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

[alsa-devel] [PATCH] ASoC: wm8994: Fix missing break in switch
by Gustavo A. R. Silva 06 Aug '18

06 Aug '18

Add missing break statement in order to prevent the code from falling through to the default case. Cc: stable(a)vger.kernel.org Addresses-Coverity-ID: 115050 ("Missing break in switch") Reported-by: Valdis Kletnieks <valdis.kletnieks(a)vt.edu> Signed-off-by: Gustavo A. R. Silva <gustavo(a)embeddedor.com> --- sound/soc/codecs/wm8994.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 62f8c5b..14f1b0c 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -2432,7 +2432,7 @@ static int wm8994_set_dai_sysclk(struct snd_soc_dai *dai, snd_soc_component_update_bits(component, WM8994_POWER_MANAGEMENT_2, WM8994_OPCLK_ENA, 0); } - /* fall through */ + break; default: return -EINVAL; -- 2.7.4

3 2

[alsa-devel] [PATCH 00/11] ASoC: codecs: Mark expected switch fall-throughs
by Gustavo A. R. Silva 06 Aug '18

06 Aug '18

Hi all, In preparation to enabling -Wimplicit-fallthrough, this patchset aims to add some annotations in order to mark switch cases where we are expecting to fall through. Thanks Gustavo A. R. Silva (11): ASoC: wm8961: Mark expected switch fall-through ASoC: rt5640: Mark expected switch fall-through ASoC: rt5677: Mark expected switch fall-through ASoC: wm8955: Mark expected switch fall-through ASoC: wm8960: Mark expected switch fall-through ASoC: wm8904: Mark expected switch fall-through ASoC: wm8996: Mark expected switch fall-through ASoC: wm8962: Mark expected switch fall-through ASoC: wm8995: Mark expected switch fall-through ASoC: wm9081: Mark expected switch fall-through ASoC: wm8994: Mark expected switch fall-through sound/soc/codecs/rt5640.c | 2 ++ sound/soc/codecs/rt5677.c | 1 + sound/soc/codecs/wm8904.c | 1 + sound/soc/codecs/wm8955.c | 1 + sound/soc/codecs/wm8960.c | 1 + sound/soc/codecs/wm8961.c | 1 + sound/soc/codecs/wm8962.c | 1 + sound/soc/codecs/wm8994.c | 1 + sound/soc/codecs/wm8995.c | 1 + sound/soc/codecs/wm8996.c | 1 + sound/soc/codecs/wm9081.c | 1 + 11 files changed, 12 insertions(+) -- 2.7.4

3 32

[alsa-devel] [PATCH 0/5] ASoC: qcom: few trival fixes
by Srinivas Kandagatla 06 Aug '18

06 Aug '18

This patchset have few trivial fixes, 3 of them to do with cleaning up the unused header files, 1 patch is to do with fixing warning on common.c and other is to add missing AIF_IN dapm for slim tx ports. Thanks, srini Srinivas Kandagatla (5): ASoC: qcom: make common.c as proper module ASoC: apq8096: remove unused header files ASoC: sdm845: remove unused header files ASoC: qcom: remove unused header files from common.h ASoC: qdsp6: q6afe-dai: add SLIM tx AIF_IN dapm sound/soc/qcom/Kconfig | 5 +++++ sound/soc/qcom/Makefile | 6 ++++-- sound/soc/qcom/apq8096.c | 2 -- sound/soc/qcom/common.h | 1 - sound/soc/qcom/qdsp6/q6afe-dai.c | 7 +++++++ sound/soc/qcom/sdm845.c | 2 -- 6 files changed, 16 insertions(+), 7 deletions(-) -- 2.16.2

2 10

[alsa-devel] [PATCH 00/12] ASoC: codecs: use true and false for boolean values
by Gustavo A. R. Silva 06 Aug '18

06 Aug '18

Hi all, This patchset aims to replace the use of integer values with a proper true or false in return statements for functions returning bool. Thanks Gustavo A. R. Silva (12): ASoC: max9850: use true and false for boolean values ASoC: rt5631: use true and false for boolean values ASoC: tda7419: use true and false for boolean values ASoC: wm8990: use true and false for boolean values ASoC: cs4270: use true and false for boolean values ASoC: wm8996: use true and false for boolean values ASoC: da7219: use true and false for boolean values ASoC: twl6040: use true and false for boolean values ASoC: da7213: use true and false for boolean values ASoC: wm5100-tables: use true and false for boolean values ASoC: da9055: use true and false for boolean values ASoC: wm8903: use true and false for boolean values sound/soc/codecs/cs4270.c | 2 +- sound/soc/codecs/da7213.c | 4 ++-- sound/soc/codecs/da7219.c | 4 ++-- sound/soc/codecs/da9055.c | 4 ++-- sound/soc/codecs/max9850.c | 4 ++-- sound/soc/codecs/rt5631.c | 8 ++++---- sound/soc/codecs/tda7419.c | 4 ++-- sound/soc/codecs/twl6040.c | 2 +- sound/soc/codecs/wm5100-tables.c | 12 ++++++------ sound/soc/codecs/wm8903.c | 4 ++-- sound/soc/codecs/wm8990.c | 4 ++-- sound/soc/codecs/wm8996.c | 8 ++++---- 12 files changed, 30 insertions(+), 30 deletions(-) -- 2.7.4

3 26

[alsa-devel] Applied "ASoC: tda7419: use true and false for boolean values" to the asoc tree
by Mark Brown 06 Aug '18

06 Aug '18

The patch ASoC: tda7419: use true and false for boolean values 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 10754bfc051256ab922b818de3ddd11391862a5f Mon Sep 17 00:00:00 2001 From: "Gustavo A. R. Silva" <gustavo(a)embeddedor.com> Date: Sat, 4 Aug 2018 16:50:27 -0500 Subject: [PATCH] ASoC: tda7419: use true and false for boolean values Return statements in functions returning bool should use true or false instead of an integer value. This code was detected with the help of Coccinelle. Signed-off-by: Gustavo A. R. Silva <gustavo(a)embeddedor.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/tda7419.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/tda7419.c b/sound/soc/codecs/tda7419.c index 225c210ac38f..7f3b79c5a563 100644 --- a/sound/soc/codecs/tda7419.c +++ b/sound/soc/codecs/tda7419.c @@ -142,9 +142,9 @@ struct tda7419_vol_control { static inline bool tda7419_vol_is_stereo(struct tda7419_vol_control *tvc) { if (tvc->reg == tvc->rreg) - return 0; + return false; - return 1; + return true; } static int tda7419_vol_info(struct snd_kcontrol *kcontrol, -- 2.18.0

1 0

[alsa-devel] Applied "ASoC: wm5100-tables: use true and false for boolean values" to the asoc tree
by Mark Brown 06 Aug '18

06 Aug '18

The patch ASoC: wm5100-tables: use true and false for boolean values 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 380ae4ec420304aa59adb36776eaf78aef37192c Mon Sep 17 00:00:00 2001 From: "Gustavo A. R. Silva" <gustavo(a)embeddedor.com> Date: Sat, 4 Aug 2018 16:54:54 -0500 Subject: [PATCH] ASoC: wm5100-tables: use true and false for boolean values Return statements in functions returning bool should use true or false instead of an integer value. This code was detected with the help of Coccinelle. Signed-off-by: Gustavo A. R. Silva <gustavo(a)embeddedor.com> Acked-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/wm5100-tables.c | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/wm5100-tables.c b/sound/soc/codecs/wm5100-tables.c index e239f4bf2460..9e987cf07450 100644 --- a/sound/soc/codecs/wm5100-tables.c +++ b/sound/soc/codecs/wm5100-tables.c @@ -30,7 +30,7 @@ bool wm5100_volatile_register(struct device *dev, unsigned int reg) case WM5100_OUTPUT_STATUS_2: case WM5100_INPUT_ENABLES_STATUS: case WM5100_MIC_DETECT_3: - return 1; + return true; default: if ((reg >= WM5100_DSP1_PM_0 && reg <= WM5100_DSP1_PM_1535) || (reg >= WM5100_DSP1_ZM_0 && reg <= WM5100_DSP1_ZM_2047) || @@ -41,9 +41,9 @@ bool wm5100_volatile_register(struct device *dev, unsigned int reg) (reg >= WM5100_DSP3_PM_0 && reg <= WM5100_DSP3_PM_1535) || (reg >= WM5100_DSP3_ZM_0 && reg <= WM5100_DSP3_ZM_2047) || (reg >= WM5100_DSP3_DM_0 && reg <= WM5100_DSP3_DM_511)) - return 1; + return true; else - return 0; + return false; } } @@ -798,7 +798,7 @@ bool wm5100_readable_register(struct device *dev, unsigned int reg) case WM5100_DSP3_CONTROL_28: case WM5100_DSP3_CONTROL_29: case WM5100_DSP3_CONTROL_30: - return 1; + return true; default: if ((reg >= WM5100_DSP1_PM_0 && reg <= WM5100_DSP1_PM_1535) || (reg >= WM5100_DSP1_ZM_0 && reg <= WM5100_DSP1_ZM_2047) || @@ -809,9 +809,9 @@ bool wm5100_readable_register(struct device *dev, unsigned int reg) (reg >= WM5100_DSP3_PM_0 && reg <= WM5100_DSP3_PM_1535) || (reg >= WM5100_DSP3_ZM_0 && reg <= WM5100_DSP3_ZM_2047) || (reg >= WM5100_DSP3_DM_0 && reg <= WM5100_DSP3_DM_511)) - return 1; + return true; else - return 0; + return false; } } -- 2.18.0

1 0

[alsa-devel] [PATCH] ALSA: dice: fix wrong copy to rx parameters for Alesis iO26
by Takashi Sakamoto 06 Aug '18

06 Aug '18

A commit 28b208f600a3 ('ALSA: dice: add parameters of stream formats for models produced by Alesis') adds wrong copy to rx parameters instead of tx parameters for Alesis iO26. This commit fixes the bug for v4.18-rc8. Fixes: 28b208f600a3 ('ALSA: dice: add parameters of stream formats for models produced by Alesis') Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/dice/dice-alesis.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/firewire/dice/dice-alesis.c b/sound/firewire/dice/dice-alesis.c index b2efb1c71a98..218292bdace6 100644 --- a/sound/firewire/dice/dice-alesis.c +++ b/sound/firewire/dice/dice-alesis.c @@ -37,7 +37,7 @@ int snd_dice_detect_alesis_formats(struct snd_dice *dice) MAX_STREAMS * SND_DICE_RATE_MODE_COUNT * sizeof(unsigned int)); } else { - memcpy(dice->rx_pcm_chs, alesis_io26_tx_pcm_chs, + memcpy(dice->tx_pcm_chs, alesis_io26_tx_pcm_chs, MAX_STREAMS * SND_DICE_RATE_MODE_COUNT * sizeof(unsigned int)); } -- 2.17.1

2 1

[alsa-devel] [PATCH 1/3] ASoC: AMD: Make ACP->SYSMEM DMA non circular
by Akshu Agrawal 06 Aug '18

06 Aug '18

In capture case we don't want ACP to SYSMEM dma to be circular. This is because if an in place DSP filter is applied to captured output then circular DMA can overwrite the filter value with stale data. Signed-off-by: Akshu Agrawal <akshu.agrawal(a)amd.com> --- sound/soc/amd/acp-pcm-dma.c | 36 +++++++++++++++++++++++++++++++----- 1 file changed, 31 insertions(+), 5 deletions(-) diff --git a/sound/soc/amd/acp-pcm-dma.c b/sound/soc/amd/acp-pcm-dma.c index 94bcf69..816abd6 100644 --- a/sound/soc/amd/acp-pcm-dma.c +++ b/sound/soc/amd/acp-pcm-dma.c @@ -400,7 +400,7 @@ static void acp_dma_cap_channel_disable(void __iomem *acp_mmio, } /* Start a given DMA channel transfer */ -static void acp_dma_start(void __iomem *acp_mmio, u16 ch_num) +static void acp_dma_start(void __iomem *acp_mmio, u16 ch_num, bool is_circular) { u32 dma_ctrl; @@ -429,8 +429,11 @@ static void acp_dma_start(void __iomem *acp_mmio, u16 ch_num) break; } - /* circular for both DMA channel */ - dma_ctrl |= ACP_DMA_CNTL_0__Circular_DMA_En_MASK; + /* enable for ACP to SRAM DMA channel */ + if (is_circular == true) + dma_ctrl |= ACP_DMA_CNTL_0__Circular_DMA_En_MASK; + else + dma_ctrl &= ~ACP_DMA_CNTL_0__Circular_DMA_En_MASK; acp_reg_write(dma_ctrl, acp_mmio, mmACP_DMA_CNTL_0 + ch_num); } @@ -674,6 +677,7 @@ static int acp_deinit(void __iomem *acp_mmio) /* ACP DMA irq handler routine for playback, capture usecases */ static irqreturn_t dma_irq_handler(int irq, void *arg) { + u16 dscr_idx; u32 intr_flag, ext_intr_status; struct audio_drv_data *irq_data; void __iomem *acp_mmio; @@ -705,6 +709,15 @@ static irqreturn_t dma_irq_handler(int irq, void *arg) if ((intr_flag & BIT(I2S_TO_ACP_DMA_CH_NUM)) != 0) { valid_irq = true; + if (acp_reg_read(acp_mmio, mmACP_DMA_CUR_DSCR_14) == + CAPTURE_START_DMA_DESCR_CH15) + dscr_idx = CAPTURE_END_DMA_DESCR_CH14; + else + dscr_idx = CAPTURE_START_DMA_DESCR_CH14; + config_acp_dma_channel(acp_mmio, ACP_TO_SYSRAM_CH_NUM, dscr_idx, + 1, 0); + acp_dma_start(acp_mmio, ACP_TO_SYSRAM_CH_NUM, false); + snd_pcm_period_elapsed(irq_data->capture_i2ssp_stream); acp_reg_write((intr_flag & BIT(I2S_TO_ACP_DMA_CH_NUM)) << 16, acp_mmio, mmACP_EXTERNAL_INTR_STAT); @@ -712,6 +725,17 @@ static irqreturn_t dma_irq_handler(int irq, void *arg) if ((intr_flag & BIT(I2S_TO_ACP_DMA_BT_INSTANCE_CH_NUM)) != 0) { valid_irq = true; + if (acp_reg_read(acp_mmio, mmACP_DMA_CUR_DSCR_10) == + CAPTURE_START_DMA_DESCR_CH11) + dscr_idx = CAPTURE_END_DMA_DESCR_CH10; + else + dscr_idx = CAPTURE_START_DMA_DESCR_CH10; + config_acp_dma_channel(acp_mmio, + ACP_TO_SYSRAM_BT_INSTANCE_CH_NUM, + dscr_idx, 1, 0); + acp_dma_start(acp_mmio, ACP_TO_SYSRAM_BT_INSTANCE_CH_NUM, + false); + snd_pcm_period_elapsed(irq_data->capture_i2sbt_stream); acp_reg_write((intr_flag & BIT(I2S_TO_ACP_DMA_BT_INSTANCE_CH_NUM)) << 16, @@ -1053,9 +1077,11 @@ static int acp_dma_trigger(struct snd_pcm_substream *substream, int cmd) acp_dma_cap_channel_enable(rtd->acp_mmio, CAP_CHANNEL1); } + acp_dma_start(rtd->acp_mmio, rtd->ch1, true); + } else { + acp_dma_start(rtd->acp_mmio, rtd->ch1, true); + acp_dma_start(rtd->acp_mmio, rtd->ch2, true); } - acp_dma_start(rtd->acp_mmio, rtd->ch1); - acp_dma_start(rtd->acp_mmio, rtd->ch2); ret = 0; break; case SNDRV_PCM_TRIGGER_STOP: -- 1.9.1

3 4

[alsa-devel] A question about period
by Yu-hsuan Hsu 06 Aug '18

06 Aug '18

Hi all, I have a question about the period we set in hw_params. I found different boards may have different explanations about it. I have two guesses about its meaning. 1. The period_size is the size of each hardware's consumption. If we set period size to N, the pcm will consume N frames each time. 2. The period_size is the size to control when hardware call interrupt. If we set period size to N, the pcm consume frames in its step. When the number of frames it consumes more than N, it will call interrupt. We can use snd_pcm_avail function to check the real available frames in the device. If guess 1 is correct, the size of consumption should be fixed. Else, setting period_size is nothing to do with hardware's consumption. I've checked some boards and found that each board has different behavior (Most of them meet guess 2). I'm confuse which one is correct. Thanks!

2 8

Re: [alsa-devel] 答复: 答复: A bug about cache inconsistency report
by Takashi Iwai 06 Aug '18

06 Aug '18

On Thu, 02 Aug 2018 15:59:31 +0200, Hans Hu(SH-RD) wrote: > > > 1. noise relate test: > > When I annotate the return after set_pages_array_wc, then > > set_memory_wc works, noise disappear static void > > __mark_pages_wc(struct azx *chip, struct snd_dma_buffer *dmab, bool > > on) { ... > > set_pages_array_wb(sgbuf->page_table, sgbuf->pages); //return; } > > #endif > > > set_memory_uc() performs __pa() for the given address, and in this case, it's vmapped address. Something bogus in your test, I'm afraid. > > [Hans:] I know __pa() handle the vmapped address is wrong, that why I don't know how to fix the problem, I try this because I see kernel older than 3.7.6. [bogus? pretending to be real or genuine?] no no no, but the noise problem machine(8086:1c20) in another city, I will make sure it again. I meant about the reliability of the test result. Or whether we're heading to a wrong fish. In anyway, which kernel version are you testing? And exactly what is your problem? Which machine, which hardware setup (codec, etc) and which output, and how is the sound backend (PulseAudio or direct ALSA write)? > > And moreover, all Intel chipsets do work with snooping well, AFAIK. > [Hans:] yes, all Intel chipsets do work with snooping well here too. And the problem case here is "non-snoop"+"aplay -D plughw:*.*"+"44.1k 16bit"+"8086:1c20", I will see if I can find more machine do the test. Ah, so you explicitly test with non-snoop mode for this chipset? For what purpose...? > > 2. change *_SG relate test: > > Just followed your proposal > > -snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV_SG, > > +snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, > > >OK, you need one more oneliner: > > >diff --git a/sound/pci/hda/hda_controller.c b/sound/pci/hda/hda_controller.c index a12e594d4e3b..2eaed4956e7f 100644 > >--- a/sound/pci/hda/hda_controller.c > >+++ b/sound/pci/hda/hda_controller.c > >@@ -707,7 +707,7 @@ static const struct snd_pcm_ops azx_pcm_ops = { > >.pointer = azx_pcm_pointer, > >.get_time_info = azx_get_time_info, > >.mmap = azx_pcm_mmap, > >-.page = snd_pcm_sgbuf_ops_page, > >+// .page = snd_pcm_sgbuf_ops_page, > > }; > > > static void azx_pcm_free(struct snd_pcm *pcm) > [Hans :] I did this, and the Oops report as the same. My bad, there are other SG-buffer codes in sound/hda/*, too. So scratch these two oneliners. The only thing you'd need is the big-hammer change (again oneliner :) --- a/sound/core/Kconfig +++ b/sound/core/Kconfig @@ -181,7 +181,7 @@ config SND_VMASTER bool config SND_DMA_SGBUF - def_bool y + def_bool n depends on X86 source "sound/core/seq/Kconfig" > >And, again, please avoid top-posting. > [Hans :] sorry, I didn't understand at first. My mailbox is Outlook, am I right now? Yes, looks better now. thanks, Takashi

2 2