February 2019 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH 1/2] ASoC: pcm3060: Add soft reset on probe
by Kirill Marinushkin 12 Feb '19

12 Feb '19

Softly reset registers values on module probe Signed-off-by: Kirill Marinushkin <kmarinushkin(a)birdec.tech> --- sound/soc/codecs/pcm3060.c | 8 ++++++++ 1 file changed, 8 insertions(+) diff --git a/sound/soc/codecs/pcm3060.c b/sound/soc/codecs/pcm3060.c index 6714aa8d9026..543cb86fd764 100644 --- a/sound/soc/codecs/pcm3060.c +++ b/sound/soc/codecs/pcm3060.c @@ -287,6 +287,14 @@ int pcm3060_probe(struct device *dev) int rc; struct pcm3060_priv *priv = dev_get_drvdata(dev); + /* soft reset */ + rc = regmap_update_bits(priv->regmap, PCM3060_REG64, + PCM3060_REG_MRST, 0); + if (rc) { + dev_err(dev, "failed to reset component, rc=%d\n", rc); + return rc; + } + if (dev->of_node) pcm3060_parse_dt(dev->of_node, priv); -- 2.13.6

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[alsa-devel] [PATCH] ASoC: mediatek: use %pOFn instead of device_node.name
by KaiChieh Chuang 12 Feb '19

12 Feb '19

In preparation to remove the node name pointer from struct device_node, convert printf users to use the %pOFn format specifier. Signed-off-by: KaiChieh Chuang <kaichieh.chuang(a)mediatek.com> --- sound/soc/mediatek/common/mtk-btcvsd.c | 2 +- sound/soc/mediatek/mt8183/mt8183-afe-pcm.c | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/mediatek/common/mtk-btcvsd.c b/sound/soc/mediatek/common/mtk-btcvsd.c index e408c1b..4b613d5 100644 --- a/sound/soc/mediatek/common/mtk-btcvsd.c +++ b/sound/soc/mediatek/common/mtk-btcvsd.c @@ -1269,7 +1269,7 @@ static int mtk_btcvsd_snd_probe(struct platform_device *pdev) /* irq */ irq_id = platform_get_irq(pdev, 0); if (irq_id <= 0) { - dev_err(dev, "%s no irq found\n", dev->of_node->name); + dev_err(dev, "%pOFn no irq found\n", dev->of_node); return irq_id < 0 ? irq_id : -ENXIO; } diff --git a/sound/soc/mediatek/mt8183/mt8183-afe-pcm.c b/sound/soc/mediatek/mt8183/mt8183-afe-pcm.c index ff3111e..4e045dd 100644 --- a/sound/soc/mediatek/mt8183/mt8183-afe-pcm.c +++ b/sound/soc/mediatek/mt8183/mt8183-afe-pcm.c @@ -1139,7 +1139,7 @@ static int mt8183_afe_pcm_dev_probe(struct platform_device *pdev) /* request irq */ irq_id = platform_get_irq(pdev, 0); if (!irq_id) { - dev_err(dev, "%s no irq found\n", dev->of_node->name); + dev_err(dev, "%pOFn no irq found\n", dev->of_node); return -ENXIO; } ret = devm_request_irq(dev, irq_id, mt8183_afe_irq_handler, -- 1.9.1

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[alsa-devel] [PATCH] ASoC: hdac_hdmi: use devm_kzalloc for all structures
by Pierre-Louis Bossart 12 Feb '19

12 Feb '19

Loading/unloading modules exposes issues with memory allocation, which is a mix of devm_kzalloc and manual kzalloc. Move to devm_k routines everywhere to simplify all this. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/soc/codecs/hdac_hdmi.c | 87 ++++++++---------------------------- 1 file changed, 18 insertions(+), 69 deletions(-) diff --git a/sound/soc/codecs/hdac_hdmi.c b/sound/soc/codecs/hdac_hdmi.c index b19d7a3e7a2c..5eeb0fe836a9 100644 --- a/sound/soc/codecs/hdac_hdmi.c +++ b/sound/soc/codecs/hdac_hdmi.c @@ -1176,13 +1176,15 @@ static int hdac_hdmi_add_cvt(struct hdac_device *hdev, hda_nid_t nid) struct hdac_hdmi_cvt *cvt; char name[NAME_SIZE]; - cvt = kzalloc(sizeof(*cvt), GFP_KERNEL); + cvt = devm_kzalloc(&hdev->dev, sizeof(*cvt), GFP_KERNEL); if (!cvt) return -ENOMEM; cvt->nid = nid; sprintf(name, "cvt %d", cvt->nid); - cvt->name = kstrdup(name, GFP_KERNEL); + cvt->name = devm_kstrdup(&hdev->dev, name, GFP_KERNEL); + if (!cvt->name) + return -ENOMEM; list_add_tail(&cvt->head, &hdmi->cvt_list); hdmi->num_cvt++; @@ -1287,8 +1289,8 @@ static void hdac_hdmi_present_sense(struct hdac_hdmi_pin *pin, mutex_unlock(&hdmi->pin_mutex); } -static int hdac_hdmi_add_ports(struct hdac_hdmi_priv *hdmi, - struct hdac_hdmi_pin *pin) +static int hdac_hdmi_add_ports(struct hdac_device *hdev, + struct hdac_hdmi_pin *pin) { struct hdac_hdmi_port *ports; int max_ports = HDA_MAX_PORTS; @@ -1300,7 +1302,7 @@ static int hdac_hdmi_add_ports(struct hdac_hdmi_priv *hdmi, * implemented. */ - ports = kcalloc(max_ports, sizeof(*ports), GFP_KERNEL); + ports = devm_kcalloc(&hdev->dev, max_ports, sizeof(*ports), GFP_KERNEL); if (!ports) return -ENOMEM; @@ -1319,14 +1321,14 @@ static int hdac_hdmi_add_pin(struct hdac_device *hdev, hda_nid_t nid) struct hdac_hdmi_pin *pin; int ret; - pin = kzalloc(sizeof(*pin), GFP_KERNEL); + pin = devm_kzalloc(&hdev->dev, sizeof(*pin), GFP_KERNEL); if (!pin) return -ENOMEM; pin->nid = nid; pin->mst_capable = false; pin->hdev = hdev; - ret = hdac_hdmi_add_ports(hdmi, pin); + ret = hdac_hdmi_add_ports(hdev, pin); if (ret < 0) return ret; @@ -1468,8 +1470,6 @@ static int hdac_hdmi_parse_and_map_nid(struct hdac_device *hdev, { hda_nid_t nid; int i, num_nodes; - struct hdac_hdmi_cvt *temp_cvt, *cvt_next; - struct hdac_hdmi_pin *temp_pin, *pin_next; struct hdac_hdmi_priv *hdmi = hdev_to_hdmi_priv(hdev); int ret; @@ -1497,51 +1497,35 @@ static int hdac_hdmi_parse_and_map_nid(struct hdac_device *hdev, case AC_WID_AUD_OUT: ret = hdac_hdmi_add_cvt(hdev, nid); if (ret < 0) - goto free_widgets; + return ret; break; case AC_WID_PIN: ret = hdac_hdmi_add_pin(hdev, nid); if (ret < 0) - goto free_widgets; + return ret; break; } } if (!hdmi->num_pin || !hdmi->num_cvt) { ret = -EIO; - goto free_widgets; + dev_err(&hdev->dev, "Bad pin/cvt setup in %s\n", __func__); + return ret; } ret = hdac_hdmi_create_dais(hdev, dais, hdmi, hdmi->num_cvt); if (ret) { dev_err(&hdev->dev, "Failed to create dais with err: %d\n", - ret); - goto free_widgets; + ret); + return ret; } *num_dais = hdmi->num_cvt; ret = hdac_hdmi_init_dai_map(hdev); if (ret < 0) - goto free_widgets; - - return ret; - -free_widgets: - list_for_each_entry_safe(temp_cvt, cvt_next, &hdmi->cvt_list, head) { - list_del(&temp_cvt->head); - kfree(temp_cvt->name); - kfree(temp_cvt); - } - - list_for_each_entry_safe(temp_pin, pin_next, &hdmi->pin_list, head) { - for (i = 0; i < temp_pin->num_ports; i++) - temp_pin->ports[i].pin = NULL; - kfree(temp_pin->ports); - list_del(&temp_pin->head); - kfree(temp_pin); - } - + dev_err(&hdev->dev, "Failed to init DAI map with err: %d\n", + ret); return ret; } @@ -1782,7 +1766,7 @@ int hdac_hdmi_jack_init(struct snd_soc_dai *dai, int device, * this is a new PCM device, create new pcm and * add to the pcm list */ - pcm = kzalloc(sizeof(*pcm), GFP_KERNEL); + pcm = devm_kzalloc(&hdev->dev, sizeof(*pcm), GFP_KERNEL); if (!pcm) return -ENOMEM; pcm->pcm_id = device; @@ -1798,7 +1782,6 @@ int hdac_hdmi_jack_init(struct snd_soc_dai *dai, int device, dev_err(&hdev->dev, "chmap control add failed with err: %d for pcm: %d\n", err, device); - kfree(pcm); return err; } } @@ -2075,42 +2058,8 @@ static int hdac_hdmi_dev_probe(struct hdac_device *hdev) static int hdac_hdmi_dev_remove(struct hdac_device *hdev) { - struct hdac_hdmi_priv *hdmi = hdev_to_hdmi_priv(hdev); - struct hdac_hdmi_pin *pin, *pin_next; - struct hdac_hdmi_cvt *cvt, *cvt_next; - struct hdac_hdmi_pcm *pcm, *pcm_next; - struct hdac_hdmi_port *port, *port_next; - int i; - snd_hdac_display_power(hdev->bus, hdev->addr, false); - list_for_each_entry_safe(pcm, pcm_next, &hdmi->pcm_list, head) { - pcm->cvt = NULL; - if (list_empty(&pcm->port_list)) - continue; - - list_for_each_entry_safe(port, port_next, - &pcm->port_list, head) - list_del(&port->head); - - list_del(&pcm->head); - kfree(pcm); - } - - list_for_each_entry_safe(cvt, cvt_next, &hdmi->cvt_list, head) { - list_del(&cvt->head); - kfree(cvt->name); - kfree(cvt); - } - - list_for_each_entry_safe(pin, pin_next, &hdmi->pin_list, head) { - for (i = 0; i < pin->num_ports; i++) - pin->ports[i].pin = NULL; - kfree(pin->ports); - list_del(&pin->head); - kfree(pin); - } - return 0; } -- 2.17.1

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[alsa-devel] [PATCH v3] ASoC: wm8741: Add digital mute callback
by Sergej Sawazki 12 Feb '19

12 Feb '19

Signed-off-by: Sergej Sawazki <sergej(a)taudac.com> --- Changes in v3: * Removed debug message. sound/soc/codecs/wm8741.c | 10 ++++++++++ 1 file changed, 10 insertions(+) diff --git a/sound/soc/codecs/wm8741.c b/sound/soc/codecs/wm8741.c index 1fedf74..fdda83b 100644 --- a/sound/soc/codecs/wm8741.c +++ b/sound/soc/codecs/wm8741.c @@ -358,6 +358,15 @@ static int wm8741_set_dai_fmt(struct snd_soc_dai *codec_dai, return 0; } +int wm8741_mute(struct snd_soc_dai *codec_dai, int mute) +{ + struct snd_soc_component *component = codec_dai->component; + + snd_soc_component_update_bits(component, WM8741_VOLUME_CONTROL, + WM8741_SOFT_MASK, !!mute << WM8741_SOFT_SHIFT); + return 0; +} + #define WM8741_RATES (SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \ SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 | \ SNDRV_PCM_RATE_96000 | SNDRV_PCM_RATE_176400 | \ @@ -371,6 +380,7 @@ static const struct snd_soc_dai_ops wm8741_dai_ops = { .hw_params = wm8741_hw_params, .set_sysclk = wm8741_set_dai_sysclk, .set_fmt = wm8741_set_dai_fmt, + .digital_mute = wm8741_mute, }; static struct snd_soc_dai_driver wm8741_dai = { -- 2.7.4

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[alsa-devel] [PATCH v2] ASoC: wm8741: Set OSR mode in hw_params()
by Sergej Sawazki 12 Feb '19

12 Feb '19

For correct operation of the digital filtering and other processing on the WM8741, the user must ensure the correct value of OSR[1:0] is set at all times.[1] Hence, depending the selected sampling rate, set the OSR (over- sampling rate) mode in hw_params(). References: [1] "WM8741 Data Sheet" Signed-off-by: Sergej Sawazki <sergej(a)taudac.com> --- Changes in v2: * Using update_bits() rather than write(). sound/soc/codecs/wm8741.c | 12 +++++++++++- 1 file changed, 11 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm8741.c b/sound/soc/codecs/wm8741.c index fdda83b..a4b8c45 100644 --- a/sound/soc/codecs/wm8741.c +++ b/sound/soc/codecs/wm8741.c @@ -196,7 +196,7 @@ static int wm8741_hw_params(struct snd_pcm_substream *substream, { struct snd_soc_component *component = dai->component; struct wm8741_priv *wm8741 = snd_soc_component_get_drvdata(component); - unsigned int iface; + unsigned int iface, mode; int i; /* The set of sample rates that can be supported depends on the @@ -240,11 +240,21 @@ static int wm8741_hw_params(struct snd_pcm_substream *substream, return -EINVAL; } + /* oversampling rate */ + if (params_rate(params) > 96000) + mode = 0x40; + else if (params_rate(params) > 48000) + mode = 0x20; + else + mode = 0x00; + dev_dbg(component->dev, "wm8741_hw_params: bit size param = %d, rate param = %d", params_width(params), params_rate(params)); snd_soc_component_update_bits(component, WM8741_FORMAT_CONTROL, WM8741_IWL_MASK, iface); + snd_soc_component_update_bits(component, WM8741_MODE_CONTROL_1, WM8741_OSR_MASK, + mode); return 0; } -- 2.7.4

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[alsa-devel] [PATCH] ASoC: mediatek: btcvsd fix rx stream assign
by KaiChieh Chuang 12 Feb '19

12 Feb '19

fix tx/rx stream assign wrong direction Signed-off-by: KaiChieh Chuang <kaichieh.chuang(a)mediatek.com> --- sound/soc/mediatek/common/mtk-btcvsd.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/mediatek/common/mtk-btcvsd.c b/sound/soc/mediatek/common/mtk-btcvsd.c index 4b613d5..1b8bcdaf 100644 --- a/sound/soc/mediatek/common/mtk-btcvsd.c +++ b/sound/soc/mediatek/common/mtk-btcvsd.c @@ -250,7 +250,7 @@ static int mtk_btcvsd_snd_rx_init(struct mtk_btcvsd_snd *bt) bt->rx->buf_size = BTCVSD_RX_BUF_SIZE; bt->rx->timeout = 0; bt->rx->rw_cnt = 0; - bt->tx->stream = SNDRV_PCM_STREAM_CAPTURE; + bt->rx->stream = SNDRV_PCM_STREAM_CAPTURE; return 0; } -- 1.9.1

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[alsa-devel] [RFC v2] tlv320aic32x4: delay i2c access by 1 ms after hardware reset
by Peter Seiderer 12 Feb '19

12 Feb '19

As stated in 'TLV320AIC3254 Application Reference Guide' ([1]): 3.2 Device Startup Lockout Times After the TLV320AIC3254 initializes through hardware reset at power-up or software reset, the internal registers initialize to default values. This initialization takes place within 1ms after pulling the RESET signal high. During this initialization phase, no register-read or register-write operation should be performed on ADC or DAC coefficient buffers. Also, no block within the codec should be powered up during the initialization phase. [1] http://www.ti.com/lit/an/slaa408a/slaa408a.pdf Signed-off-by: Peter Seiderer <ps.report(a)gmx.net> --- Changes v1 -> v2: - call mdelay only in case the reset gpio was toggled (suggested by Mark Brown) Note: This came up after the electronic departement took a deeper look at the i2c-bus/vcc/reset pins with a logic analyzer and the documentation because of a vcc/reset timing problem (now fixed on custom board). Not 100% sure if the 1 ms is needed (as restricted to ADC/DAC coefficent and codec block power up). --- sound/soc/codecs/tlv320aic32x4.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/tlv320aic32x4.c b/sound/soc/codecs/tlv320aic32x4.c index e1bfba62fc08..96f1526cb258 100644 --- a/sound/soc/codecs/tlv320aic32x4.c +++ b/sound/soc/codecs/tlv320aic32x4.c @@ -970,6 +970,7 @@ static int aic32x4_component_probe(struct snd_soc_component *component) if (gpio_is_valid(aic32x4->rstn_gpio)) { ndelay(10); gpio_set_value(aic32x4->rstn_gpio, 1); + mdelay(1); } snd_soc_component_write(component, AIC32X4_RESET, 0x01); -- 2.20.1

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[alsa-devel] [PATCH] ASoC: Use __printf markup to silence compiler
by Mathieu Malaterre 12 Feb '19

12 Feb '19

Silence warnings (triggered at W=1) by adding relevant __printf attributes. sound/soc/soc-dapm.c:149:2: warning: function 'pop_dbg' might be a candidate for 'gnu_printf' format attribute [-Wsuggest-attribute=format] Signed-off-by: Mathieu Malaterre <malat(a)debian.org> --- sound/soc/soc-dapm.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index a5178845065b..92a2e7dab5fe 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -133,6 +133,7 @@ static void pop_wait(u32 pop_time) schedule_timeout_uninterruptible(msecs_to_jiffies(pop_time)); } +__printf(3, 4) static void pop_dbg(struct device *dev, u32 pop_time, const char *fmt, ...) { va_list args; -- 2.19.2

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[alsa-devel] [PATCH 0/2][RFC] ASoC: soc-core: modern style dai_link control
by Kuninori Morimoto 12 Feb '19

12 Feb '19

Hi Mark These are [RFC] patches which adds new rule for modern style dai_link. Many sound are using CPU component as Platform component, and then it has below settings on legacy style dai_link dai_link->platform_of_node = dai_link->cpu_of_node But, this duplicated component will be just ignored because it is already selected when CPU component binding. We want to allow "no Platform" on modern style dai_link. And, current legacy style dai_link allows no Platform settings, then, soc-core will select snd-soc-dummy platform for it automatically. In modern style, we need to have CPU/Codec/Platform as array/pointer style. Thus, this kind of "implicit" selection will be just un-understandable. These 2 patches adds new rule for modern style, and have no effect to legacy style. But, I can't test for *all* board/situation. Thus, I added [RFC] for these. Kuninori Morimoto (2): ASoC: soc-core: allow no Platform on dai_link ASoC: soc-core: no more implicit snd-soc-dummy platform on modern dai_link include/sound/soc.h | 2 +- sound/soc/generic/simple-card-utils.c | 4 ++ sound/soc/soc-core.c | 80 ++++++++++++++++++++++------------- 3 files changed, 56 insertions(+), 30 deletions(-) -- 2.7.4

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[alsa-devel] [PATCH] pcm: dshare: Fix overflow when slave_hw_ptr rolls over boundary
by Brendan Shanks 12 Feb '19

12 Feb '19

In snd_pcm_dshare_sync_area() when 'slave_hw_ptr' rolls over 'slave_boundary', the wrong variable is checked ('dshare->slave_hw_ptr' vs the local 'slave_hw_ptr'). In some cases, this results in 'slave_hw_ptr' not rolling over correctly. 'slave_size' and 'size' are then much too large, and the for loop blocks for several minutes copying samples. This was likely only triggered on 32-bit systems, since the PCM boundary is computed based on LONG_MAX and is much larger on 64-bit systems. This same change was made to pcm_dmix in commit 6c7f60f7a982fdba828e4530a9d7aa0aa2b704ae ("Fix boundary overlap”) from June 2005. Signed-off-by: Brendan Shanks <brendan.shanks(a)teradek.com> diff --git a/src/pcm/pcm_dshare.c b/src/pcm/pcm_dshare.c index 2bb735fe..f135b5df 100644 --- a/src/pcm/pcm_dshare.c +++ b/src/pcm/pcm_dshare.c @@ -121,7 +121,7 @@ static void snd_pcm_dshare_sync_area(snd_pcm_t *pcm) */ slave_hw_ptr -= slave_hw_ptr % dshare->slave_period_size; slave_hw_ptr += dshare->slave_buffer_size; - if (dshare->slave_hw_ptr > dshare->slave_boundary) + if (slave_hw_ptr >= dshare->slave_boundary) slave_hw_ptr -= dshare->slave_boundary; if (slave_hw_ptr < dshare->slave_appl_ptr) slave_size = slave_hw_ptr + (dshare->slave_boundary - dshare->slave_appl_ptr); -- 2.17.1

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