April 2014 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] snd-usb-audio issues??
by Richard Storer 14 Apr '14

14 Apr '14

Hi, this is my first time post on the ML so apologies if my message is not to the required standard. I am currently using a Raspberry Pi ARM computer it's running a customised version of Debian called Raspbian: Linux raspberrypi 3.10.36+ #665 PREEMPT Wed Apr 9 15:01:53 BST 2014 armv6l GNU/Linux For audio playback and capture I am using a Creative Sound Blaster Live 24bit external: ID 041e:3040 Creative Technology, Ltd SoundBlaster Live! 24-bit External SB0490 I am currently experiencing 2 problems with this device which I believe are related to the snd-usb-audio driver. My first issue is for some reason booting Linux with my device already attached to the USB port produces no audio output, however capture is functioning. I've tried mute, unmute via alsamixer but it seems like the amplifier circuit inside hasn't turned itself on properly. The cure for this (and 100% reproducible) would be to unplug / replug the USB cable or simply connect to USB after Linux has fully loaded. I don't understand why this should be the case surely it doesn't matter at what stage the device is connected?? My next issue I feel is a little bit more serious and from what I understand affects all versions of Linux even the x86 distros. I use my Sound Blaster to capture stereo audio via Line-In which I then stream in realtime via Darkice. While Line-In capture does function on this device the signal levels are extremely low. Unfortunately there is no Line-In mixer control for my device the only capture control I have is Mic and that doesn't control Line-In. I have search for many months for a solution to the lack of Line-In control but I have come to the conclusion that something is broken in the driver or it got missed out. Is anyone aware of these issues or could at least offer a solution to increase the Line-In mixer levels maybe communicate with the USB device directly without ALSA's assistance?? Richard S.

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[alsa-devel] [PATCH 1/1] driver/snd: es18xx driver should use udelay error
by Li, Zhen-Hua 14 Apr '14

14 Apr '14

From: "Li, Zhen-Hua" <zhen-hual(a)hp.com> udelay with more than 20000 may cause __bad_udelay. Use mdelay for instead. Signed-off-by: Li, Zhen-Hua <zhen-hual(a)hp.com> --- sound/isa/es18xx.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/isa/es18xx.c b/sound/isa/es18xx.c index 1c16830..8d3a3b6 100644 --- a/sound/isa/es18xx.c +++ b/sound/isa/es18xx.c @@ -520,7 +520,7 @@ static int snd_es18xx_playback1_trigger(struct snd_es18xx *chip, snd_es18xx_mixer_write(chip, 0x78, 0x93); #ifdef AVOID_POPS /* Avoid pops */ - udelay(100000); + mudelay(100); if (chip->caps & ES18XX_PCM2) /* Restore Audio 2 volume */ snd_es18xx_mixer_write(chip, 0x7C, chip->audio2_vol); @@ -537,7 +537,7 @@ static int snd_es18xx_playback1_trigger(struct snd_es18xx *chip, /* Stop DMA */ snd_es18xx_mixer_write(chip, 0x78, 0x00); #ifdef AVOID_POPS - udelay(25000); + mdelay(25); if (chip->caps & ES18XX_PCM2) /* Set Audio 2 volume to 0 */ snd_es18xx_mixer_write(chip, 0x7C, 0); @@ -596,7 +596,7 @@ static int snd_es18xx_capture_prepare(struct snd_pcm_substream *substream) snd_es18xx_write(chip, 0xA5, count >> 8); #ifdef AVOID_POPS - udelay(100000); + mdelay(100); #endif /* Set format */ @@ -691,7 +691,7 @@ static int snd_es18xx_playback2_trigger(struct snd_es18xx *chip, snd_es18xx_write(chip, 0xB8, 0x05); #ifdef AVOID_POPS /* Avoid pops */ - udelay(100000); + mdelay(100); /* Enable Audio 1 */ snd_es18xx_dsp_command(chip, 0xD1); #endif @@ -705,7 +705,7 @@ static int snd_es18xx_playback2_trigger(struct snd_es18xx *chip, snd_es18xx_write(chip, 0xB8, 0x00); #ifdef AVOID_POPS /* Avoid pops */ - udelay(25000); + mdelay(25); /* Disable Audio 1 */ snd_es18xx_dsp_command(chip, 0xD3); #endif -- 1.7.10.4

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[alsa-devel] [PATCH 1/1] driver/snd: es18xx driver should use udelay error
by Li, Zhen-Hua 14 Apr '14

14 Apr '14

From: "Li, Zhen-Hua" <zhen-hual(a)hp.com> udelay with more than 20000 may cause __bad_udelay. Use mdelay for instead. Signed-off-by: Li, Zhen-Hua <zhen-hual(a)hp.com> --- sound/isa/es18xx.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/isa/es18xx.c b/sound/isa/es18xx.c index 1c16830..8d3a3b6 100644 --- a/sound/isa/es18xx.c +++ b/sound/isa/es18xx.c @@ -520,7 +520,7 @@ static int snd_es18xx_playback1_trigger(struct snd_es18xx *chip, snd_es18xx_mixer_write(chip, 0x78, 0x93); #ifdef AVOID_POPS /* Avoid pops */ - udelay(100000); + mudelay(100); if (chip->caps & ES18XX_PCM2) /* Restore Audio 2 volume */ snd_es18xx_mixer_write(chip, 0x7C, chip->audio2_vol); @@ -537,7 +537,7 @@ static int snd_es18xx_playback1_trigger(struct snd_es18xx *chip, /* Stop DMA */ snd_es18xx_mixer_write(chip, 0x78, 0x00); #ifdef AVOID_POPS - udelay(25000); + mdelay(25); if (chip->caps & ES18XX_PCM2) /* Set Audio 2 volume to 0 */ snd_es18xx_mixer_write(chip, 0x7C, 0); @@ -596,7 +596,7 @@ static int snd_es18xx_capture_prepare(struct snd_pcm_substream *substream) snd_es18xx_write(chip, 0xA5, count >> 8); #ifdef AVOID_POPS - udelay(100000); + mdelay(100); #endif /* Set format */ @@ -691,7 +691,7 @@ static int snd_es18xx_playback2_trigger(struct snd_es18xx *chip, snd_es18xx_write(chip, 0xB8, 0x05); #ifdef AVOID_POPS /* Avoid pops */ - udelay(100000); + mdelay(100); /* Enable Audio 1 */ snd_es18xx_dsp_command(chip, 0xD1); #endif @@ -705,7 +705,7 @@ static int snd_es18xx_playback2_trigger(struct snd_es18xx *chip, snd_es18xx_write(chip, 0xB8, 0x00); #ifdef AVOID_POPS /* Avoid pops */ - udelay(25000); + mdelay(25); /* Disable Audio 1 */ snd_es18xx_dsp_command(chip, 0xD3); #endif -- 1.7.10.4

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[alsa-devel] Add support of alc288
by Kailang 14 Apr '14

14 Apr '14

Hi Takashi, We have new codec release for ALC288. Thanks. BR, Kailang

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[alsa-devel] [PATCH] ASoC: rsnd: fix clock prepare/unprepare
by Kuninori Morimoto 14 Apr '14

14 Apr '14

From: Ben Dooks <ben.dooks(a)codethink.co.uk> As with the previous commit, before a clock can be used it must be prepared for use. Change from clk_enable() and clk_disable() to the versions of the calls which also prepare and un-prepare the clocks. Will fix warnings from the clock code when this is used. Signed-off-by: Ben Dooks <ben.dooks(a)codethink.co.uk> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- This patch is for current linus/master tree sound/soc/sh/rcar/src.c | 4 ++-- sound/soc/sh/rcar/ssi.c | 4 ++-- 2 files changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/sh/rcar/src.c b/sound/soc/sh/rcar/src.c index 6232b7d..4d0720e 100644 --- a/sound/soc/sh/rcar/src.c +++ b/sound/soc/sh/rcar/src.c @@ -258,7 +258,7 @@ static int rsnd_src_init(struct rsnd_mod *mod, { struct rsnd_src *src = rsnd_mod_to_src(mod); - clk_enable(src->clk); + clk_prepare_enable(src->clk); return 0; } @@ -269,7 +269,7 @@ static int rsnd_src_quit(struct rsnd_mod *mod, { struct rsnd_src *src = rsnd_mod_to_src(mod); - clk_disable(src->clk); + clk_disable_unprepare(src->clk); return 0; } diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c index 4b7e206..1d8387c 100644 --- a/sound/soc/sh/rcar/ssi.c +++ b/sound/soc/sh/rcar/ssi.c @@ -171,7 +171,7 @@ static void rsnd_ssi_hw_start(struct rsnd_ssi *ssi, u32 cr; if (0 == ssi->usrcnt) { - clk_enable(ssi->clk); + clk_prepare_enable(ssi->clk); if (rsnd_dai_is_clk_master(rdai)) { if (rsnd_ssi_clk_from_parent(ssi)) @@ -230,7 +230,7 @@ static void rsnd_ssi_hw_stop(struct rsnd_ssi *ssi, rsnd_ssi_master_clk_stop(ssi); } - clk_disable(ssi->clk); + clk_disable_unprepare(ssi->clk); } dev_dbg(dev, "ssi%d hw stopped\n", rsnd_mod_id(&ssi->mod)); -- 1.7.9.5

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[alsa-devel] Behringer Uphoria UMC204 usb audio capture only pops and clicks
by Andrew Reyes 14 Apr '14

14 Apr '14

I have a usb Behringer Uphoria UMC204 and it seems to work fine at 44.1khz full duplex but anything higher than that I get consistent pops and clicks. I notice it's on the capure side of things as when I have it in playback only it's quiet. I have to say I'm testing this using JACK so this is how I noticed it on the capture only side of things. I'm not sure exactly whats needed to debug this issue. Any help would be appreciated. Thanks in advance.

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[alsa-devel] Invalid PCM pointer with CMI8738 and pcm.c buffer underuns
by andoru 13 Apr '14

13 Apr '14

Continued from here: http://article.gmane.org/gmane.linux.alsa.devel/121575 > http://www.alsa-project.org/main/index.php/XRUN_Debug > > # Enable basic debugging and dump stack, check hardware pointer on the > period update > # Usefull to just see, if PCM stream is stopped for a reason (usually wrong > audio process timing from scheduler) > # And to check the values from driver > > echo 11 > /proc/asound/card0/pcm0p/xrun_debug Okay, I not compiled my own kernel with CONFIG_SND_PCM_XRUN_DEBUG, CONFIG_SND_VERBOSE_PROCFS, CONFIG_SND_DEBUG all enabled in the kernel configuration, I have an xrun_debug file, but it doesn't seem to list anything, then I also tried with cat instead of echo, and just a 0 showed up: ___________________________________________________________________ $ echo 11 > /proc/asound/card0/pcm0p/xrun_debug bash: /proc/asound/card0/pcm0p/xrun_debug: Permission denied # echo 11 > /proc/asound/card0/pcm0p/xrun_debug bash: /proc/asound/card0/pcm0p/xrun_debug: Permission denied $ cat /proc/asound/card0/pcm0p/xrun_debug 0 ___________________________________________________________________

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[alsa-devel] Behringer Uphoria UMC204 usb audio capture only pops and clicks
by Andrew Reyes 12 Apr '14

12 Apr '14

I have a usb Behringer Uphoria UMC204 and it seems to work fine at 44.1khz full duplex but anything higher than that I get consistent pops and clicks. I notice it's on the capure side of things as when I have it in playback only it's quiet. I have to say I'm testing this using JACK so this is how I noticed it on the capture only side of things. I'm not sure exactly whats needed to debug this issue. Any help would be appreciated. Thanks in advance.

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[alsa-devel] [PATCH] ALSA: pcm: 'BUG:' message unnecessarily triggers kerneloops
by Tim Gardner 11 Apr '14

11 Apr '14

BugLink: http://bugs.launchpad.net/bugs/1305480 The kerneloops-daemon scans dmesg for common crash signatures, among which is 'BUG:'. The message emitted by the PCM library is really a warning, so the most expedient thing to do seems to be to change the string. Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.de> Cc: Mark Brown <broonie(a)linaro.org> Cc: Lars-Peter Clausen <lars(a)metafoo.de> Cc: JongHo Kim <furmuwon(a)gmail.com> Signed-off-by: Tim Gardner <tim.gardner(a)canonical.com> --- sound/core/pcm_lib.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index ce83def..2a0563d 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -345,7 +345,7 @@ static int snd_pcm_update_hw_ptr0(struct snd_pcm_substream *substream, snd_pcm_debug_name(substream, name, sizeof(name)); xrun_log_show(substream); pcm_err(substream->pcm, - "BUG: %s, pos = %ld, buffer size = %ld, period size = %ld\n", + "OVER RUN: %s, pos = %ld, buffer size = %ld, period size = %ld\n", name, pos, runtime->buffer_size, runtime->period_size); } -- 1.7.9.5

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[alsa-devel] [PATCH v6] ASoC: fsl_sai: Add clock controls for SAI
by Nicolin Chen 11 Apr '14

11 Apr '14

The SAI mainly has the following clocks: bus clock control and configure registers and to generate synchronous interrupts and DMA requests. mclk1, mclk2, mclk3 to generate the bit clock when the receiver or transmitter is configured for an internally generated bit clock. So this patch adds these clocks and their clock controls to the driver. [ To concern the old DTB cases, I've added a bit of extra code to make the driver compatible with them. And by marking clock NULL if failed to get, the clk_prepare() or clk_get_rate() would easily return 0 so no further path should be broken. -- by Nicolin ] Signed-off-by: Nicolin Chen <Guangyu.Chen(a)freescale.com> Acked-by: Xiubo Li <Li.Xiubo(a)freescale.com> --- Changelog v6: * Dropped the dtsi change since it's no more demanded with this patch. * Removed Shawn's ACK since we no long change dtsi in this case. v5: * Dropped mclk preparing and enabling since we are not using them currecntly. * Made the change compatible to the old DTB. * Added returned error value print. v4: * Merged into single patch. * Fixed bus clock ID on vf610. v3: * Use int type for ret instead of u32. * Added Acked-by and Tested-by from Xiubo Li. v2: * Appended two extra mclks to the driver since SAI actually has three. * Renamed clock name to 'bus' and 'mclk' according to the reference manual. .../devicetree/bindings/sound/fsl-sai.txt | 9 +++-- sound/soc/fsl/fsl_sai.c | 38 ++++++++++++++++++++-- sound/soc/fsl/fsl_sai.h | 4 +++ 3 files changed, 46 insertions(+), 5 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/fsl-sai.txt b/Documentation/devicetree/bindings/sound/fsl-sai.txt index 35c09fe..0f4e238 100644 --- a/Documentation/devicetree/bindings/sound/fsl-sai.txt +++ b/Documentation/devicetree/bindings/sound/fsl-sai.txt @@ -10,7 +10,8 @@ Required properties: - compatible: Compatible list, contains "fsl,vf610-sai" or "fsl,imx6sx-sai". - reg: Offset and length of the register set for the device. - clocks: Must contain an entry for each entry in clock-names. -- clock-names : Must include the "sai" entry. +- clock-names : Must include the "bus" for register access and "mclk1" "mclk2" + "mclk3" for bit clock and frame clock providing. - dmas : Generic dma devicetree binding as described in Documentation/devicetree/bindings/dma/dma.txt. - dma-names : Two dmas have to be defined, "tx" and "rx". @@ -30,8 +31,10 @@ sai2: sai@40031000 { reg = <0x40031000 0x1000>; pinctrl-names = "default"; pinctrl-0 = <&pinctrl_sai2_1>; - clocks = <&clks VF610_CLK_SAI2>; - clock-names = "sai"; + clocks = <&clks VF610_CLK_PLATFORM_BUS>, + <&clks VF610_CLK_SAI2>, + <&clks 0>, <&clks 0>; + clock-names = "bus", "mclk1", "mclk2", "mclk3"; dma-names = "tx", "rx"; dmas = <&edma0 0 VF610_EDMA_MUXID0_SAI2_TX>, <&edma0 0 VF610_EDMA_MUXID0_SAI2_RX>; diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index db9f75e..5fff2e1 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -401,7 +401,15 @@ static int fsl_sai_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *cpu_dai) { struct fsl_sai *sai = snd_soc_dai_get_drvdata(cpu_dai); + struct device *dev = &sai->pdev->dev; u32 reg; + int ret; + + ret = clk_prepare_enable(sai->bus_clk); + if (ret) { + dev_err(dev, "failed to enable bus clock: %d\n", ret); + return ret; + } if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) reg = FSL_SAI_TCR3; @@ -427,6 +435,8 @@ static void fsl_sai_shutdown(struct snd_pcm_substream *substream, regmap_update_bits(sai->regmap, reg, FSL_SAI_CR3_TRCE, ~FSL_SAI_CR3_TRCE); + + clk_disable_unprepare(sai->bus_clk); } static const struct snd_soc_dai_ops fsl_sai_pcm_dai_ops = { @@ -559,7 +569,8 @@ static int fsl_sai_probe(struct platform_device *pdev) struct fsl_sai *sai; struct resource *res; void __iomem *base; - int irq, ret; + char tmp[8]; + int irq, ret, i; sai = devm_kzalloc(&pdev->dev, sizeof(*sai), GFP_KERNEL); if (!sai) @@ -582,12 +593,35 @@ static int fsl_sai_probe(struct platform_device *pdev) return PTR_ERR(base); sai->regmap = devm_regmap_init_mmio_clk(&pdev->dev, - "sai", base, &fsl_sai_regmap_config); + "bus", base, &fsl_sai_regmap_config); + + /* Compatible with old DTB cases */ + if (IS_ERR(sai->regmap)) + sai->regmap = devm_regmap_init_mmio_clk(&pdev->dev, + "sai", base, &fsl_sai_regmap_config); if (IS_ERR(sai->regmap)) { dev_err(&pdev->dev, "regmap init failed\n"); return PTR_ERR(sai->regmap); } + /* No error out for old DTB cases but only mark the clock NULL */ + sai->bus_clk = devm_clk_get(&pdev->dev, "bus"); + if (IS_ERR(sai->bus_clk)) { + dev_err(&pdev->dev, "failed to get bus clock: %ld\n", + PTR_ERR(sai->bus_clk)); + sai->bus_clk = NULL; + } + + for (i = 0; i < FSL_SAI_MCLK_MAX; i++) { + sprintf(tmp, "mclk%d", i + 1); + sai->mclk_clk[i] = devm_clk_get(&pdev->dev, tmp); + if (IS_ERR(sai->mclk_clk[i])) { + dev_err(&pdev->dev, "failed to get mclk%d clock: %ld\n", + i + 1, PTR_ERR(sai->mclk_clk[i])); + sai->mclk_clk[i] = NULL; + } + } + irq = platform_get_irq(pdev, 0); if (irq < 0) { dev_err(&pdev->dev, "no irq for node %s\n", np->full_name); diff --git a/sound/soc/fsl/fsl_sai.h b/sound/soc/fsl/fsl_sai.h index 677670d..0e6c9f5 100644 --- a/sound/soc/fsl/fsl_sai.h +++ b/sound/soc/fsl/fsl_sai.h @@ -119,6 +119,8 @@ #define FSL_SAI_CLK_MAST2 2 #define FSL_SAI_CLK_MAST3 3 +#define FSL_SAI_MCLK_MAX 3 + /* SAI data transfer numbers per DMA request */ #define FSL_SAI_MAXBURST_TX 6 #define FSL_SAI_MAXBURST_RX 6 @@ -126,6 +128,8 @@ struct fsl_sai { struct platform_device *pdev; struct regmap *regmap; + struct clk *bus_clk; + struct clk *mclk_clk[FSL_SAI_MCLK_MAX]; bool big_endian_regs; bool big_endian_data; -- 1.8.4

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