September 2015 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH V4 1/2] ACPI / EC: Fix broken 64bit big-endian users of 'global_lock'
by Viresh Kumar 28 Sep '15

28 Sep '15

global_lock is defined as an unsigned long and accessing only its lower 32 bits from sysfs is incorrect, as we need to consider other 32 bits for big endian 64 bit systems. There are no such platforms yet, but the code needs to be robust for such a case. Fix that by passing a local variable to debugfs_create_bool() and assigning its value to global_lock later. Signed-off-by: Viresh Kumar <viresh.kumar(a)linaro.org> --- V3->V4: - Create a local variable instead of changing type of global_lock (Rafael) - Drop the stable tag - BCC'd a lot of people (rather than cc'ing them) to make sure - the series reaches them - mailing lists do not block the patchset due to long cc list - and we don't spam the BCC'd people for every reply --- drivers/acpi/ec_sys.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) diff --git a/drivers/acpi/ec_sys.c b/drivers/acpi/ec_sys.c index b4c216bab22b..b44b91331a56 100644 --- a/drivers/acpi/ec_sys.c +++ b/drivers/acpi/ec_sys.c @@ -110,6 +110,7 @@ static int acpi_ec_add_debugfs(struct acpi_ec *ec, unsigned int ec_device_count) struct dentry *dev_dir; char name[64]; umode_t mode = 0400; + u32 val; if (ec_device_count == 0) { acpi_ec_debugfs_dir = debugfs_create_dir("ec", NULL); @@ -127,10 +128,11 @@ static int acpi_ec_add_debugfs(struct acpi_ec *ec, unsigned int ec_device_count) if (!debugfs_create_x32("gpe", 0444, dev_dir, (u32 *)&first_ec->gpe)) goto error; - if (!debugfs_create_bool("use_global_lock", 0444, dev_dir, - (u32 *)&first_ec->global_lock)) + if (!debugfs_create_bool("use_global_lock", 0444, dev_dir, &val)) goto error; + first_ec->global_lock = val; + if (write_support) mode = 0600; if (!debugfs_create_file("io", mode, dev_dir, ec, &acpi_ec_io_ops)) -- 2.4.0

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[alsa-devel] [patch] ALSA: usb-audio: harmless underflow in snd_audigy2nx_led_put()
by Dan Carpenter 28 Sep '15

28 Sep '15

We want to verify that "value" is either zero or one, so we test if it is greater than one. Unfortunately, this is a signed int so it could also be negative. I think this is harmless but it introduces a static checker warning. Let's make "value" unsigned. Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c index 337c317..4602a78 100644 --- a/sound/usb/mixer_quirks.c +++ b/sound/usb/mixer_quirks.c @@ -340,7 +340,7 @@ static int snd_audigy2nx_led_put(struct snd_kcontrol *kcontrol, struct usb_mixer_elem_list *list = snd_kcontrol_chip(kcontrol); struct usb_mixer_interface *mixer = list->mixer; int index = kcontrol->private_value & 0xff; - int value = ucontrol->value.integer.value[0]; + unsigned int value = ucontrol->value.integer.value[0]; int old_value = kcontrol->private_value >> 8; int err;

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[alsa-devel] [PATCH] ALSA: hda - Fix typos in snd_hdac_regmap_*() documents
by Takashi Iwai 28 Sep '15

28 Sep '15

Fixes the wrong reference names to regmap amp functions. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- include/sound/hda_regmap.h | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/include/sound/hda_regmap.h b/include/sound/hda_regmap.h index df705908480a..2767c55a641e 100644 --- a/include/sound/hda_regmap.h +++ b/include/sound/hda_regmap.h @@ -67,7 +67,7 @@ int snd_hdac_regmap_update_raw(struct hdac_device *codec, unsigned int reg, * @reg: verb to write * @val: value to write * - * For writing an amp value, use snd_hda_regmap_amp_update(). + * For writing an amp value, use snd_hdac_regmap_update_amp(). */ static inline int snd_hdac_regmap_write(struct hdac_device *codec, hda_nid_t nid, @@ -85,7 +85,7 @@ snd_hdac_regmap_write(struct hdac_device *codec, hda_nid_t nid, * @mask: bit mask to update * @val: value to update * - * For updating an amp value, use snd_hda_regmap_amp_update(). + * For updating an amp value, use snd_hdac_regmap_update_amp(). */ static inline int snd_hdac_regmap_update(struct hdac_device *codec, hda_nid_t nid, -- 2.5.3

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Re: [alsa-devel] [PATCH RFC v4 3/8] ASoC: hdmi-codec: Add hdmi-codec for external HDMI-encoders
by Arnaud Pouliquen 28 Sep '15

28 Sep '15

few questions/remarks BR, Arnaud > +static void hdmi_codec_abort(struct device *dev) > +{ > + struct hdmi_codec_priv *hcp = dev_get_drvdata(dev); > + > + dev_dbg(dev, "%s()\n", __func__); > + > + mutex_lock(&hcp->current_stream_lock); > + if (hcp->current_stream && hcp->current_stream->runtime && > + snd_pcm_running(hcp->current_stream)) { > + dev_info(dev, "HDMI audio playback aborted\n"); > + snd_pcm_stream_lock_irq(hcp->current_stream); > + snd_pcm_stop(hcp->current_stream, SNDRV_PCM_STATE_DISCONNECTED); > + snd_pcm_stream_unlock_irq(hcp->current_stream); > + } > + mutex_unlock(&hcp->current_stream_lock); > +} Does driver should stop the stream in case of unplug? This could generate unexpected behavior at application level. Perhaps should be better if this part is managed in DRM driver. if HDMI master, I2S bus should be maintained ON to consume the audio stream until application action. > + > +static int hdmi_codec_hw_params(struct snd_pcm_substream *substream, > + struct snd_pcm_hw_params *params, > + struct snd_soc_dai *dai) > +{ > + struct hdmi_codec_priv *hcp = snd_soc_dai_get_drvdata(dai); > + struct hdmi_codec_params hp = { > + .iec = { > + .status = { 0 }, > + .subcode = { 0 }, > + .pad = 0, > + .dig_subframe = { 0 }, > + } > + }; > + int ret; > + > + dev_dbg(dai->dev, "%s() width %d rate %d channels %d\n", __func__, > + params_width(params), params_rate(params), > + params_channels(params)); > + > + ret = snd_pcm_create_iec958_consumer_hw_params(params, hp.iec.status, > + sizeof(hp.iec.status)); Tell me if i am wrong, but in case of SPDIF format, IEC status is managed by cpu_dai not by the codec. To manage IEC61937 a control should be needed to update IEC status bits... > + if (ret < 0) { > + dev_err(dai->dev, "Creating IEC958 channel status failed %d\n", > + ret); > + return ret; > + } > + > + ret = hdmi_codec_new_stream(substream, dai); > + if (ret) > + return ret; > + > + hdmi_audio_infoframe_init(&hp.cea); > + hp.cea.channels = params_channels(params); > + hp.cea.coding_type = HDMI_AUDIO_CODING_TYPE_STREAM; > + hp.cea.sample_size = HDMI_AUDIO_SAMPLE_SIZE_STREAM; > + hp.cea.sample_frequency = HDMI_AUDIO_SAMPLE_FREQUENCY_STREAM; > + > + hp.sample_width = params_width(params); > + hp.sample_rate = params_rate(params); > + hp.channels = params_channels(params); > + > + return hcp->hcd.ops->hw_params(dai->dev->parent, > &hcp->daifmt[dai->id], > + &hp); > +} > +

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[alsa-devel] [PATCH 0/3] Add support for Wolfson Micro WM8998 codec
by Richard Fitzgerald 28 Sep '15

28 Sep '15

This set of patches is based on broonie/regulator.git/for-next and broonie/sound.git/for-next branches. Richard Fitzgerald (3): regulator: arizona: add support for WM8998 and WM1814 ASoC: arizona: add defines for single-input gain control ASoC: wm8998: Initial WM8998 codec driver drivers/regulator/arizona-ldo1.c | 2 + sound/soc/codecs/Kconfig | 6 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/arizona.c | 33 + sound/soc/codecs/arizona.h | 12 +- sound/soc/codecs/wm8998.c | 1429 ++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/wm8998.h | 23 + 7 files changed, 1506 insertions(+), 1 deletion(-) create mode 100644 sound/soc/codecs/wm8998.c create mode 100644 sound/soc/codecs/wm8998.h -- 1.9.1

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[alsa-devel] [PATCH v2 1/1] ASoC: dwc: fix dma stop transferring issue
by yitian 28 Sep '15

28 Sep '15

Designware I2S uses tx empty and rx available signals as the DMA handshaking signals. during music playing, if XRUN occurs, i2s_stop() function will be executed and both tx and rx irq are masked, when music contintes to be played, i2s_start() is executed but both tx and rx irq are not unmasked which cause I2S stop sending DMA handshaking signal to DMA controller, and it finally cause music playing will be stopped once XRUN occurs for the first time. Signed-off-by: Yitian Bu <yitian.bu(a)tangramtek.com> --- changes in V2: - add definition for i and irq --- sound/soc/dwc/designware_i2s.c | 15 ++++++++++++--- 1 file changed, 12 insertions(+), 3 deletions(-) diff --git a/sound/soc/dwc/designware_i2s.c b/sound/soc/dwc/designware_i2s.c index a3e97b4..76b2e19 100644 --- a/sound/soc/dwc/designware_i2s.c +++ b/sound/soc/dwc/designware_i2s.c @@ -141,13 +141,22 @@ static inline void i2s_clear_irqs(struct dw_i2s_dev *dev, u32 stream) static void i2s_start(struct dw_i2s_dev *dev, struct snd_pcm_substream *substream) { - + u32 i, irq; i2s_write_reg(dev->i2s_base, IER, 1); - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + for (i = 0; i < 4; i++) { + irq = i2s_read_reg(dev->i2s_base, IMR(i)); + i2s_write_reg(dev->i2s_base, IMR(i), irq & ~0x30); + } i2s_write_reg(dev->i2s_base, ITER, 1); - else + } else { + for (i = 0; i < 4; i++) { + irq = i2s_read_reg(dev->i2s_base, IMR(i)); + i2s_write_reg(dev->i2s_base, IMR(i), irq & ~0x03); + } i2s_write_reg(dev->i2s_base, IRER, 1); + } i2s_write_reg(dev->i2s_base, CER, 1); } -- 1.7.12.4

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[alsa-devel] [PATCH 1/1] ASoC: dwc: fix dma stop transferring issue
by yitian 28 Sep '15

28 Sep '15

Designware I2S uses tx empty and rx available signals as the DMA handshaking signals. during music playing, if XRUN occurs, i2s_stop() function will be executed and both tx and rx irq are masked, when music contintes to be played, i2s_start() is executed but both tx and rx irq are not unmasked which cause I2S stop sending DMA handshaking signal to DMA controller, and it finally cause music playing will be stopped once XRUN occurs for the first time. Signed-off-by: Yitian Bu <yitian.bu(a)tangramtek.com> --- sound/soc/dwc/designware_i2s.c | 13 +++++++++++-- 1 file changed, 11 insertions(+), 2 deletions(-) diff --git a/sound/soc/dwc/designware_i2s.c b/sound/soc/dwc/designware_i2s.c index a3e97b4..4d3e1d3 100644 --- a/sound/soc/dwc/designware_i2s.c +++ b/sound/soc/dwc/designware_i2s.c @@ -144,10 +144,19 @@ static void i2s_start(struct dw_i2s_dev *dev, i2s_write_reg(dev->i2s_base, IER, 1); - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + for (i = 0; i < 4; i++) { + irq = i2s_read_reg(dev->i2s_base, IMR(i)); + i2s_write_reg(dev->i2s_base, IMR(i), irq & ~0x30); + } i2s_write_reg(dev->i2s_base, ITER, 1); - else + } else { + for (i = 0; i < 4; i++) { + irq = i2s_read_reg(dev->i2s_base, IMR(i)); + i2s_write_reg(dev->i2s_base, IMR(i), irq & ~0x03); + } i2s_write_reg(dev->i2s_base, IRER, 1); + } i2s_write_reg(dev->i2s_base, CER, 1); } -- 1.7.12.4

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[alsa-devel] [PATCH v2 0/2] ASoC: atmel-classd: add driver for Atmel CLASSD
by Songjun Wu 28 Sep '15

28 Sep '15

The Audio Class D Amplifier driver includes two parts. 1) Driver code to implement the Audio Class D Amplifier function. 2) Device tree binding document, it describes how to add the Audio Class D Amplifier in device tree. Changes in v2: - Change the "Mono", "Swap" and "Deemphasis" controls to "Mono Switch", "Swap Switch" and "Deemphasis Switch". - Merge "EQ Bass", "EQ Medium" and "EQ Treble" controls into one "EQ" control. - Change the "single-ended" and "differential" to "Single ended", "Differential". - Merge separate left and right controls into one single stereo control. - Add a warning if the user trys to specify an invalid value in function "atmel_classd_codec_probe". - Use the "dev_get_regmap()" in function "atmel_classd_codec_get_remap". - Check for errors when invoking the "clk_prepare_enable". - Remove the "dev_info" in function "atmel_classd_probe". - Add some code to create a sound card in function "atmel_classd_probe". - Remove the DT node "Sound" and the related code. - Delete the device node "Sound". Songjun Wu (2): ASoC: atmel-classd: add the Audio Class D Amplifier ASoC: atmel-classd: DT binding for Class D audio amplifier driver .../devicetree/bindings/sound/atmel-classd.txt | 52 ++ sound/soc/atmel/Kconfig | 9 + sound/soc/atmel/Makefile | 2 + sound/soc/atmel/atmel-classd.c | 680 ++++++++++++++++++++ sound/soc/atmel/atmel-classd.h | 120 ++++ 5 files changed, 863 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/atmel-classd.txt create mode 100644 sound/soc/atmel/atmel-classd.c create mode 100644 sound/soc/atmel/atmel-classd.h -- 1.7.9.5

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[alsa-devel] [PATCH] mfd: arizona: Call the runtime PM function if the state is runtime resumed
by Inha Song 27 Sep '15

27 Sep '15

This patch call runtiem PM function manually if the state is runtime resumed before enter suspend. This case has occurred when wake-up by the resume after entering to suspend during the playback. $ aplay -> runtime_resume() -> suspend() -> resume() In this case, Can occurred an error when enter suspend. -- [ 82.559234] PM: suspend of devices complete after 57.252 msecs [ 82.567978] arizona spi1.0: Failed to read IRQ status: -108 [ 82.567989] arizona spi1.0: Failed to read main IRQ status: -108 [ 82.568027] arizona spi1.0: Failed to read IRQ status: -108 [ 82.568036] arizona spi1.0: Failed to read main IRQ status: -108 .... So, we should call runtime_suspend() manually when enter suspend if the state is runtime resume. Also should call runtime_resume() when wake-up by resume if the state was runtime resume before entering suspend. Signed-off-by: Inha Song <ideal.song(a)samsung.com> --- drivers/mfd/arizona-core.c | 8 ++++++++ 1 file changed, 8 insertions(+) diff --git a/drivers/mfd/arizona-core.c b/drivers/mfd/arizona-core.c index 908c69b..de55706 100644 --- a/drivers/mfd/arizona-core.c +++ b/drivers/mfd/arizona-core.c @@ -683,6 +683,10 @@ static int arizona_suspend(struct device *dev) struct arizona *arizona = dev_get_drvdata(dev); dev_dbg(arizona->dev, "Suspend, disabling IRQ\n"); + + if (!pm_runtime_status_suspended(dev)) + arizona_runtime_suspend(dev); + disable_irq(arizona->irq); return 0; @@ -713,6 +717,10 @@ static int arizona_resume(struct device *dev) struct arizona *arizona = dev_get_drvdata(dev); dev_dbg(arizona->dev, "Late resume, reenabling IRQ\n"); + + if (!pm_runtime_status_suspended(dev)) + arizona_runtime_resume(dev); + enable_irq(arizona->irq); return 0; -- 2.0.0.390.gcb682f8

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[alsa-devel] Baytrail based HDMI Stick Orbsmart AW-01: No sound
by Michael Leun 26 Sep '15

26 Sep '15

Hi, I've an baytrail / Z3735F based HDMI Stick "Orbsmart AW-01", but sound does not work. I'm a little lost finding out what is missing / required to make it work. I compiled an 4.3-rc2 vanilla kernel with hopefully each and everything turned on relating soc/sound. In dmesg I get: [...] [ 4.252522] sst-acpi 80860F28:00: No matching ASoC machine driver found [ 5.863177] intel_sst_acpi 80860F28:00: No matching machine driver found [...] It looks like there is rt5640 based sound recognized: [ml@orb ~]$ find /sys -name '*5640*' -print /sys/bus/i2c/drivers/rt5640 /sys/bus/acpi/devices/10EC5640:00 /sys/bus/acpi/devices/10EC5640:01 /sys/devices/LNXSYSTM:00/LNXSYBUS:00/80860F41:01/10EC5640:00 /sys/devices/LNXSYSTM:00/LNXSYBUS:00/80860F41:01/10EC5640:01 find: "/sys/kernel/debug": Keine Berechtigung /sys/module/i2c_core/holders/snd_soc_rt5640 /sys/module/snd_soc_rl6231/holders/snd_soc_rt5640 /sys/module/snd_soc_rt5640 /sys/module/snd_soc_rt5640/drivers/i2c:rt5640 /sys/module/snd_soc_core/holders/snd_soc_rt5640 /sys/module/regmap_i2c/holders/snd_soc_rt5640 /sys/module/snd_pcm/holders/snd_soc_rt5640 [ml@orb ~]$ This are the sound related modules loaded: [ml@orb ~]$ lsmod|grep snd snd_intel_sst_acpi 16384 0 snd_intel_sst_core 65536 1 snd_intel_sst_acpi snd_soc_rt5640 106496 0 snd_soc_sst_mfld_platform 73728 1 snd_intel_sst_core snd_soc_rl6231 16384 1 snd_soc_rt5640 snd_soc_core 155648 2 snd_soc_rt5640,snd_soc_sst_mfld_platform snd_compress 20480 1 snd_soc_core regmap_i2c 16384 1 snd_soc_rt5640 snd_pcm_dmaengine 16384 1 snd_soc_core ac97_bus 16384 1 snd_soc_core snd_pcm 86016 4 snd_soc_rt5640,snd_soc_core,snd_soc_sst_mfld_platform,snd_pcm_dmaengine snd_timer 28672 1 snd_pcm snd 65536 5 snd_soc_core,snd_timer,snd_pcm,snd_soc_sst_mfld_platform,snd_compress soundcore 16384 1 snd i2c_core 49152 7 drm,i915,snd_soc_rt5640,i2c_designware_platform,regmap_i2c,drm_kms_helper,i2c_algo_bit snd_soc_sst_acpi 16384 0 [ml@orb ~]$ -- Best Regards, Michael Leun

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