January 2017 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH 0/7] Move dell-led to drivers/platform/x86
by Michał Kępień 10 Jan '17

10 Jan '17

This patch series moves the dell-led driver from the LED subsystem to the x86 platform driver subsystem. I decided to also CC the sound subsystem contacts for the whole series as sound/pci/hda/dell_wmi_helper.c is also affected. The original motivation behind this effort was to move all code using the dell-smbios module to the x86 platform driver subsystem. While I was investigating the possibilites to do that, it quickly emerged that dell-led can and in fact should be moved to the x86 platform driver subsystem in its entirety. dell-led consists of two major parts: - the part exposing a microphone mute LED interface, introduced in db6d8cc ("dell-led: add mic mute led interface"); this interface is used by sound/pci/hda/dell_wmi_helper.c; while the original implementation used a WMI interface, it was changed to use dell-smbios in cf0d7ea ("dell-led: use dell_smbios_find_token() for finding mic DMI tokens") and 0c41a08 ("dell-led: use dell_smbios_send_request() for performing SMBIOS calls"), - the part handling an activity LED present in Dell Latitude 2100 netbooks, introduced in 72dcd8d ("leds: Add Dell Business Class Netbook LED driver"); it binds to a specific WMI GUID and then registers a LED device which is controlled using WMI (i.e. it is basically a WMI driver). Patches 1-4 clean up the microphone mute LED interface to minimize the amount of code moved around. Patch 5 moves the microphone mute LED interface to drivers/platform/x86/dell-laptop.c, effectively causing sound/pci/hda/dell_wmi_helper.c to depend on CONFIG_DELL_LAPTOP instead of CONFIG_LEDS_DELL_NETBOOKS. Patch 6 reverts dell-led to the state it was in after its initial commit 72dcd8d ("leds: Add Dell Business Class Netbook LED driver") by removing all remnants of the microphone mute LED handling code. Patch 7 moves all that is left of dell-led (i.e. the activity LED part, as originally implemented), to a new module which is placed in drivers/platform/x86/dell-wmi-led.c. This patch series is based on linux-leds/for-4.11 as the LED subsystem is affected by all patches except patch 3. If anyone reading this has access to a Dell device which has an activity LED and/or a microphone mute LED currently supported by dell-led, I would love to hear from you as I do not have the hardware needed to practically test this patch series. drivers/leds/Kconfig | 9 --- drivers/leds/Makefile | 1 - drivers/platform/x86/Kconfig | 8 +++ drivers/platform/x86/Makefile | 1 + drivers/platform/x86/dell-laptop.c | 28 ++++++++ .../dell-led.c => platform/x86/dell-wmi-led.c} | 75 +++------------------- include/linux/dell-led.h | 6 +- sound/pci/hda/dell_wmi_helper.c | 18 +++--- 8 files changed, 55 insertions(+), 91 deletions(-) rename drivers/{leds/dell-led.c => platform/x86/dell-wmi-led.c} (73%) -- 2.10.2

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[alsa-devel] [PATCH] ASoC: ak4642: Replace mdelay function to msleep
by Kuninori Morimoto 10 Jan '17

10 Jan '17

From: Harunobu Kurokawa <harunobu.kurokawa.dn(a)renesas.com> Replace mdelay to msleep to avoid busy loop on ak4642_lout_event(). Otherwise, sometimes playback doesn't work correctly when pulseaudio was used. Signed-off-by: Harunobu Kurokawa <harunobu.kurokawa.dn(a)renesas.com> --- sound/soc/codecs/ak4642.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c index 40500cd..0205ae1 100644 --- a/sound/soc/codecs/ak4642.c +++ b/sound/soc/codecs/ak4642.c @@ -186,7 +186,7 @@ static int ak4642_lout_event(struct snd_soc_dapm_widget *w, break; case SND_SOC_DAPM_POST_PMU: /* after widget power up */ /* Power save mode OFF */ - mdelay(popup_wait); + msleep(popup_wait); snd_soc_update_bits(codec, SG_SL2, LOPS, 0); break; case SND_SOC_DAPM_PRE_PMD: /* before widget power down */ -- 2.9.2

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[alsa-devel] [PATCH] ASoC: dpcm: Avoid putting stream state to STOP when FE stream is paused
by Patrick Lai 09 Jan '17

09 Jan '17

When multiple front-ends are using the same back-end, putting state of a front-end to STOP state upon receiving pause command will result in backend stream getting released by DPCM framework unintentionally. In order to avoid backend to be released when another active front-end stream is present, put the stream state to PAUSED state instead of STOP state. Signed-off-by: Patrick Lai <plai(a)codeaurora.org> --- sound/soc/soc-pcm.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index e7a1eaa..6aba140 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -2184,9 +2184,11 @@ static int dpcm_fe_dai_do_trigger(struct snd_pcm_substream *substream, int cmd) break; case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_SUSPEND: - case SNDRV_PCM_TRIGGER_PAUSE_PUSH: fe->dpcm[stream].state = SND_SOC_DPCM_STATE_STOP; break; + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + fe->dpcm[stream].state = SND_SOC_DPCM_STATE_PAUSED; + break; } out: -- The Qualcomm Innovation Center, Inc. is a member of the Code Aurora Forum,\na Linux Foundation Collaborative Project

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[alsa-devel] [PATCH 1/2] ASoC: rt5645: set sel_i2s_pre_div1 to 2
by Bard Liao 09 Jan '17

09 Jan '17

The i2s clock pre-divider 1 is used for both i2s1 and sysclk. The i2s1 is usually used for the main i2s and the pre-divider will be set in hw_params function. However, if i2s2 is used, the pre-divider is not set in the hw_params function and the default value of i2s clock pre-divider 1 is too high for sysclk and DMIC usage. Fix by overriding default divider value to 2. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=95681 Tested-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- sound/soc/codecs/rt5645.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 10c2a56..1ac96ef 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -3833,6 +3833,9 @@ static int rt5645_i2c_probe(struct i2c_client *i2c, } } + regmap_update_bits(rt5645->regmap, RT5645_ADDA_CLK1, + RT5645_I2S_PD1_MASK, RT5645_I2S_PD1_2); + if (rt5645->pdata.jd_invert) { regmap_update_bits(rt5645->regmap, RT5645_IRQ_CTRL2, RT5645_JD_1_1_MASK, RT5645_JD_1_1_INV); -- 1.8.1.1.439.g50a6b54

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[alsa-devel] [PATCH 1/2] ASoC: dwc: Add record capability in PIO mode
by Jose Abreu 09 Jan '17

09 Jan '17

Up until now PIO mode offered only playback support. With this patch we add support for record mode. The PCM was refactored so that we could reuse the existing infrastructure without many changes. We have support for 16 and 32 bits of sample size using only 2 channels. Tested in a x86_64 platform and in ARC AXS101 SDP platform. Signed-off-by: Jose Abreu <joabreu(a)synopsys.com> Cc: Carlos Palminha <palminha(a)synopsys.com> Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: alsa-devel(a)alsa-project.org Cc: linux-kernel(a)vger.kernel.org --- sound/soc/dwc/designware_i2s.c | 9 +++- sound/soc/dwc/designware_pcm.c | 97 +++++++++++++++++++++++++++++++++--------- sound/soc/dwc/local.h | 9 +++- 3 files changed, 92 insertions(+), 23 deletions(-) diff --git a/sound/soc/dwc/designware_i2s.c b/sound/soc/dwc/designware_i2s.c index bdf8398..9c46e41 100644 --- a/sound/soc/dwc/designware_i2s.c +++ b/sound/soc/dwc/designware_i2s.c @@ -121,9 +121,14 @@ static irqreturn_t i2s_irq_handler(int irq, void *dev_id) irq_valid = true; } - /* Data available. Record mode not supported in PIO mode */ - if (isr[i] & ISR_RXDA) + /* + * Data available. Retrieve samples from FIFO + * NOTE: Only two channels supported + */ + if ((isr[i] & ISR_RXDA) && (i == 0) && dev->use_pio) { + dw_pcm_pop_rx(dev); irq_valid = true; + } /* Error Handling: TX */ if (isr[i] & ISR_TXFO) { diff --git a/sound/soc/dwc/designware_pcm.c b/sound/soc/dwc/designware_pcm.c index 4a83a22..b063c86 100644 --- a/sound/soc/dwc/designware_pcm.c +++ b/sound/soc/dwc/designware_pcm.c @@ -41,10 +41,33 @@ return tx_ptr; \ } +#define dw_pcm_rx_fn(sample_bits) \ +static unsigned int dw_pcm_rx_##sample_bits(struct dw_i2s_dev *dev, \ + struct snd_pcm_runtime *runtime, unsigned int rx_ptr, \ + bool *period_elapsed) \ +{ \ + u##sample_bits (*p)[2] = (void *)runtime->dma_area; \ + unsigned int period_pos = rx_ptr % runtime->period_size; \ + int i; \ +\ + for (i = 0; i < dev->fifo_th; i++) { \ + p[rx_ptr][0] = ioread32(dev->i2s_base + LRBR_LTHR(0)); \ + p[rx_ptr][1] = ioread32(dev->i2s_base + RRBR_RTHR(0)); \ + period_pos++; \ + if (++rx_ptr >= runtime->buffer_size) \ + rx_ptr = 0; \ + } \ + *period_elapsed = period_pos >= runtime->period_size; \ + return rx_ptr; \ +} + dw_pcm_tx_fn(16); dw_pcm_tx_fn(32); +dw_pcm_rx_fn(16); +dw_pcm_rx_fn(32); #undef dw_pcm_tx_fn +#undef dw_pcm_rx_fn static const struct snd_pcm_hardware dw_pcm_hardware = { .info = SNDRV_PCM_INFO_INTERLEAVED | @@ -68,27 +91,51 @@ .fifo_size = 16, }; -void dw_pcm_push_tx(struct dw_i2s_dev *dev) +static void dw_pcm_transfer(struct dw_i2s_dev *dev, bool push) { - struct snd_pcm_substream *tx_substream; - bool tx_active, period_elapsed; + struct snd_pcm_substream *substream; + bool active, period_elapsed; rcu_read_lock(); - tx_substream = rcu_dereference(dev->tx_substream); - tx_active = tx_substream && snd_pcm_running(tx_substream); - if (tx_active) { - unsigned int tx_ptr = READ_ONCE(dev->tx_ptr); - unsigned int new_tx_ptr = dev->tx_fn(dev, tx_substream->runtime, - tx_ptr, &period_elapsed); - cmpxchg(&dev->tx_ptr, tx_ptr, new_tx_ptr); + if (push) + substream = rcu_dereference(dev->tx_substream); + else + substream = rcu_dereference(dev->rx_substream); + active = substream && snd_pcm_running(substream); + if (active) { + unsigned int ptr; + unsigned int new_ptr; + + if (push) { + ptr = READ_ONCE(dev->tx_ptr); + new_ptr = dev->tx_fn(dev, substream->runtime, ptr, + &period_elapsed); + cmpxchg(&dev->tx_ptr, ptr, new_ptr); + } else { + ptr = READ_ONCE(dev->rx_ptr); + new_ptr = dev->rx_fn(dev, substream->runtime, ptr, + &period_elapsed); + cmpxchg(&dev->rx_ptr, ptr, new_ptr); + } if (period_elapsed) - snd_pcm_period_elapsed(tx_substream); + snd_pcm_period_elapsed(substream); } rcu_read_unlock(); } + +void dw_pcm_push_tx(struct dw_i2s_dev *dev) +{ + dw_pcm_transfer(dev, true); +} EXPORT_SYMBOL_GPL(dw_pcm_push_tx); +void dw_pcm_pop_rx(struct dw_i2s_dev *dev) +{ + dw_pcm_transfer(dev, false); +} +EXPORT_SYMBOL_GPL(dw_pcm_pop_rx); + static int dw_pcm_open(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; @@ -126,20 +173,17 @@ static int dw_pcm_hw_params(struct snd_pcm_substream *substream, switch (params_format(hw_params)) { case SNDRV_PCM_FORMAT_S16_LE: dev->tx_fn = dw_pcm_tx_16; + dev->rx_fn = dw_pcm_rx_16; break; case SNDRV_PCM_FORMAT_S32_LE: dev->tx_fn = dw_pcm_tx_32; + dev->rx_fn = dw_pcm_rx_32; break; default: dev_err(dev->dev, "invalid format\n"); return -EINVAL; } - if (substream->stream != SNDRV_PCM_STREAM_PLAYBACK) { - dev_err(dev->dev, "only playback is available\n"); - return -EINVAL; - } - ret = snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(hw_params)); if (ret < 0) @@ -163,13 +207,21 @@ static int dw_pcm_trigger(struct snd_pcm_substream *substream, int cmd) case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - WRITE_ONCE(dev->tx_ptr, 0); - rcu_assign_pointer(dev->tx_substream, substream); + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + WRITE_ONCE(dev->tx_ptr, 0); + rcu_assign_pointer(dev->tx_substream, substream); + } else { + WRITE_ONCE(dev->rx_ptr, 0); + rcu_assign_pointer(dev->rx_substream, substream); + } break; case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_SUSPEND: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - rcu_assign_pointer(dev->tx_substream, NULL); + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + rcu_assign_pointer(dev->tx_substream, NULL); + else + rcu_assign_pointer(dev->rx_substream, NULL); break; default: ret = -EINVAL; @@ -183,7 +235,12 @@ static snd_pcm_uframes_t dw_pcm_pointer(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; struct dw_i2s_dev *dev = runtime->private_data; - snd_pcm_uframes_t pos = READ_ONCE(dev->tx_ptr); + snd_pcm_uframes_t pos; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + pos = READ_ONCE(dev->tx_ptr); + else + pos = READ_ONCE(dev->rx_ptr); return pos < runtime->buffer_size ? pos : 0; } diff --git a/sound/soc/dwc/local.h b/sound/soc/dwc/local.h index 68afd75..91dc70a 100644 --- a/sound/soc/dwc/local.h +++ b/sound/soc/dwc/local.h @@ -105,20 +105,27 @@ struct dw_i2s_dev { struct i2s_clk_config_data config; int (*i2s_clk_cfg)(struct i2s_clk_config_data *config); - /* data related to PIO transfers (TX) */ + /* data related to PIO transfers */ bool use_pio; struct snd_pcm_substream __rcu *tx_substream; + struct snd_pcm_substream __rcu *rx_substream; unsigned int (*tx_fn)(struct dw_i2s_dev *dev, struct snd_pcm_runtime *runtime, unsigned int tx_ptr, bool *period_elapsed); + unsigned int (*rx_fn)(struct dw_i2s_dev *dev, + struct snd_pcm_runtime *runtime, unsigned int rx_ptr, + bool *period_elapsed); unsigned int tx_ptr; + unsigned int rx_ptr; }; #if IS_ENABLED(CONFIG_SND_DESIGNWARE_PCM) void dw_pcm_push_tx(struct dw_i2s_dev *dev); +void dw_pcm_pop_rx(struct dw_i2s_dev *dev); int dw_pcm_register(struct platform_device *pdev); #else void dw_pcm_push_tx(struct dw_i2s_dev *dev) { } +void dw_pcm_pop_rx(struct dw_i2s_dev *dev) { } int dw_pcm_register(struct platform_device *pdev) { return -EINVAL; -- 1.9.1

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[alsa-devel] [PATCH v3 0/6] ASoC: sun4i-codec: Add support for A31 Codec
by Chen-Yu Tsai 09 Jan '17

09 Jan '17

Hi everyone, This is v3 of my Allwinner A31 audio codec support series. These are the remaining driver patches not yet merged, with 1 new fix, and the dts file patches. The A31's internal codec is similar (in terms of DMA, interface and control layouts) to the one found in the A10/A13/A20 SoCs. However it has more external inputs and outputs, the mixer controls are now stereo (left/right separated), and it also has some audio processing features (not supported yet). Changes since v2: - Added patch """ASoC: sun4i-codec: Add "Right Mixer" to "Line Out Mono Diff." route""" which fixes line out when all the sources have their left channel muted. - Dropped unused codec_pa_pin from A31 hummingbird dts patch. In the past we always provided a pinmux setting for GPIOs used, to avoid accidentally muxing a pin that was already used for GPIO, or vice versa. The pinctrl subsystem now supports exclusive use of a pin for GPIO, which the sunxi platform wants to migrate to. This drops the requirement of the explicit pinmux setting for GPIOs. - Added Maxime's Acked-by for the two existing patches from v2. After these are merged I plan to submit patches to support the audio codec found in Allwinner's A23 and H3 SoCs. Regards ChenYu Chen-Yu Tsai (6): ASoC: sun4i-codec: Add support for optional reset control to quirks ASoC: sun4i-codec: Add support for A31 ADC capture path ASoC: sun4i-codec: Add "Right Mixer" to "Line Out Mono Diff." route ARM: dts: sun6i: Add audio codec device node ARM: dts: sun6i: hummingbird: Enable internal audio codec ARM: dts: sun6i: sina31s: Enable internal audio codec arch/arm/boot/dts/sun6i-a31-hummingbird.dts | 13 ++++ arch/arm/boot/dts/sun6i-a31.dtsi | 13 ++++ arch/arm/boot/dts/sun6i-a31s-sina31s.dts | 8 +++ sound/soc/sunxi/sun4i-codec.c | 94 ++++++++++++++++++++++++++++- 4 files changed, 127 insertions(+), 1 deletion(-) -- 2.10.2

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[alsa-devel] [PATCH] ASoC: nau8825: Add Class G function options
by Ben Zhang 09 Jan '17

09 Jan '17

Add a DAPM_MUX for userspace to control the Class G function: "Switching": Use +-1.8V or +-0.9V supply, based on signal amplitude. "1.8V": Always use +-1.8V. The default is "Switching" which is the existing behavior without this patch, so no effect on current users. On some devices, the switching may cause a clicking noise at low volume. "1.8V" eliminates the noise at the expense of a few additional mW. Signed-off-by: Ben Zhang <benzh(a)chromium.org> --- sound/soc/codecs/nau8825.c | 24 +++++++++++++++++++++--- 1 file changed, 21 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/nau8825.c b/sound/soc/codecs/nau8825.c index efe3a44..ddd15a0 100644 --- a/sound/soc/codecs/nau8825.c +++ b/sound/soc/codecs/nau8825.c @@ -1062,6 +1062,10 @@ static const char * const nau8825_dac_src[] = { "DACL", "DACR", }; +static const char * const nau8825_class_g_src[] = { + "Switching", "1.8V" +}; + static SOC_ENUM_SINGLE_DECL( nau8825_dacl_enum, NAU8825_REG_DACL_CTRL, NAU8825_DACL_CH_SEL_SFT, nau8825_dac_src); @@ -1070,12 +1074,17 @@ static SOC_ENUM_SINGLE_DECL( nau8825_dacr_enum, NAU8825_REG_DACR_CTRL, NAU8825_DACR_CH_SEL_SFT, nau8825_dac_src); +static SOC_ENUM_SINGLE_DECL( + nau8825_class_g_enum, SND_SOC_NOPM, 0, nau8825_class_g_src); + static const struct snd_kcontrol_new nau8825_dacl_mux = SOC_DAPM_ENUM("DACL Source", nau8825_dacl_enum); static const struct snd_kcontrol_new nau8825_dacr_mux = SOC_DAPM_ENUM("DACR Source", nau8825_dacr_enum); +static const struct snd_kcontrol_new nau8825_class_g_mux = + SOC_DAPM_ENUM("Class G Source", nau8825_class_g_enum); static const struct snd_soc_dapm_widget nau8825_dapm_widgets[] = { SND_SOC_DAPM_AIF_OUT("AIFTX", "Capture", 0, NAU8825_REG_I2S_PCM_CTRL2, @@ -1154,9 +1163,16 @@ static const struct snd_soc_dapm_widget nau8825_dapm_widgets[] = { SND_SOC_DAPM_PGA_S("HP Boost Driver", 9, NAU8825_REG_BOOST, 9, 1, NULL, 0), - /* Class G operation control*/ - SND_SOC_DAPM_PGA_S("Class G", 10, + /* Class G operation control. "Class G" MUX options: + * "Switching": Use +-1.8V or +-0.9V supply, based on signal amplitude. + * "1.8V": Always use +-1.8V. + * The PGA_S widget is used to ensure the Class G control bit is set at + * the correct time w.r.t the other widgets. + */ + SND_SOC_DAPM_PGA_S("Class G Driver", 10, NAU8825_REG_CLASSG_CTRL, 0, 0, NULL, 0), + SND_SOC_DAPM_MUX("Class G", SND_SOC_NOPM, 0, 0, + &nau8825_class_g_mux), SND_SOC_DAPM_OUTPUT("HPOL"), SND_SOC_DAPM_OUTPUT("HPOR"), @@ -1197,7 +1213,9 @@ static const struct snd_soc_dapm_route nau8825_dapm_routes[] = { {"HPOR Pulldown", NULL, "Output DACR"}, {"HP Boost Driver", NULL, "HPOL Pulldown"}, {"HP Boost Driver", NULL, "HPOR Pulldown"}, - {"Class G", NULL, "HP Boost Driver"}, + {"Class G Driver", NULL, "HP Boost Driver"}, + {"Class G", "1.8V", "HP Boost Driver"}, + {"Class G", "Switching", "Class G Driver"}, {"HPOL", NULL, "Class G"}, {"HPOR", NULL, "Class G"}, }; -- 2.8.0.rc3.226.g39d4020

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[alsa-devel] Input sound doesn't work without power
by Gustavo Duarte 09 Jan '17

09 Jan '17

Hi all, We are facing a problem with audio intput (mic), on a laptop with sound device ALC283. When the laptop boots connected to power line, all works fine. When the laptop boots without power line connection, sound input doesn't work. No audio can be recorder, using arecord or qarecord. I got the output of "arecord -l" comand in each case. Follwoing are the results. With power line connection: **** Lista de CAPTURE dispositivos hardware **** tarjeta 0: PCH [HDA Intel PCH], dispositivo 0: ALC283 Analog [ALC283 Analog] Subdispositivos: 1/1 Subdispositivo #0: subdevice #0 Without power line connection: **** Lista de CAPTURE dispositivos hardware **** tarjeta 0: PCH [HDA Intel PCH], dispositivo 0: ALC283 Analog [ALC283 Analog] Subdispositivos: 0/1 Subdispositivo #0: subdevice #0 As yo can see, the difference is on, Subdispositivos :1/1 (with power line) and Subdispositivos: 0/1 without power line. Anybody can help me resolve this ? Do you need some additional info ? Thanks advance.

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[alsa-devel] Building a new out-of-tree ALSA module on Ubuntu Xenial (was gitcompile fail)
by Mark Brand 09 Jan '17

09 Jan '17

Hi all My abundant thanks to Clemens and Takashi for your kind and patient assistance. May I now answer my own question by offering the following step-by-step procedure in case anyone comes looking for it? Any improvements or suggestions would be greatly appreciated too, thank you. $ sudo apt-get install linux-headers-$(uname -r) $ sudo mkdir /lib/modules/$(uname -r)/updates $ sudo apt-get build-dep linux-image-$(uname -r) $ mkdir $HOME/src; cd $HOME/src $ apt-get source linux-image-$(uname -r) $ cd linux-$(uname -r | awk -F - '{print $1}') Now I am at the root of the kernel source tree. In my case, I want to add a new driver for an as-yet unsupported device. I find a supported device that appears substantially similar to what I need (sound/drivers/mtpav.c) and base my hack on that - producing sound/drivers/mq8portse.c. I then add the following lines to the sound/drivers/Makefile (each mirroring entries found in respect of mtpav.c) snd-mq8portse-objs := mq8portse.o obj-$(CONFIG_SND_MQ8PORTSE) += snd-mq8portse.o Now (from the root of the kernel source tree): $ cd sound/drivers/ $ CONFIG_SND_MQ8PORTSE=m make -C /lib/modules/$(uname -r)/build M=$(pwd) snd-mq8portse.ko This produces my module in the same folder, which I copy to the updates folder. $ sudo cp snd-mq8portse.ko /lib/modules/$(uname -r)/updates/ $ sudo depmod -a Now, because my particular device is attached to the LPT port, I need to unload the lp driver before snd-mq8portse will load. $ sudo modprobe -r lp $ sudo modprobe snd-mq8portse $ modinfo snd-mq8portse I try it out in my ALSA client ... then remove ... $ sudo modprobe -r snd-mq8portse The MusicQuest 8Port/SE is attached to the LPT port, so a more permanent setup requires that I blacklist all typical parport stuff. I add the following as /etc/modprobe.d/blacklist-parport.conf: # keep LPT port open for mq8portse alias parport_pc off alias parport off blacklist parport_pc blacklist parport blacklist ppdev blacklist lp I also comment out autoloading of lp in /etc/modules. I can then add mq8portse to /etc/modules for automatic loading, unless I want to defer to manual modprobe-ing. Once again, thank you ever so much, one and all. Next I need to find out how to submit this driver for inclusion in ALSA. *Mark Brand* Lecturer in Music Technology School of Music, Art & Design Summerstrand Campus (South) Tel: +27 (0)41 5042250 Fax: +27 (0)41 5042507

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[alsa-devel] [PATCH v2 0/2] ASoC: dwc: Add support for record mode
by Jose Abreu 09 Jan '17

09 Jan '17

This patchset adds support for record mode in I2S PIO pcm and enables 24bit transaction. With this patchset we now have PIO support for playback and record using only two channels, at sample rate 32k, 44.1k and 48k and sample width 16/24 and 32 bit. Tested in a x86_64 platform and in ARC AXS101 SDP platform. Jose Abreu (2): ASoC: dwc: Add record capability in PIO mode ASoC: dwc: Enable 24 bit sample size in PIO mode sound/soc/dwc/designware_i2s.c | 9 +++- sound/soc/dwc/designware_pcm.c | 99 +++++++++++++++++++++++++++++++++--------- sound/soc/dwc/local.h | 9 +++- 3 files changed, 94 insertions(+), 23 deletions(-) -- Cc: Carlos Palminha <palminha(a)synopsys.com> Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: alsa-devel(a)alsa-project.org Cc: linux-kernel(a)vger.kernel.org v2: - Added missing cover letter 1.9.1

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