January 2021 - Alsa-devel - mailman.alsa-project.org

[PATCH] sound: pci/rme9652 - implement and expose controls for output
by fassl 02 Feb '21

02 Feb '21

>From 980aa253c17d233966e05a43f3611693e2e02040 Mon Sep 17 00:00:00 2001 From: Jasmin Fazlic <superfassl(a)gmail.com> Date: Sun, 31 Jan 2021 22:17:22 +0100 Subject: [PATCH] sound: pci/rme9652 - implement and expose controls for output loopback - so far only tested and enabled for RME HDSP9632 Signed-off-by: Jasmin Fazlic <superfassl(a)gmail.com> --- sound/pci/rme9652/hdsp.c | 85 +++++++++++++++++++++++++++++++++++++++- 1 file changed, 84 insertions(+), 1 deletion(-) diff --git a/sound/pci/rme9652/hdsp.c b/sound/pci/rme9652/hdsp.c index cea53a878c36..7e832a502cea 100644 --- a/sound/pci/rme9652/hdsp.c +++ b/sound/pci/rme9652/hdsp.c @@ -469,6 +469,7 @@ struct hdsp { unsigned char qs_out_channels; unsigned char ds_out_channels; unsigned char ss_out_channels; + u32 io_loopback; /* output loopback channel states*/ struct snd_dma_buffer capture_dma_buf; struct snd_dma_buffer playback_dma_buf; @@ -3253,6 +3254,71 @@ static const struct snd_kcontrol_new snd_hdsp_96xx_aeb = HDSP_AnalogExtensionBoard); static struct snd_kcontrol_new snd_hdsp_adat_sync_check = HDSP_ADAT_SYNC_CHECK; + +static bool hdsp_loopback_get(struct hdsp *const hdsp, const u8 channel) +{ + return hdsp->io_loopback & (1 << channel); +} + +static int hdsp_loopback_set(struct hdsp *const hdsp, const u8 channel, const bool enable) +{ + if (hdsp_loopback_get(hdsp, channel) == enable) + return 0; + + hdsp->io_loopback ^= (1 << channel); + + hdsp_write(hdsp, HDSP_inputEnable + (4 * (hdsp->max_channels + channel)), enable); + + return 1; +} + +static int snd_hdsp_loopback_info(struct snd_kcontrol *const kcontrol, + struct snd_ctl_elem_info *const uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; + uinfo->count = 1; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = 1; + return 0; +} + +static int snd_hdsp_loopback_get(struct snd_kcontrol *const kcontrol, + struct snd_ctl_elem_value *const ucontrol) +{ + struct hdsp *const hdsp = snd_kcontrol_chip(kcontrol); + const u8 channel = kcontrol->id.index; + + if (channel >= hdsp->max_channels) + return -ENOENT; + + ucontrol->value.integer.value[0] = hdsp_loopback_get(hdsp, channel); + + return 0; +} + +static int snd_hdsp_loopback_put(struct snd_kcontrol *const kcontrol, + struct snd_ctl_elem_value *const ucontrol) +{ + struct hdsp *const hdsp = snd_kcontrol_chip(kcontrol); + const u8 channel = ucontrol->id.index; + const bool enable = ucontrol->value.integer.value[0] & 1; + + if (channel >= hdsp->max_channels) + return -ENOENT; + + return hdsp_loopback_set(hdsp, channel, enable); +} + +static struct snd_kcontrol_new snd_hdsp_loopback_control = { + .iface = SNDRV_CTL_ELEM_IFACE_HWDEP, + .name = "Output Loopback", + .index = 0, + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .info = snd_hdsp_loopback_info, + .get = snd_hdsp_loopback_get, + .put = snd_hdsp_loopback_put +}; + static int snd_hdsp_create_controls(struct snd_card *card, struct hdsp *hdsp) { unsigned int idx; @@ -3297,6 +3363,17 @@ static int snd_hdsp_create_controls(struct snd_card *card, struct hdsp *hdsp) } } + /* Output loopback controls for H9632 cards */ + if (hdsp->io_type == H9632) { + for (idx = 0; idx < hdsp->max_channels; idx++) { + snd_hdsp_loopback_control.index = idx; + kctl = snd_ctl_new1(&snd_hdsp_loopback_control, hdsp); + err = snd_ctl_add(card, kctl); + if (err < 0) + return err; + } + } + /* AEB control for H96xx card */ if (hdsp->io_type == H9632 || hdsp->io_type == H9652) { if ((err = snd_ctl_add(card, kctl = snd_ctl_new1(&snd_hdsp_96xx_aeb, hdsp))) < 0) @@ -4956,7 +5033,7 @@ static int snd_hdsp_enable_io (struct hdsp *hdsp) static void snd_hdsp_initialize_channels(struct hdsp *hdsp) { - int status, aebi_channels, aebo_channels; + int status, aebi_channels, aebo_channels, i; switch (hdsp->io_type) { case Digiface: @@ -4983,6 +5060,12 @@ static void snd_hdsp_initialize_channels(struct hdsp *hdsp) hdsp->ss_out_channels = H9632_SS_CHANNELS+aebo_channels; hdsp->ds_out_channels = H9632_DS_CHANNELS+aebo_channels; hdsp->qs_out_channels = H9632_QS_CHANNELS+aebo_channels; + /* Disable loopback of output channels, as the set function + * only sets on a change we fake all bits (channels) as enabled. + */ + hdsp->io_loopback = 0xffffffff; + for (i = 0; i < hdsp->max_channels; ++i) + hdsp_loopback_set(hdsp, i, false); break; case Multiface: -- 2.27.0

2 5

[PATCH] ASoC: tegra: SND_SOC_TEGRA_AUDIO_GRAPH_CARD should depend on SND_SOC_TEGRA
by Geert Uytterhoeven 01 Feb '21

01 Feb '21

Audio Graph Card based Tegra driver is only useful on NVIDIA Tegra SoCs. Hence add a dependency on SND_SOC_TEGRA, to prevent asking the user about this driver when configuring a kernel without Tegra sound support. Wrap all Tegra sound config options inside a big if/endif block, instead of just adding the dependency to the single config option that does not have it yet, to preventing similar future mistakes. Fixes: 202e2f7745437aa5 ("ASoC: tegra: Add audio graph based card driver") Signed-off-by: Geert Uytterhoeven <geert+renesas(a)glider.be> --- sound/soc/tegra/Kconfig | 33 +++++++++++++-------------------- 1 file changed, 13 insertions(+), 20 deletions(-) diff --git a/sound/soc/tegra/Kconfig b/sound/soc/tegra/Kconfig index 6dc83ad8607a1de2..a4e6760944d028cf 100644 --- a/sound/soc/tegra/Kconfig +++ b/sound/soc/tegra/Kconfig @@ -9,9 +9,10 @@ config SND_SOC_TEGRA help Say Y or M here if you want support for SoC audio on Tegra. +if SND_SOC_TEGRA + config SND_SOC_TEGRA20_AC97 tristate "Tegra20 AC97 interface" - depends on SND_SOC_TEGRA select SND_SOC_AC97_BUS select SND_SOC_TEGRA20_DAS help @@ -21,7 +22,6 @@ config SND_SOC_TEGRA20_AC97 config SND_SOC_TEGRA20_DAS tristate "Tegra20 DAS module" - depends on SND_SOC_TEGRA help Say Y or M if you want to add support for the Tegra20 DAS module. You will also need to select the individual machine drivers to @@ -29,7 +29,6 @@ config SND_SOC_TEGRA20_DAS config SND_SOC_TEGRA20_I2S tristate "Tegra20 I2S interface" - depends on SND_SOC_TEGRA select SND_SOC_TEGRA20_DAS help Say Y or M if you want to add support for codecs attached to the @@ -38,7 +37,6 @@ config SND_SOC_TEGRA20_I2S config SND_SOC_TEGRA20_SPDIF tristate "Tegra20 SPDIF interface" - depends on SND_SOC_TEGRA help Say Y or M if you want to add support for the Tegra20 SPDIF interface. You will also need to select the individual machine drivers to support @@ -46,7 +44,6 @@ config SND_SOC_TEGRA20_SPDIF config SND_SOC_TEGRA30_AHUB tristate "Tegra30 AHUB module" - depends on SND_SOC_TEGRA help Say Y or M if you want to add support for the Tegra30 AHUB module. You will also need to select the individual machine drivers to @@ -54,7 +51,6 @@ config SND_SOC_TEGRA30_AHUB config SND_SOC_TEGRA30_I2S tristate "Tegra30 I2S interface" - depends on SND_SOC_TEGRA select SND_SOC_TEGRA30_AHUB help Say Y or M if you want to add support for codecs attached to the @@ -63,7 +59,6 @@ config SND_SOC_TEGRA30_I2S config SND_SOC_TEGRA210_AHUB tristate "Tegra210 AHUB module" - depends on SND_SOC_TEGRA help Config to enable Audio Hub (AHUB) module, which comprises of a switch called Audio Crossbar (AXBAR) used to configure or modify @@ -73,7 +68,6 @@ config SND_SOC_TEGRA210_AHUB config SND_SOC_TEGRA210_DMIC tristate "Tegra210 DMIC module" - depends on SND_SOC_TEGRA help Config to enable the Digital MIC (DMIC) controller which is used to interface with Pulse Density Modulation (PDM) input devices. @@ -84,7 +78,6 @@ config SND_SOC_TEGRA210_DMIC config SND_SOC_TEGRA210_I2S tristate "Tegra210 I2S module" - depends on SND_SOC_TEGRA help Config to enable the Inter-IC Sound (I2S) Controller which implements full-duplex and bidirectional and single direction @@ -94,7 +87,6 @@ config SND_SOC_TEGRA210_I2S config SND_SOC_TEGRA186_DSPK tristate "Tegra186 DSPK module" - depends on SND_SOC_TEGRA help Config to enable the Digital Speaker Controller (DSPK) which converts the multi-bit Pulse Code Modulation (PCM) audio input to @@ -107,7 +99,6 @@ config SND_SOC_TEGRA186_DSPK config SND_SOC_TEGRA210_ADMAIF tristate "Tegra210 ADMAIF module" - depends on SND_SOC_TEGRA help Config to enable ADMAIF which is the interface between ADMA and Audio Hub (AHUB). Each ADMA channel that sends/receives data to/ @@ -128,7 +119,7 @@ config SND_SOC_TEGRA_AUDIO_GRAPH_CARD config SND_SOC_TEGRA_RT5640 tristate "SoC Audio support for Tegra boards using an RT5640 codec" - depends on SND_SOC_TEGRA && I2C && GPIOLIB + depends on I2C && GPIOLIB select SND_SOC_RT5640 help Say Y or M here if you want to add support for SoC audio on Tegra @@ -136,7 +127,7 @@ config SND_SOC_TEGRA_RT5640 config SND_SOC_TEGRA_WM8753 tristate "SoC Audio support for Tegra boards using a WM8753 codec" - depends on SND_SOC_TEGRA && I2C && GPIOLIB + depends on I2C && GPIOLIB select SND_SOC_WM8753 help Say Y or M here if you want to add support for SoC audio on Tegra @@ -144,7 +135,7 @@ config SND_SOC_TEGRA_WM8753 config SND_SOC_TEGRA_WM8903 tristate "SoC Audio support for Tegra boards using a WM8903 codec" - depends on SND_SOC_TEGRA && I2C && GPIOLIB + depends on I2C && GPIOLIB select SND_SOC_WM8903 help Say Y or M here if you want to add support for SoC audio on Tegra @@ -153,7 +144,7 @@ config SND_SOC_TEGRA_WM8903 config SND_SOC_TEGRA_WM9712 tristate "SoC Audio support for Tegra boards using a WM9712 codec" - depends on SND_SOC_TEGRA && GPIOLIB + depends on GPIOLIB select SND_SOC_TEGRA20_AC97 select SND_SOC_WM9712 help @@ -162,7 +153,7 @@ config SND_SOC_TEGRA_WM9712 config SND_SOC_TEGRA_TRIMSLICE tristate "SoC Audio support for TrimSlice board" - depends on SND_SOC_TEGRA && I2C + depends on I2C select SND_SOC_TLV320AIC23_I2C help Say Y or M here if you want to add support for SoC audio on the @@ -170,7 +161,7 @@ config SND_SOC_TEGRA_TRIMSLICE config SND_SOC_TEGRA_ALC5632 tristate "SoC Audio support for Tegra boards using an ALC5632 codec" - depends on SND_SOC_TEGRA && I2C && GPIOLIB + depends on I2C && GPIOLIB select SND_SOC_ALC5632 help Say Y or M here if you want to add support for SoC audio on the @@ -178,7 +169,7 @@ config SND_SOC_TEGRA_ALC5632 config SND_SOC_TEGRA_MAX98090 tristate "SoC Audio support for Tegra boards using a MAX98090 codec" - depends on SND_SOC_TEGRA && I2C && GPIOLIB + depends on I2C && GPIOLIB select SND_SOC_MAX98090 help Say Y or M here if you want to add support for SoC audio on Tegra @@ -186,7 +177,7 @@ config SND_SOC_TEGRA_MAX98090 config SND_SOC_TEGRA_RT5677 tristate "SoC Audio support for Tegra boards using a RT5677 codec" - depends on SND_SOC_TEGRA && I2C && GPIOLIB + depends on I2C && GPIOLIB select SND_SOC_RT5677 help Say Y or M here if you want to add support for SoC audio on Tegra @@ -194,9 +185,11 @@ config SND_SOC_TEGRA_RT5677 config SND_SOC_TEGRA_SGTL5000 tristate "SoC Audio support for Tegra boards using a SGTL5000 codec" - depends on SND_SOC_TEGRA && I2C && GPIOLIB + depends on I2C && GPIOLIB select SND_SOC_SGTL5000 help Say Y or M here if you want to add support for SoC audio on Tegra boards using the SGTL5000 codec, such as Apalis T30, Apalis TK1 or Colibri T30. + +endif -- 2.25.1

3 2

[bug report] ASoC: cpcap: new codec
by Dan Carpenter 01 Feb '21

01 Feb '21

Hello Sebastian Reichel, The patch f6cdf2d3445d: "ASoC: cpcap: new codec" from Feb 23, 2018, leads to the following static checker warning: sound/soc/codecs/cpcap.c:1276 cpcap_voice_hw_params() warn: 'CPCAP_BIT_MIC1_RX_TIMESLOT0' is a shifter (not for '|='). sound/soc/codecs/cpcap.c:1279 cpcap_voice_hw_params() warn: 'CPCAP_BIT_MIC2_TIMESLOT0' is a shifter (not for '|='). sound/soc/codecs/cpcap.c 1253 static int cpcap_voice_hw_params(struct snd_pcm_substream *substream, 1254 struct snd_pcm_hw_params *params, 1255 struct snd_soc_dai *dai) 1256 { 1257 struct snd_soc_component *component = dai->component; 1258 struct device *dev = component->dev; 1259 struct cpcap_audio *cpcap = snd_soc_component_get_drvdata(component); 1260 static const u16 reg_cdi = CPCAP_REG_CDI; 1261 int rate = params_rate(params); 1262 int channels = params_channels(params); 1263 int direction = substream->stream; 1264 u16 val, mask; 1265 int err; 1266 1267 dev_dbg(dev, "Voice setup HW params: rate=%d, direction=%d, chan=%d", 1268 rate, direction, channels); 1269 1270 err = cpcap_set_samprate(cpcap, CPCAP_DAI_VOICE, rate); 1271 if (err) 1272 return err; 1273 1274 if (direction == SNDRV_PCM_STREAM_CAPTURE) { 1275 mask = 0x0000; 1276 mask |= CPCAP_BIT_MIC1_RX_TIMESLOT0; ^^^^^^^^^^^^^^^^^^^^^^^^^^^ This should probably be BIT(CPCAP_BIT_MIC1_RX_TIMESLOT0). 1277 mask |= CPCAP_BIT_MIC1_RX_TIMESLOT1; 1278 mask |= CPCAP_BIT_MIC1_RX_TIMESLOT2; 1279 mask |= CPCAP_BIT_MIC2_TIMESLOT0; ^^^^^^^^^^^^^^^^^^^^^^^^ Same for all the others as well I guess. 1280 mask |= CPCAP_BIT_MIC2_TIMESLOT1; 1281 mask |= CPCAP_BIT_MIC2_TIMESLOT2; 1282 val = 0x0000; 1283 if (channels >= 2) 1284 val = BIT(CPCAP_BIT_MIC1_RX_TIMESLOT0); ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ Here it's BIT(CPCAP_BIT_MIC1_RX_TIMESLOT0). 1285 err = regmap_update_bits(cpcap->regmap, reg_cdi, mask, val); 1286 if (err) 1287 return err; 1288 } 1289 1290 return 0; 1291 } regards, dan carpenter

4 3

[PATCH v2] ASoC: qcom: lpass: Fix i2s ctl register bit map
by Jun Nie 01 Feb '21

01 Feb '21

Fix bitwidth mapping in i2s ctl register per APQ8016 document. Fixes: b5022a36d28f ("ASoC: qcom: lpass: Use regmap_field for i2sctl and dmactl registers") Signed-off-by: Jun Nie <jun.nie(a)linaro.org> --- sound/soc/qcom/lpass-apq8016.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/qcom/lpass-apq8016.c b/sound/soc/qcom/lpass-apq8016.c index 8507ef8f6679..3efa133d1c64 100644 --- a/sound/soc/qcom/lpass-apq8016.c +++ b/sound/soc/qcom/lpass-apq8016.c @@ -250,7 +250,7 @@ static struct lpass_variant apq8016_data = { .micmode = REG_FIELD_ID(0x1000, 4, 7, 4, 0x1000), .micmono = REG_FIELD_ID(0x1000, 3, 3, 4, 0x1000), .wssrc = REG_FIELD_ID(0x1000, 2, 2, 4, 0x1000), - .bitwidth = REG_FIELD_ID(0x1000, 0, 0, 4, 0x1000), + .bitwidth = REG_FIELD_ID(0x1000, 0, 1, 4, 0x1000), .rdma_dyncclk = REG_FIELD_ID(0x8400, 12, 12, 2, 0x1000), .rdma_bursten = REG_FIELD_ID(0x8400, 11, 11, 2, 0x1000), -- 2.25.1

3 3

[PATCH] soundwire: return earlier if no slave is attached
by Bard Liao 01 Feb '21

01 Feb '21

From: Chao Song <chao.song(a)linux.intel.com> If there is no slave attached to soundwire bus, we can return earlier from sdw_bus_prep_clk_stop() and sdw_bus_exit_clk_stop(), this saves a redundant value check. Signed-off-by: Chao Song <chao.song(a)linux.intel.com> Reviewed-by: Rander Wang <rander.wang(a)intel.com> Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> --- drivers/soundwire/bus.c | 23 ++++++++++------------- 1 file changed, 10 insertions(+), 13 deletions(-) diff --git a/drivers/soundwire/bus.c b/drivers/soundwire/bus.c index 82df088c9333..d9deafdcf495 100644 --- a/drivers/soundwire/bus.c +++ b/drivers/soundwire/bus.c @@ -951,17 +951,17 @@ int sdw_bus_prep_clk_stop(struct sdw_bus *bus) simple_clk_stop = false; } - if (is_slave && !simple_clk_stop) { + /* Skip remaining clock stop preparation if no Slave is attached */ + if (!is_slave) + return ret; + + if (!simple_clk_stop) { ret = sdw_bus_wait_for_clk_prep_deprep(bus, SDW_BROADCAST_DEV_NUM); if (ret < 0) return ret; } - /* Don't need to inform slaves if there is no slave attached */ - if (!is_slave) - return ret; - /* Inform slaves that prep is done */ list_for_each_entry(slave, &bus->slaves, node) { if (!slave->dev_num) @@ -1075,16 +1075,13 @@ int sdw_bus_exit_clk_stop(struct sdw_bus *bus) "clk stop deprep failed:%d", ret); } - if (is_slave && !simple_clk_stop) + /* Skip remaining clock stop de-preparation if no Slave is attached */ + if (!is_slave) + return 0; + + if (!simple_clk_stop) sdw_bus_wait_for_clk_prep_deprep(bus, SDW_BROADCAST_DEV_NUM); - /* - * Don't need to call slave callback function if there is no slave - * attached - */ - if (!is_slave) - return 0; - list_for_each_entry(slave, &bus->slaves, node) { if (!slave->dev_num) continue; -- 2.17.1

2 1

[PATCH] soundwire: bus: add better dev_dbg to track complete() calls
by Bard Liao 01 Feb '21

01 Feb '21

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Add a dev_dbg() log for both enumeration and initialization completion to better track suspend-resume issues. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Rander Wang <rander.wang(a)intel.com> Reviewed-by: Chao Song <chao.song(a)intel.com> Signed-off-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> --- drivers/soundwire/bus.c | 11 ++++++++--- 1 file changed, 8 insertions(+), 3 deletions(-) diff --git a/drivers/soundwire/bus.c b/drivers/soundwire/bus.c index 6e1c988f3845..82df088c9333 100644 --- a/drivers/soundwire/bus.c +++ b/drivers/soundwire/bus.c @@ -784,7 +784,7 @@ static void sdw_modify_slave_status(struct sdw_slave *slave, if (status == SDW_SLAVE_UNATTACHED) { dev_dbg(&slave->dev, - "%s: initializing completion for Slave %d\n", + "%s: initializing enumeration and init completion for Slave %d\n", __func__, slave->dev_num); init_completion(&slave->enumeration_complete); @@ -793,7 +793,7 @@ static void sdw_modify_slave_status(struct sdw_slave *slave, } else if ((status == SDW_SLAVE_ATTACHED) && (slave->status == SDW_SLAVE_UNATTACHED)) { dev_dbg(&slave->dev, - "%s: signaling completion for Slave %d\n", + "%s: signaling enumeration completion for Slave %d\n", __func__, slave->dev_num); complete(&slave->enumeration_complete); @@ -1758,8 +1758,13 @@ int sdw_handle_slave_status(struct sdw_bus *bus, if (ret) dev_err(slave->bus->dev, "Update Slave status failed:%d\n", ret); - if (attached_initializing) + if (attached_initializing) { + dev_dbg(&slave->dev, + "%s: signaling initialization completion for Slave %d\n", + __func__, slave->dev_num); + complete(&slave->initialization_complete); + } } return ret; -- 2.17.1

2 1

jack doesn't detected plugged after suspend
by Mert Can Demir 01 Feb '21

01 Feb '21

Hi, I opened an issue on pulseaudio repository at https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/issues/1125 about the problem about the detection of jack output, and I have been forwarded there, that mailing list, at https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/issues/1125#note_784…. The related outputs and logs are baked into the issue, and seems that the outputs tells that alsa driver did not reported jack as plugged after suspend. Thanks for understanding, Sincerely, validatedev

2 1

[PATCH] ALSA: intel8x0: Fix missing check in snd_intel8x0m_create
by Dinghao Liu 01 Feb '21

01 Feb '21

When device_type == DEVICE_ALI, we should also check the return value of pci_iomap() to avoid potential null pointer dereference. Signed-off-by: Dinghao Liu <dinghao.liu(a)zju.edu.cn> --- sound/pci/intel8x0m.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/pci/intel8x0m.c b/sound/pci/intel8x0m.c index 1b7df0c4e57c..19872cecc9d2 100644 --- a/sound/pci/intel8x0m.c +++ b/sound/pci/intel8x0m.c @@ -1129,13 +1129,14 @@ static int snd_intel8x0m_create(struct snd_card *card, chip->bmaddr = pci_iomap(pci, 3, 0); else chip->bmaddr = pci_iomap(pci, 1, 0); + +port_inited: if (!chip->bmaddr) { dev_err(card->dev, "Controller space ioremap problem\n"); snd_intel8x0m_free(chip); return -EIO; } - port_inited: /* initialize offsets */ chip->bdbars_count = 2; tbl = intel_regs; -- 2.17.1

2 1

[PATCH 0/2] ALSA: Constify static attribute_group structs
by Rikard Falkeborn 31 Jan '21

31 Jan '21

Constify two static attribute_group structs that are never modified. These two were the only static non-const ones in sound/. Rikard Falkeborn (2): ALSA: ac97: Constify static struct attribute_group ALSA: hda: Constify static attribute_group sound/ac97/bus.c | 2 +- sound/hda/hdac_sysfs.c | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) -- 2.30.0

2 4

Extend card search paths
by GitHub pull_request - opened 30 Jan '21

30 Jan '21

alsa-project/alsa-ucm-conf pull request #76 was opened from lumag: On Qualcomm platforms is is typical for a single kernel module to support several sound cards. Add support for using `ucm2/module/${KernelModule}/${CardLongName}.conf` as a use case config file. See https://github.com/alsa-project/alsa-ucm-conf/pull/70 for the reference. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/76 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/76.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0