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

[alsa-devel] [PATCHv1 0/5] ASoC: da7213: support for usage with simple-card
by Sebastian Reichel 13 Nov '19

13 Nov '19

Hi, This extends the da7213 driver to be used with simple-audio-card in combination with a fixed clock. Here is a snippet of the downstream board's DT, that is supposed to be supported by this patchset. --------------------------------------------------------------------- / { sound { compatible = "simple-audio-card"; simple-audio-card,name = "audio-card"; simple-audio-card,format = "i2s"; simple-audio-card,bitclock-master = <&dailink_master>; simple-audio-card,frame-master = <&dailink_master>; simple-audio-card,widgets = "Speaker", "Ext Spk"; simple-audio-card,audio-routing = "Ext Spk", "LINE"; simple-audio-card,cpu { sound-dai = <&ssi1>; }; dailink_master: simple-audio-card,codec { sound-dai = <&codec>; }; }; clk_ext_audio_codec: clock-codec { compatible = "fixed-clock"; #clock-cells = <0>; clock-frequency = <12288000>; }; }; &i2c1 { codec: audio-codec@1a { compatible = "dlg,da7212"; reg = <0x1a>; #sound-dai-cells = <0>; VDDA-supply = <&reg_2v5_audio>; VDDSP-supply = <&reg_5v0_audio>; VDDMIC-supply = <&reg_3v3_audio>; VDDIO-supply = <&reg_3v3_audio>; clocks = <&clk_ext_audio_codec>; clock-names = "mclk"; }; }; --------------------------------------------------------------------- -- Sebastian Sebastian Reichel (5): ASoC: da7213: Add regulator support ASoC: Add DA7213 audio codec as selectable option ASoC: da7213: move set_sysclk to codec level ASoC: da7213: move set_pll to codec level ASoC: da7213: add default clock handling .../devicetree/bindings/sound/da7213.txt | 4 + sound/soc/codecs/Kconfig | 3 +- sound/soc/codecs/da7213.c | 128 ++++++++++++++++-- sound/soc/codecs/da7213.h | 3 + 4 files changed, 125 insertions(+), 13 deletions(-) -- 2.24.0.rc1

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[alsa-devel] [RESEND PATCH v5 4/6] ASoC: amd: add ACP3x TDM mode support
by Ravulapati Vishnu vardhan rao 13 Nov '19

13 Nov '19

ACP3x I2S (CPU DAI) can act in normal I2S and TDM modes. Added support for TDM mode. Desired mode can be selected from ASoC machine driver. Signed-off-by: Ravulapati Vishnu vardhan rao <Vishnuvardhanrao.Ravulapati(a)amd.com> --- sound/soc/amd/raven/acp3x-i2s.c | 51 +++++++++++++++++++++++++++++++++-------- sound/soc/amd/raven/acp3x.h | 2 ++ 2 files changed, 43 insertions(+), 10 deletions(-) diff --git a/sound/soc/amd/raven/acp3x-i2s.c b/sound/soc/amd/raven/acp3x-i2s.c index bd0ec68..f6b306d 100644 --- a/sound/soc/amd/raven/acp3x-i2s.c +++ b/sound/soc/amd/raven/acp3x-i2s.c @@ -43,7 +43,7 @@ static int acp3x_i2s_set_tdm_slot(struct snd_soc_dai *cpu_dai, u32 tx_mask, u32 rx_mask, int slots, int slot_width) { u32 val; - u16 slot_len; + u16 slot_len, flen; struct i2s_dev_data *adata = snd_soc_dai_get_drvdata(cpu_dai); @@ -66,16 +66,47 @@ static int acp3x_i2s_set_tdm_slot(struct snd_soc_dai *cpu_dai, u32 tx_mask, return -EINVAL; } - val = rv_readl(adata->acp3x_base + mmACP_BTTDM_ITER); - rv_writel((val | 0x2), adata->acp3x_base + mmACP_BTTDM_ITER); - val = rv_readl(adata->acp3x_base + mmACP_BTTDM_IRER); - rv_writel((val | 0x2), adata->acp3x_base + mmACP_BTTDM_IRER); + /* Enable I2S / BT channels TDM and respective + * I2S/BT`s TX/RX Formats frame lengths. + */ + flen = (FRM_LEN | (slots << 15) | (slot_len << 18)); - val = (FRM_LEN | (slots << 15) | (slot_len << 18)); - rv_writel(val, adata->acp3x_base + mmACP_BTTDM_TXFRMT); - rv_writel(val, adata->acp3x_base + mmACP_BTTDM_RXFRMT); - - adata->tdm_fmt = val; + if (adata->substream_type == SNDRV_PCM_STREAM_PLAYBACK) { + switch (adata->i2s_instance) { + case I2S_BT_INSTANCE: + val = rv_readl(adata->acp3x_base + mmACP_BTTDM_ITER); + rv_writel((val | 0x2), + adata->acp3x_base + mmACP_BTTDM_ITER); + rv_writel(flen, + adata->acp3x_base + mmACP_BTTDM_TXFRMT); + break; + case I2S_SP_INSTANCE: + default: + val = rv_readl(adata->acp3x_base + mmACP_I2STDM_ITER); + rv_writel((val | 0x2), + adata->acp3x_base + mmACP_I2STDM_ITER); + rv_writel(flen, + adata->acp3x_base + mmACP_I2STDM_TXFRMT); + } + } else { + switch (adata->i2s_instance) { + case I2S_BT_INSTANCE: + val = rv_readl(adata->acp3x_base + mmACP_BTTDM_IRER); + rv_writel((val | 0x2), + adata->acp3x_base + mmACP_BTTDM_IRER); + rv_writel(flen, + adata->acp3x_base + mmACP_BTTDM_RXFRMT); + break; + case I2S_SP_INSTANCE: + default: + val = rv_readl(adata->acp3x_base + mmACP_I2STDM_IRER); + rv_writel((val | 0x2), + adata->acp3x_base + mmACP_I2STDM_IRER); + rv_writel(flen, + adata->acp3x_base + mmACP_I2STDM_RXFRMT); + } + } + adata->tdm_fmt = flen; return 0; } diff --git a/sound/soc/amd/raven/acp3x.h b/sound/soc/amd/raven/acp3x.h index c071477..01b283a 100644 --- a/sound/soc/amd/raven/acp3x.h +++ b/sound/soc/amd/raven/acp3x.h @@ -76,6 +76,8 @@ struct i2s_dev_data { bool tdm_mode; unsigned int i2s_irq; u32 tdm_fmt; + u16 i2s_instance; + u32 substream_type; void __iomem *acp3x_base; struct snd_pcm_substream *play_stream; struct snd_pcm_substream *capture_stream; -- 2.7.4

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[alsa-devel] [PATCH] usb-audio: not submit urb for stopped endpoint
by Henry Lin 13 Nov '19

13 Nov '19

While output urb's snd_complete_urb() is executing, calling prepare_outbound_urb() may cause endpoint stopped before prepare_outbound_urb() returns and result in next urb submitted to stopped endpoint. usb-audio driver cannot re-use it afterwards as the urb is still hold by usb stack. This change checks EP_FLAG_RUNNING flag after prepare_outbound_urb() again to let snd_complete_urb() know the endpoint already stopped and does not submit next urb. We observed two scenario have this issue: 1. While executing snd_complete_urb() to complete an output urb, calling prepare_outbound_urb() let deactive_urbs() get called to unlink all active urbs. [ 268.097066] [<ffffffc000af7638>] deactivate_urbs+0xd4/0x108 [ 268.102633] [<ffffffc000af87fc>] snd_usb_endpoint_stop+0x30/0x58 [ 268.108636] [<ffffffc000b0272c>] snd_usb_substream_playback_trigger+0xa4/0xf4 [ 268.115765] [<ffffffc000acdbd0>] snd_pcm_do_stop+0x4c/0x58 [ 268.121245] [<ffffffc000acda24>] snd_pcm_action_single+0x40/0x88 [ 268.127245] [<ffffffc000ace984>] snd_pcm_action+0x30/0xf0 [ 268.132632] [<ffffffc000acea68>] snd_pcm_stop+0x24/0x2c [ 268.137851] [<ffffffc000ad5e14>] xrun+0x60/0x6c [ 268.142374] [<ffffffc000ad7a98>] snd_pcm_update_state+0xa8/0x10c [ 268.148374] [<ffffffc000ad7e24>] snd_pcm_update_hw_ptr0+0x328/0x344 [ 268.154635] [<ffffffc000ad7ed8>] snd_pcm_period_elapsed+0x98/0xb0 [ 268.160723] [<ffffffc000b02510>] prepare_playback_urb+0x46c/0x488 [ 268.166810] [<ffffffc000af7d60>] prepare_outbound_urb+0x60/0x1d4 [ 268.172805] [<ffffffc000af8d60>] snd_complete_urb+0x244/0x264 [ 268.178548] [<ffffffc00081fb38>] __usb_hcd_giveback_urb+0x94/0x104 [ 268.184721] [<ffffffc00081fbe4>] usb_hcd_giveback_urb+0x3c/0x114 [ 268.190724] [<ffffffc00084d4b4>] handle_tx_event+0x1304/0x1434 [ 268.196552] [<ffffffc00084dbc0>] xhci_handle_event+0x5dc/0x788 [ 268.202378] [<ffffffc00084dee4>] xhci_irq+0x178/0x280 2. Userspace application stops playback from sound subsystem with below call stack: [ 28.506477] CPU: 5 PID: 1274 Comm: AudioOut_25 Not tainted 4.4.38-tegra #31 [ 28.513430] Hardware name: quill (DT) [ 28.517085] Call trace: [ 28.519531] [<ffffffc000089a84>] dump_backtrace+0x0/0xf8 [ 28.524837] [<ffffffc000089c44>] show_stack+0x14/0x1c [ 28.529885] [<ffffffc000401c54>] dump_stack+0xac/0xe0 [ 28.534931] [<ffffffc000b35f94>] deactivate_urbs+0x148/0x180 [ 28.540578] [<ffffffc000b37160>] snd_usb_endpoint_stop+0x30/0x58 [ 28.546571] [<ffffffc000b410d8>] snd_usb_substream_playback_trigger+0xa4/0xf4 [ 28.553699] [<ffffffc000b0c160>] snd_pcm_do_stop+0x4c/0x58 [ 28.559179] [<ffffffc000b0bfb4>] snd_pcm_action_single+0x40/0x88 [ 28.565178] [<ffffffc000b0cf14>] snd_pcm_action+0x30/0xf0 [ 28.570568] [<ffffffc000b0fbc8>] snd_pcm_drop+0xac/0x140 [ 28.575873] [<ffffffc000b0fc84>] snd_pcm_release_substream+0x28/0xb0 [ 28.582212] [<ffffffc000b0fd48>] snd_pcm_release+0x3c/0x98 [ 28.587686] [<ffffffc0001e3210>] __fput+0xe0/0x1ac [ 28.592469] [<ffffffc0001e3334>] ____fput+0xc/0x14 [ 28.597253] [<ffffffc0000c2904>] task_work_run+0xa0/0xc0 [ 28.602558] [<ffffffc0000897bc>] do_notify_resume+0x48/0x60 [ 28.608123] [<ffffffc000084ee8>] work_pending+0x1c/0x20 In the call path, snd_pcm_stream spinlock has been acquired in snd_pcm_drop(). If an output urb is completed between the spinlock acquired and deactivate_urbs() clears EP_FLAG_RUNNING for the endpoint, its executing of snd_complete_urb() will be blocked for acquiring snd_pcm_stream spinlock in snd_pcm_period_elapsed() until the lock is released in snd_pcm_drop(). When snd_complete_urb() continues, all jobs for deactivate_urbs() are finished. Signed-off-by: Henry Lin <henryl(a)nvidia.com> --- sound/usb/endpoint.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c index a2ab8e8d3a93..4a9a2f6ef5a4 100644 --- a/sound/usb/endpoint.c +++ b/sound/usb/endpoint.c @@ -388,6 +388,9 @@ static void snd_complete_urb(struct urb *urb) } prepare_outbound_urb(ep, ctx); + /* can be stopped during prepare callback */ + if (unlikely(!test_bit(EP_FLAG_RUNNING, &ep->flags))) + goto exit_clear; } else { retire_inbound_urb(ep, ctx); /* can be stopped during retire callback */ -- 2.17.1

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[alsa-devel] [PATCH 0/9] ASoC: soc-core: cleanup step5
by Kuninori Morimoto 13 Nov '19

13 Nov '19

Hi Mark These are soc-core cleanup step5. My apology that it is randomly cleanupping. Kuninori Morimoto (9): ASoC: soc-core: remove soc_is_dai_link_bound() ASoC: soc-core: merge snd_soc_register_dai() and soc_add_dai() ASoC: soc-core: merge snd_soc_unregister_dai() and soc_del_dai() ASoC: soc-core: merge snd_soc_add_dai_link() and soc_bind_dai_link() ASoC: soc-core: merge snd_soc_remove_dai_link() and soc_unbind_dai_link() ASoC: soc-core: merge soc_remove_component() and soc_cleanup_component() ASoC: soc-core: merge snd_soc_bind_card() and snd_soc_instantiate_card() ASoC: soc-core: tidyup soc_probe_aux_devices() ASoC: soc-core: call snd_soc_dapm_shutdown() at soc_cleanup_card_resources() sound/soc/soc-core.c | 281 ++++++++++++++++++++------------------------------- 1 file changed, 107 insertions(+), 174 deletions(-) -- 2.7.4

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[alsa-devel] [PATCH] dt-bindings: vendor-prefixes: Add 'fsi[ab]' to the list of non-vendor prefixes
by Rob Herring 12 Nov '19

12 Nov '19

'fsia' and 'fsib' property prefixes were added as schema in commit 2f52475bac7e ("ASoC: fsi: switch to yaml base Documentation"). Unfortunately to do checks on actual vendor prefixes, we have to track the handful of prefixes which are not vendors like 'fsia' and 'fsib'. Fixes: 2f52475bac7e ASoC: fsi: switch to yaml base Documentation Cc: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Cc: Mark Brown <broonie(a)kernel.org> Signed-off-by: Rob Herring <robh(a)kernel.org> --- Seems to be an old binding, but there aren't any upstream dts files using these properties. If there's not backwards compatibility concerns, another option would be to rename these properties. Mark, probably should go via your tree as that is where the error is introduced. Rob Documentation/devicetree/bindings/vendor-prefixes.yaml | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/Documentation/devicetree/bindings/vendor-prefixes.yaml b/Documentation/devicetree/bindings/vendor-prefixes.yaml index 967e78c5ec0a..67b89db54aa2 100644 --- a/Documentation/devicetree/bindings/vendor-prefixes.yaml +++ b/Documentation/devicetree/bindings/vendor-prefixes.yaml @@ -16,7 +16,7 @@ properties: {} patternProperties: # Prefixes which are not vendors, but followed the pattern # DO NOT ADD NEW PROPERTIES TO THIS LIST - "^(at25|devbus|dmacap|dsa|exynos|gpio-fan|gpio|gpmc|hdmi|i2c-gpio),.*": true + "^(at25|devbus|dmacap|dsa|exynos|fsi[ab]|gpio-fan|gpio|gpmc|hdmi|i2c-gpio),.*": true "^(keypad|m25p|max8952|max8997|max8998|mpmc),.*": true "^(pinctrl-single|#pinctrl-single|PowerPC),.*": true "^(pl022|pxa-mmc|rcar_sound|rotary-encoder|s5m8767|sdhci),.*": true -- 2.20.1

2 2

[alsa-devel] [PATCH] ASoC: rt5682: cancel jack_detect_work if hs_jack is set to null
by Pierre-Louis Bossart 12 Nov '19

12 Nov '19

From: Bard liao <yung-chuan.liao(a)linux.intel.com> jack_detect_work will be triggered by rt5682_irq. We should cancel it if hs_jack is set to null. Signed-off-by: Bard liao <yung-chuan.liao(a)linux.intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/soc/codecs/rt5682.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/rt5682.c b/sound/soc/codecs/rt5682.c index 35dcec135c8a..0d8e9866c01a 100644 --- a/sound/soc/codecs/rt5682.c +++ b/sound/soc/codecs/rt5682.c @@ -1003,6 +1003,7 @@ static int rt5682_set_jack_detect(struct snd_soc_component *component, RT5682_JD1_EN_MASK, RT5682_JD1_DIS); regmap_update_bits(rt5682->regmap, RT5682_RC_CLK_CTRL, RT5682_POW_JDH | RT5682_POW_JDL, 0); + cancel_delayed_work_sync(&rt5682->jack_detect_work); return 0; } -- 2.20.1

2 1

[alsa-devel] [PATCH 0/4] ASoC: SOF: Intel: only use D0i3 if needed
by Pierre-Louis Bossart 12 Nov '19

12 Nov '19

The earlier patches for D0i3/S0ix support went one bridge too far, and forced the use of D0i3 even when it was not needed, which also had the side effect of breaking some of our s2idle suspend/resume stress tests. Add documentation on supported transitions and only use D0i3 if necessary. Keyon Jie (4): ASoC: SOF: PM: add state machine to comments ASoC: SOF: PM: add check before setting d0_substate ASoC: SOF: add helper to check if we should enter d0i3 suspend ASoC: SOF: PM: only suspend to D0I3 when needed sound/soc/sof/core.c | 13 +++++++++++++ sound/soc/sof/pm.c | 39 +++++++++++++++++++++++++++++++++++++-- sound/soc/sof/sof-priv.h | 2 ++ 3 files changed, 52 insertions(+), 2 deletions(-) -- 2.20.1

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[alsa-devel] [PATCH] ASoC: SOF: topology: free kcontrol memory on error
by Pierre-Louis Bossart 12 Nov '19

12 Nov '19

From: Dragos Tarcatu <dragos_tarcatu(a)mentor.com> The volume and bytes kcontrols are currently not freeing their memory on initialization failures. When an error occurs, all the widgets loaded so far are unloaded via sof_widget_unload(). But this only happens for the widgets that got successfully loaded. Fix that by kfree()-ing the allocated memory on load error. Fixes: 311ce4fe7637d ("ASoC: SOF: Add support for loading topologies") Reviewed-by: Paul Olaru <paul.olaru(a)nxp.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Signed-off-by: Dragos Tarcatu <dragos_tarcatu(a)mentor.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/soc/sof/topology.c | 67 +++++++++++++++++++++++++++++----------- 1 file changed, 49 insertions(+), 18 deletions(-) diff --git a/sound/soc/sof/topology.c b/sound/soc/sof/topology.c index 6096731e89ce..d82ab981e840 100644 --- a/sound/soc/sof/topology.c +++ b/sound/soc/sof/topology.c @@ -942,18 +942,22 @@ static int sof_control_load_volume(struct snd_soc_component *scomp, struct sof_ipc_ctrl_data *cdata; int tlv[TLV_ITEMS]; unsigned int i; - int ret; + int ret = 0; /* validate topology data */ - if (le32_to_cpu(mc->num_channels) > SND_SOC_TPLG_MAX_CHAN) - return -EINVAL; + if (le32_to_cpu(mc->num_channels) > SND_SOC_TPLG_MAX_CHAN) { + ret = -EINVAL; + goto out; + } /* init the volume get/put data */ scontrol->size = struct_size(scontrol->control_data, chanv, le32_to_cpu(mc->num_channels)); scontrol->control_data = kzalloc(scontrol->size, GFP_KERNEL); - if (!scontrol->control_data) - return -ENOMEM; + if (!scontrol->control_data) { + ret = -ENOMEM; + goto out; + } scontrol->comp_id = sdev->next_comp_id; scontrol->min_volume_step = le32_to_cpu(mc->min); @@ -963,7 +967,7 @@ static int sof_control_load_volume(struct snd_soc_component *scomp, /* set cmd for mixer control */ if (le32_to_cpu(mc->max) == 1) { scontrol->cmd = SOF_CTRL_CMD_SWITCH; - goto out; + goto skip; } scontrol->cmd = SOF_CTRL_CMD_VOLUME; @@ -971,14 +975,15 @@ static int sof_control_load_volume(struct snd_soc_component *scomp, /* extract tlv data */ if (get_tlv_data(kc->tlv.p, tlv) < 0) { dev_err(sdev->dev, "error: invalid TLV data\n"); - return -EINVAL; + ret = -EINVAL; + goto out_free; } /* set up volume table */ ret = set_up_volume_table(scontrol, tlv, le32_to_cpu(mc->max) + 1); if (ret < 0) { dev_err(sdev->dev, "error: setting up volume table\n"); - return ret; + goto out_free; } /* set default volume values to 0dB in control */ @@ -988,7 +993,7 @@ static int sof_control_load_volume(struct snd_soc_component *scomp, cdata->chanv[i].value = VOL_ZERO_DB; } -out: +skip: /* set up possible led control from mixer private data */ ret = sof_parse_tokens(scomp, &scontrol->led_ctl, led_tokens, ARRAY_SIZE(led_tokens), mc->priv.array, @@ -996,13 +1001,21 @@ static int sof_control_load_volume(struct snd_soc_component *scomp, if (ret != 0) { dev_err(sdev->dev, "error: parse led tokens failed %d\n", le32_to_cpu(mc->priv.size)); - return ret; + goto out_free_table; } dev_dbg(sdev->dev, "tplg: load kcontrol index %d chans %d\n", scontrol->comp_id, scontrol->num_channels); - return 0; + return ret; + +out_free_table: + if (le32_to_cpu(mc->max) > 1) + kfree(scontrol->volume_table); +out_free: + kfree(scontrol->control_data); +out: + return ret; } static int sof_control_load_enum(struct snd_soc_component *scomp, @@ -1047,6 +1060,7 @@ static int sof_control_load_bytes(struct snd_soc_component *scomp, container_of(hdr, struct snd_soc_tplg_bytes_control, hdr); struct soc_bytes_ext *sbe = (struct soc_bytes_ext *)kc->private_value; int max_size = sbe->max; + int ret = 0; /* init the get/put bytes data */ scontrol->size = sizeof(struct sof_ipc_ctrl_data) + @@ -1055,13 +1069,16 @@ static int sof_control_load_bytes(struct snd_soc_component *scomp, if (scontrol->size > max_size) { dev_err(sdev->dev, "err: bytes data size %d exceeds max %d.\n", scontrol->size, max_size); - return -EINVAL; + ret = -EINVAL; + goto out; } scontrol->control_data = kzalloc(max_size, GFP_KERNEL); cdata = scontrol->control_data; - if (!scontrol->control_data) - return -ENOMEM; + if (!scontrol->control_data) { + ret = -ENOMEM; + goto out; + } scontrol->comp_id = sdev->next_comp_id; scontrol->cmd = SOF_CTRL_CMD_BINARY; @@ -1076,23 +1093,32 @@ static int sof_control_load_bytes(struct snd_soc_component *scomp, if (cdata->data->magic != SOF_ABI_MAGIC) { dev_err(sdev->dev, "error: Wrong ABI magic 0x%08x.\n", cdata->data->magic); - return -EINVAL; + ret = -EINVAL; + goto out_free; } if (SOF_ABI_VERSION_INCOMPATIBLE(SOF_ABI_VERSION, cdata->data->abi)) { dev_err(sdev->dev, "error: Incompatible ABI version 0x%08x.\n", cdata->data->abi); - return -EINVAL; + ret = -EINVAL; + goto out_free; } if (cdata->data->size + sizeof(const struct sof_abi_hdr) != le32_to_cpu(control->priv.size)) { dev_err(sdev->dev, "error: Conflict in bytes vs. priv size.\n"); - return -EINVAL; + ret = -EINVAL; + goto out_free; } } - return 0; + + return ret; + +out_free: + kfree(scontrol->control_data); +out: + return ret; } /* external kcontrol init - used for any driver specific init */ @@ -1150,6 +1176,11 @@ static int sof_control_load(struct snd_soc_component *scomp, int index, return 0; } + if (ret < 0) { + kfree(scontrol); + return ret; + } + dobj->private = scontrol; list_add(&scontrol->list, &sdev->kcontrol_list); return ret; -- 2.20.1

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[alsa-devel] Several problems when trying to run alsa-info.sh
by Rolf Eike Beer 12 Nov '19

12 Nov '19

Greetings, I had the need to run alsa-info.sh --stdout from 1.1.9 on our embedded device. There were multiple things I noticed: -it requires /bin/bash, but seems to run ok with just /bin/sh. We don't have bash. -it requires pgrep, which we don't have, but replacing all calls with pidof seems to work well enough -it requires whereis, and does not use that -at least in stdout mode the absence of lspci should not be a fatal error Greetings Eike -- Rolf Eike Beer, emlix GmbH, http://www.emlix.com Fon +49 551 30664-0, Fax +49 551 30664-11 Gothaer Platz 3, 37083 Göttingen, Germany Sitz der Gesellschaft: Göttingen, Amtsgericht Göttingen HR B 3160 Geschäftsführung: Heike Jordan, Dr. Uwe Kracke – Ust-IdNr.: DE 205 198 055 emlix - smart embedded open source

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[alsa-devel] [PATCH] ALSA: au88x0: Fix incorrect device pointer for preallocation
by Takashi Iwai 12 Nov '19

12 Nov '19

The code change in commit 6974f8ad4494 ("ALSA: pci: Avoid non-standard macro usage") contained an incorrect conversion, which left the invalid pointer passed to the allocator for au88x0 driver. Fix it. Fixes: 6974f8ad4494 ("ALSA: pci: Avoid non-standard macro usage") Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/au88x0/au88x0_pcm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/au88x0/au88x0_pcm.c b/sound/pci/au88x0/au88x0_pcm.c index 1012e8da29e7..a2dcf43beedf 100644 --- a/sound/pci/au88x0/au88x0_pcm.c +++ b/sound/pci/au88x0/au88x0_pcm.c @@ -637,7 +637,7 @@ static int snd_vortex_new_pcm(vortex_t *chip, int idx, int nr) /* pre-allocation of Scatter-Gather buffers */ snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV_SG, - &chip->pci_dev, + &chip->pci_dev->dev, 0x10000, 0x10000); switch (VORTEX_PCM_TYPE(pcm)) { -- 2.16.4

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