIXER("Output Right", SND_SOC_NOPM, 0, 0, + aic31xx_right_output_switches, + ARRAY_SIZE(aic31xx_right_output_switches)), }; static const struct snd_soc_dapm_widget aic311x_dapm_widgets[] = { @@ -554,7 +585,7 @@ static const struct snd_soc_dapm_widget aic310x_dapm_widgets[] = { }; static const struct snd_soc_dapm_route -aic31xx_audio_map[] = { +common31xx_audio_map[] = { /* DAC Input Routing */ {"DAC Left Input", "Left Data", "DAC IN"}, {"DAC Left Input", "Right Data", "DAC IN"}, @@ -565,6 +596,31 @@ aic31xx_audio_map[] = { {"DAC Left", NULL, "DAC Left Input"}, {"DAC Right", NULL, "DAC Right Input"}, + /* HPL path */ + {"HP Left", "Switch", "Output Left"}, + {"HPL Driver", NULL, "HP Left"}, + {"HPL", NULL, "HPL Driver"}, + + /* HPR path */ + {"HP Right", "Switch", "Output Right"}, + {"HPR Driver", NULL, "HP Right"}, + {"HPR", NULL, "HPR Driver"}, +}; + +static const struct snd_soc_dapm_route +dac31xx_audio_map[] = { + /* Left Output */ + {"Output Left", "From Left DAC", "DAC Left"}, + {"Output Left", "From AIN1", "AIN1"}, + {"Output Left", "From AIN2", "AIN2"}, + + /* Right Output */ + {"Output Right", "From Right DAC", "DAC Right"}, + {"Output Right", "From AIN2", "AIN2"}, +}; + +static const struct snd_soc_dapm_route +aic31xx_audio_map[] = { /* Mic input */ {"MIC1LP P-Terminal", "FFR 10 Ohm", "MIC1LP"}, {"MIC1LP P-Terminal", "FFR 20 Ohm", "MIC1LP"}, @@ -595,16 +651,6 @@ aic31xx_audio_map[] = { /* Right Output */ {"Output Right", "From Right DAC", "DAC Right"}, {"Output Right", "From MIC1RP", "MIC1RP"}, - - /* HPL path */ - {"HP Left", "Switch", "Output Left"}, - {"HPL Driver", NULL, "HP Left"}, - {"HPL", NULL, "HPL Driver"}, - - /* HPR path */ - {"HP Right", "Switch", "Output Right"}, - {"HPR Driver", NULL, "HP Right"}, - {"HPR", NULL, "HPR Driver"}, }; static const struct snd_soc_dapm_route @@ -633,6 +679,13 @@ static int aic31xx_add_controls(struct snd_soc_codec *codec) int ret = 0; struct aic31xx_priv *aic31xx = snd_soc_codec_get_drvdata(codec); + if (!(aic31xx->pdata.codec_type & DAC31XX_BIT)) + ret = snd_soc_add_codec_controls( + codec, aic31xx_snd_controls, + ARRAY_SIZE(aic31xx_snd_controls)); + if (ret) + return ret; + if (aic31xx->pdata.codec_type & AIC31XX_STEREO_CLASS_D_BIT) ret = snd_soc_add_codec_controls( codec, aic311x_snd_controls, @@ -651,6 +704,30 @@ static int aic31xx_add_widgets(struct snd_soc_codec *codec) struct aic31xx_priv *aic31xx = snd_soc_codec_get_drvdata(codec); int ret = 0; + if (aic31xx->pdata.codec_type & DAC31XX_BIT) { + ret = snd_soc_dapm_new_controls( + dapm, dac31xx_dapm_widgets, + ARRAY_SIZE(dac31xx_dapm_widgets)); + if (ret) + return ret; + + ret = snd_soc_dapm_add_routes(dapm, dac31xx_audio_map, + ARRAY_SIZE(dac31xx_audio_map)); + if (ret) + return ret; + } else { + ret = snd_soc_dapm_new_controls( + dapm, aic31xx_dapm_widgets, + ARRAY_SIZE(aic31xx_dapm_widgets)); + if (ret) + return ret; + + ret = snd_soc_dapm_add_routes(dapm, aic31xx_audio_map, + ARRAY_SIZE(aic31xx_audio_map)); + if (ret) + return ret; + } + if (aic31xx->pdata.codec_type & AIC31XX_STEREO_CLASS_D_BIT) { ret = snd_soc_dapm_new_controls( dapm, aic311x_dapm_widgets, @@ -1114,12 +1191,12 @@ static struct snd_soc_codec_driver soc_codec_driver_aic31xx = { .set_bias_level = aic31xx_set_bias_level, .suspend_bias_off = true, - .controls = aic31xx_snd_controls, - .num_controls = ARRAY_SIZE(aic31xx_snd_controls), - .dapm_widgets = aic31xx_dapm_widgets, - .num_dapm_widgets = ARRAY_SIZE(aic31xx_dapm_widgets), - .dapm_routes = aic31xx_audio_map, - .num_dapm_routes = ARRAY_SIZE(aic31xx_audio_map), + .controls = common31xx_snd_controls, + .num_controls = ARRAY_SIZE(common31xx_snd_controls), + .dapm_widgets = common31xx_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(common31xx_dapm_widgets), + .dapm_routes = common31xx_audio_map, + .num_dapm_routes = ARRAY_SIZE(common31xx_audio_map), }; static const struct snd_soc_dai_ops aic31xx_dai_ops = { @@ -1129,6 +1206,21 @@ static const struct snd_soc_dai_ops aic31xx_dai_ops = { .digital_mute = aic31xx_dac_mute, }; +static struct snd_soc_dai_driver dac31xx_dai_driver[] = { + { + .name = "tlv32dac31xx-hifi", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rates = AIC31XX_RATES, + .formats = AIC31XX_FORMATS, + }, + .ops = &aic31xx_dai_ops, + .symmetric_rates = 1, + } +}; + static struct snd_soc_dai_driver aic31xx_dai_driver[] = { { .name = "tlv320aic31xx-hifi", @@ -1259,9 +1351,16 @@ static int aic31xx_i2c_probe(struct i2c_client *i2c, if (ret) return ret; - return snd_soc_register_codec(&i2c->dev, &soc_codec_driver_aic31xx, - aic31xx_dai_driver, - ARRAY_SIZE(aic31xx_dai_driver)); + if (aic31xx->pdata.codec_type & DAC31XX_BIT) + return snd_soc_register_codec(&i2c->dev, + &soc_codec_driver_aic31xx, + dac31xx_dai_driver, + ARRAY_SIZE(dac31xx_dai_driver)); + else + return snd_soc_register_codec(&i2c->dev, + &soc_codec_driver_aic31xx, + aic31xx_dai_driver, + ARRAY_SIZE(aic31xx_dai_driver)); } static int aic31xx_i2c_remove(struct i2c_client *i2c) @@ -1277,6 +1376,7 @@ static const struct i2c_device_id aic31xx_i2c_id[] = { { "tlv320aic3110", AIC3110 }, { "tlv320aic3120", AIC3120 }, { "tlv320aic3111", AIC3111 }, + { "tlv320dac3100", DAC3100 }, { } }; MODULE_DEVICE_TABLE(i2c, aic31xx_i2c_id); diff --git a/sound/soc/codecs/tlv320aic31xx.h b/sound/soc/codecs/tlv320aic31xx.h index ac9b146..5acd5b6 100644 --- a/sound/soc/codecs/tlv320aic31xx.h +++ b/sound/soc/codecs/tlv320aic31xx.h @@ -24,12 +24,14 @@ #define AIC31XX_STEREO_CLASS_D_BIT 0x1 #define AIC31XX_MINIDSP_BIT 0x2 +#define DAC31XX_BIT 0x4 enum aic31xx_type { AIC3100 = 0, AIC3110 = AIC31XX_STEREO_CLASS_D_BIT, AIC3120 = AIC31XX_MINIDSP_BIT, AIC3111 = (AIC31XX_STEREO_CLASS_D_BIT | AIC31XX_MINIDSP_BIT), + DAC3100 = DAC31XX_BIT, }; struct aic31xx_pdata { -- 2.1.4

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[alsa-devel] [PATCH 0/8] ASoC: s3c24xx_uda134x card fixes and cleanups
by Sylwester Nawrocki 25 Oct '16

25 Oct '16

This series might have a bit of an archaeological aspect, nevertheless my goal was to have sound working on one of the older S3C24XX SoC based boards for testing any patches touching the older Samsung SoC IP block related code. Together with patches: ASoC: samsung: Fix clock handling in S3C24XX_UDA134X card ARM: S3C24XX: Add dma_mask assignments for DMA devices ARM: S3C24XX: Add missing DMA device for Mini2440 board this series fixes sound on Micro2440SDK/mini2440 board broken since the 3.12 kernel release. I'm not sure how it worked before but currently the uda134x codec always fails to probe due to uninitialized platform_data, which is supposed to be set by the ASoC machine driver. This series addresses the issue by moving all required codec platform_data to the board file and exposing optional GPIO bitbang L3 bus ops which are selected by the codec. ASoC dependency patches: ASoC: samsung: Fix clock handling in S3C24XX_UDA134X card The patch "ARM: S3C24XX: Specify audio codec platform_data for mini2440 board" is for arm-soc but it would be good to merge whole series through one tree to avoid compilation breaks due to unmet dependencies. Sylwester Nawrocki (8): ASoC: L3 bus: Add default gpio ops ARM: S3C24XX: Specify audio codec platform_data for mini2440 board ASoC: uda134x: Optionally initialize L3 ops to default GPIO ops ASoC: s3c24xx_uda134x: Remove unused power() callback ASoC: s3c24xx_uda134x: Drop initialization of codec's platform data ASoC: samsung: Convert s3c24xx_uda134x to use devm_snd_soc_register_card() ASoC: samsung: s3c24xx_uda134x: debug/error trace cleanup ASoC: s3c244_uda134x: Allocate private data dynamically arch/arm/mach-s3c24xx/mach-mini2440.c | 19 ++++ include/sound/l3.h | 15 ++- include/sound/s3c24xx_uda134x.h | 1 - sound/soc/codecs/l3.c | 71 ++++++++++-- sound/soc/codecs/uda134x.c | 7 ++ sound/soc/samsung/s3c24xx_uda134x.c | 205 ++++++++++------------------------ 6 files changed, 158 insertions(+), 160 deletions(-) -- 1.9.1

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