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

[alsa-devel] [PATCH - maemo fixes 1/1] maemo plugin has two crashes I was able to see in a valgrind log from another user:
by stian.skjelstad＠gmail.com 05 May '19

05 May '19

From: Stian Skjelstad <stian.skjelstad(a)gmail.com> * maximum write size was calculated in words (16bit), but checked against byte-size length. This causes memcpy later to overflow the buffer (normally by up to 12KB). * remove a double free (by marking free'd data with NULL) * mmap returns MMAP_FAILED on error, not NULL I suspect that this plugin/driver might have other issues aswell, since I am unable to find any logic for checking DSP buffer status, and no implementation for odelay reporting. Author: Stian Skjelstad <stian.skjelstad(a)gmail.com> Signed-off-by: Stian Skjelstad <stian.skjelstad(a)gmail.com> diff --git a/maemo/alsa-dsp.c b/maemo/alsa-dsp.c index 7e04f6a..3307384 100644 --- a/maemo/alsa-dsp.c +++ b/maemo/alsa-dsp.c @@ -135,18 +135,18 @@ static snd_pcm_sframes_t alsa_dsp_transfer(snd_pcm_ioplug_t * io, snd_pcm_alsa_dsp_t *alsa_dsp = io->private_data; DENTER(); char *buf; - int words; + int bytes, words; ssize_t result; - words = size * alsa_dsp->bytes_per_frame; - words /= 2; - DPRINT("***** Info: words %d size %lu bpf: %d\n", words, size, - alsa_dsp->bytes_per_frame); - if (words > alsa_dsp->dsp_protocol->mmap_buffer_size) { - DERROR("Requested too much data transfer (playing only %d)\n", - alsa_dsp->dsp_protocol->mmap_buffer_size); - words = alsa_dsp->dsp_protocol->mmap_buffer_size; + bytes = size * alsa_dsp->bytes_per_frame; + DPRINT("***** Info: samples %lu * bpf %d => bytes %d\n", + size, alsa_dsp->bytes_per_frame, bytes); + if (bytes > alsa_dsp->dsp_protocol->mmap_buffer_size) { + DERROR("Requested too much data transfer (requested %d, playing only %d)\n", + bytes, alsa_dsp->dsp_protocol->mmap_buffer_size); + bytes = alsa_dsp->dsp_protocol->mmap_buffer_size; } + words = bytes / 2; if (alsa_dsp->dsp_protocol->state != STATE_PLAYING) { DPRINT("I did nothing - No start sent\n"); alsa_dsp_start(io); diff --git a/maemo/dsp-ctl.c b/maemo/dsp-ctl.c index 5bcda36..ac05942 100644 --- a/maemo/dsp-ctl.c +++ b/maemo/dsp-ctl.c @@ -93,6 +93,7 @@ static void dsp_ctl_close(snd_ctl_ext_t * ext) snd_ctl_dsp_t *dsp_ctl = ext->private_data; DENTER(); free(dsp_ctl->controls); + dsp_ctl->controls = NULL; free_control_list(&(dsp_ctl->playback_devices)); free_control_list(&(dsp_ctl->recording_devices)); // free(dsp_ctl); diff --git a/maemo/dsp-protocol.c b/maemo/dsp-protocol.c index af67251..32193d3 100644 --- a/maemo/dsp-protocol.c +++ b/maemo/dsp-protocol.c @@ -194,7 +194,7 @@ int dsp_protocol_open_node(dsp_protocol_t * dsp_protocol, const char *device) mmap((void *)0, dsp_protocol->mmap_buffer_size, PROT_READ | PROT_WRITE, MAP_SHARED, dsp_protocol->fd, 0); - if (dsp_protocol->mmap_buffer == NULL) { + if (dsp_protocol->mmap_buffer == MAP_FAILED) { report_dsp_protocol("Cannot mmap data buffer", dsp_protocol); ret = -ENOMEM; goto unlock; -- 2.17.1

2 1

Re: [alsa-devel] [PATCH V4] ASoC: fsl_esai: Add pm runtime function
by S.j. Wang 05 May '19

05 May '19

Hi > Hi Mark, > On Fri, May 03, 2019 at 01:27:31PM +0900, Mark Brown wrote: > > On Thu, May 02, 2019 at 09:13:58AM +0000, S.j. Wang wrote: > > > > > I am checking, but I don't know why this patch failed in your side. > > > I Tried to apply this patch on for-5.1, for 5.2, for-linus and > > > for-next, all are Successful. The git is > git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git. > > > > > I can't reproduce your problem. Is there any my operation wrong? > > > > The error message I got was: > > > > Applying: ASoC: fsl_esai: Add pm runtime function > > error: patch failed: sound/soc/fsl/fsl_esai.c:9 > > error: sound/soc/fsl/fsl_esai.c: patch does not apply Patch failed at > > 0001 ASoC: fsl_esai: Add pm runtime function > > > > which is the header addition. I can't spot any obvious issues > > visually looking at the patch, only thing I can think is some kind of > > whitespace damage somewhere. > > I downloaded this v4 from patchwork and resubmitted a v5 for a test. > Would you please try to apply that one? > > If my v5 works vs. having merge conflict at v4, maybe something wrong with > Git version of Shengjiu's? I compared my v5 and his > v4 using vimdiff, there is no much difference of whitespace. > > Thanks > Nicolin We find that maybe it is caused by the Transfer-Encoding format. We sent the patch by the --transfer-encoding=8bit, but in the receiver side it shows: Content-Type: text/plain; charset="utf-8" Content-Transfer-Encoding: base64 It may be caused by our company's mail server. We are checking... Best regards Wang shengjiu

1 0

[alsa-devel] [PATCH] ALSA: hda/hdmi - Don't report Jack event if no need to do that
by Hui Wang 04 May '19

04 May '19

On the machines with AMD GPU or Nvidia GPU, we often meet this issues: after s3, there are 4 HDMI/DP audio devices in the gnome-sound-setting even there is no any monitors plugged. When this problem happens, we check the /proc/asound/cardX/eld#N.M, we will find the monitor_present=1, eld_valid=0. The root cause is somehow the pin_sense reports the monitor is present and eld is valid when there is no monitor plugged. The current driver will read the eld data if the pin_sense reports the eld is valid, because of no monitor is plugged, there is no valid eld data, then the eld->valid is set to 0. If we don't let driver report Jack event when monitor_present=1 while eld_valid=0, there will be no this issue. After this change, the driver only reports Jack event with one of the below 2 conditons: eld->monitor_present=1 and eld->eld_valid=1 (a valid monitor detect) eld->monitor_present=0 (a monitor is unplugged) Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- sound/pci/hda/patch_hdmi.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 8b3ac690efa3..e5a34fa6f358 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -1548,7 +1548,7 @@ static bool hdmi_present_sense_via_verbs(struct hdmi_spec_per_pin *per_pin, else update_eld(codec, per_pin, eld); - ret = !repoll || !eld->monitor_present || eld->eld_valid; + ret = !eld->monitor_present || eld->eld_valid; jack = snd_hda_jack_tbl_get(codec, pin_nid); if (jack) -- 2.17.1

2 10

[alsa-devel] [PATCH V4] ASoC: fsl_esai: Add pm runtime function
by S.j. Wang 03 May '19

03 May '19

Add pm runtime support and move clock handling there. Close the clocks at suspend to reduce the power consumption. fsl_esai_suspend is replaced by pm_runtime_force_suspend. fsl_esai_resume is replaced by pm_runtime_force_resume. Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> Acked-by: Nicolin Chen <nicoleotsuka(a)gmail.com> --- Changes in v4 -resend base on for-5.2 Changes in v3 -refine the commit comments. -add acked-by Changes in v2 -refine the commit comments. -move regcache_mark_dirty to runtime suspend. sound/soc/fsl/fsl_esai.c | 141 ++++++++++++++++++++++++++--------------------- 1 file changed, 77 insertions(+), 64 deletions(-) diff --git a/sound/soc/fsl/fsl_esai.c b/sound/soc/fsl/fsl_esai.c index bad0dfed6b68..10d2210c91ef 100644 --- a/sound/soc/fsl/fsl_esai.c +++ b/sound/soc/fsl/fsl_esai.c @@ -9,6 +9,7 @@ #include <linux/module.h> #include <linux/of_irq.h> #include <linux/of_platform.h> +#include <linux/pm_runtime.h> #include <sound/dmaengine_pcm.h> #include <sound/pcm_params.h> @@ -466,30 +467,6 @@ static int fsl_esai_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { struct fsl_esai *esai_priv = snd_soc_dai_get_drvdata(dai); - int ret; - - /* - * Some platforms might use the same bit to gate all three or two of - * clocks, so keep all clocks open/close at the same time for safety - */ - ret = clk_prepare_enable(esai_priv->coreclk); - if (ret) - return ret; - if (!IS_ERR(esai_priv->spbaclk)) { - ret = clk_prepare_enable(esai_priv->spbaclk); - if (ret) - goto err_spbaclk; - } - if (!IS_ERR(esai_priv->extalclk)) { - ret = clk_prepare_enable(esai_priv->extalclk); - if (ret) - goto err_extalck; - } - if (!IS_ERR(esai_priv->fsysclk)) { - ret = clk_prepare_enable(esai_priv->fsysclk); - if (ret) - goto err_fsysclk; - } if (!dai->active) { /* Set synchronous mode */ @@ -506,16 +483,6 @@ static int fsl_esai_startup(struct snd_pcm_substream *substream, return 0; -err_fsysclk: - if (!IS_ERR(esai_priv->extalclk)) - clk_disable_unprepare(esai_priv->extalclk); -err_extalck: - if (!IS_ERR(esai_priv->spbaclk)) - clk_disable_unprepare(esai_priv->spbaclk); -err_spbaclk: - clk_disable_unprepare(esai_priv->coreclk); - - return ret; } static int fsl_esai_hw_params(struct snd_pcm_substream *substream, @@ -576,20 +543,6 @@ static int fsl_esai_hw_params(struct snd_pcm_substream *substream, return 0; } -static void fsl_esai_shutdown(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai) -{ - struct fsl_esai *esai_priv = snd_soc_dai_get_drvdata(dai); - - if (!IS_ERR(esai_priv->fsysclk)) - clk_disable_unprepare(esai_priv->fsysclk); - if (!IS_ERR(esai_priv->extalclk)) - clk_disable_unprepare(esai_priv->extalclk); - if (!IS_ERR(esai_priv->spbaclk)) - clk_disable_unprepare(esai_priv->spbaclk); - clk_disable_unprepare(esai_priv->coreclk); -} - static int fsl_esai_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *dai) { @@ -658,7 +611,6 @@ static int fsl_esai_trigger(struct snd_pcm_substream *substream, int cmd, static const struct snd_soc_dai_ops fsl_esai_dai_ops = { .startup = fsl_esai_startup, - .shutdown = fsl_esai_shutdown, .trigger = fsl_esai_trigger, .hw_params = fsl_esai_hw_params, .set_sysclk = fsl_esai_set_dai_sysclk, @@ -947,6 +899,10 @@ static int fsl_esai_probe(struct platform_device *pdev) return ret; } + pm_runtime_enable(&pdev->dev); + + regcache_cache_only(esai_priv->regmap, true); + ret = imx_pcm_dma_init(pdev, IMX_ESAI_DMABUF_SIZE); if (ret) dev_err(&pdev->dev, "failed to init imx pcm dma: %d\n", ret); @@ -954,6 +910,13 @@ static int fsl_esai_probe(struct platform_device *pdev) return ret; } +static int fsl_esai_remove(struct platform_device *pdev) +{ + pm_runtime_disable(&pdev->dev); + + return 0; +} + static const struct of_device_id fsl_esai_dt_ids[] = { { .compatible = "fsl,imx35-esai", }, { .compatible = "fsl,vf610-esai", }, @@ -961,22 +924,35 @@ static int fsl_esai_probe(struct platform_device *pdev) }; MODULE_DEVICE_TABLE(of, fsl_esai_dt_ids); -#ifdef CONFIG_PM_SLEEP -static int fsl_esai_suspend(struct device *dev) -{ - struct fsl_esai *esai = dev_get_drvdata(dev); - - regcache_cache_only(esai->regmap, true); - regcache_mark_dirty(esai->regmap); - - return 0; -} - -static int fsl_esai_resume(struct device *dev) +#ifdef CONFIG_PM +static int fsl_esai_runtime_resume(struct device *dev) { struct fsl_esai *esai = dev_get_drvdata(dev); int ret; + /* + * Some platforms might use the same bit to gate all three or two of + * clocks, so keep all clocks open/close at the same time for safety + */ + ret = clk_prepare_enable(esai->coreclk); + if (ret) + return ret; + if (!IS_ERR(esai->spbaclk)) { + ret = clk_prepare_enable(esai->spbaclk); + if (ret) + goto err_spbaclk; + } + if (!IS_ERR(esai->extalclk)) { + ret = clk_prepare_enable(esai->extalclk); + if (ret) + goto err_extalclk; + } + if (!IS_ERR(esai->fsysclk)) { + ret = clk_prepare_enable(esai->fsysclk); + if (ret) + goto err_fsysclk; + } + regcache_cache_only(esai->regmap, false); /* FIFO reset for safety */ @@ -987,22 +963,59 @@ static int fsl_esai_resume(struct device *dev) ret = regcache_sync(esai->regmap); if (ret) - return ret; + goto err_regcache_sync; /* FIFO reset done */ regmap_update_bits(esai->regmap, REG_ESAI_TFCR, ESAI_xFCR_xFR, 0); regmap_update_bits(esai->regmap, REG_ESAI_RFCR, ESAI_xFCR_xFR, 0); return 0; + +err_regcache_sync: + if (!IS_ERR(esai->fsysclk)) + clk_disable_unprepare(esai->fsysclk); +err_fsysclk: + if (!IS_ERR(esai->extalclk)) + clk_disable_unprepare(esai->extalclk); +err_extalclk: + if (!IS_ERR(esai->spbaclk)) + clk_disable_unprepare(esai->spbaclk); +err_spbaclk: + clk_disable_unprepare(esai->coreclk); + + return ret; } -#endif /* CONFIG_PM_SLEEP */ + +static int fsl_esai_runtime_suspend(struct device *dev) +{ + struct fsl_esai *esai = dev_get_drvdata(dev); + + regcache_cache_only(esai->regmap, true); + regcache_mark_dirty(esai->regmap); + + if (!IS_ERR(esai->fsysclk)) + clk_disable_unprepare(esai->fsysclk); + if (!IS_ERR(esai->extalclk)) + clk_disable_unprepare(esai->extalclk); + if (!IS_ERR(esai->spbaclk)) + clk_disable_unprepare(esai->spbaclk); + clk_disable_unprepare(esai->coreclk); + + return 0; +} +#endif /* CONFIG_PM */ static const struct dev_pm_ops fsl_esai_pm_ops = { - SET_SYSTEM_SLEEP_PM_OPS(fsl_esai_suspend, fsl_esai_resume) + SET_RUNTIME_PM_OPS(fsl_esai_runtime_suspend, + fsl_esai_runtime_resume, + NULL) + SET_SYSTEM_SLEEP_PM_OPS(pm_runtime_force_suspend, + pm_runtime_force_resume) }; static struct platform_driver fsl_esai_driver = { .probe = fsl_esai_probe, + .remove = fsl_esai_remove, .driver = { .name = "fsl-esai-dai", .pm = &fsl_esai_pm_ops, -- 1.9.1

3 5

[alsa-devel] [PATCH v2] ASoC: core: Allow topology to override machine driver FE DAI link config.
by Liam Girdwood 03 May '19

03 May '19

Machine drivers statically define a number of DAI links that currently cannot be changed or removed by topology. This means PCMs and platform components cannot be changed by topology at runtime AND machine drivers are tightly coupled to topology. This patch allows topology to override the machine driver DAI link config in order to reuse machine drivers with different topologies and platform components. The patch supports :- 1) create new FE PCMs with a topology defined PCM ID. 2) destroy existing static FE PCMs 3) change the platform component driver. 4) assign any new HW params fixups. 5) assign a new card name prefix to differentiate this topology to userspace. The patch requires no changes to the machine drivers, but does add some platform component flags that the platform component driver can assign before loading topologies. Signed-off-by: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> --- Changes since V1. o Now iterate over component_list to fix crash with DT enumerated cards. o Make sure name prefix is only added once with deferred probiing. include/sound/soc.h | 13 ++++++ sound/soc/soc-core.c | 101 +++++++++++++++++++++++++++++++++++++++++-- sound/soc/soc-pcm.c | 12 +++++ 3 files changed, 123 insertions(+), 3 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index f7579f82cc00..b1d65fcb8756 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -806,6 +806,14 @@ struct snd_soc_component_driver { unsigned int use_pmdown_time:1; /* care pmdown_time at stop */ unsigned int endianness:1; unsigned int non_legacy_dai_naming:1; + + /* this component uses topology and ignore machine driver FEs */ + const char *ignore_machine; + const char *topology_name_prefix; + int (*be_hw_params_fixup)(struct snd_soc_pcm_runtime *rtd, + struct snd_pcm_hw_params *params); + bool use_dai_pcm_id; /* use the DAI link PCM ID as PCM device number */ + int be_pcm_base; /* base device ID for all BE PCMs */ }; struct snd_soc_component { @@ -963,6 +971,9 @@ struct snd_soc_dai_link { /* pmdown_time is ignored at stop */ unsigned int ignore_pmdown_time:1; + /* Do not create a PCM for this DAI link (Backend link) */ + unsigned int ignore:1; + struct list_head list; /* DAI link list of the soc card */ struct snd_soc_dobj dobj; /* For topology */ }; @@ -1002,6 +1013,7 @@ struct snd_soc_card { const char *long_name; const char *driver_name; char dmi_longname[80]; + char topology_shortname[32]; struct device *dev; struct snd_card *snd_card; @@ -1011,6 +1023,7 @@ struct snd_soc_card { struct mutex dapm_mutex; bool instantiated; + bool topology_shortname_created; int (*probe)(struct snd_soc_card *card); int (*late_probe)(struct snd_soc_card *card); diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 4663de3cf495..38bf7d01894b 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -852,6 +852,9 @@ static int soc_bind_dai_link(struct snd_soc_card *card, const char *platform_name; int i; + if (dai_link->ignore) + return 0; + dev_dbg(card->dev, "ASoC: binding %s\n", dai_link->name); if (soc_is_dai_link_bound(card, dai_link)) { @@ -1461,7 +1464,9 @@ static int soc_probe_link_dais(struct snd_soc_card *card, { struct snd_soc_dai_link *dai_link = rtd->dai_link; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - int i, ret; + struct snd_soc_rtdcom_list *rtdcom; + struct snd_soc_component *component; + int i, ret, num; dev_dbg(card->dev, "ASoC: probe %s dai link %d late %d\n", card->name, rtd->num, order); @@ -1507,9 +1512,28 @@ static int soc_probe_link_dais(struct snd_soc_card *card, soc_dpcm_debugfs_add(rtd); #endif + num = rtd->num; + + /* + * most drivers will register their PCMs using DAI link ordering but + * topology based drivers can use the DAI link id field to set PCM + * device number and then use rtd + a base offset of the BEs. + */ + for_each_rtdcom(rtd, rtdcom) { + component = rtdcom->component; + + if (!component->driver->use_dai_pcm_id) + continue; + + if (rtd->dai_link->no_pcm) + num += component->driver->be_pcm_base; + else + num = rtd->dai_link->id; + } + if (cpu_dai->driver->compress_new) { /*create compress_device"*/ - ret = cpu_dai->driver->compress_new(rtd, rtd->num); + ret = cpu_dai->driver->compress_new(rtd, num); if (ret < 0) { dev_err(card->dev, "ASoC: can't create compress %s\n", dai_link->stream_name); @@ -1519,7 +1543,7 @@ static int soc_probe_link_dais(struct snd_soc_card *card, if (!dai_link->params) { /* create the pcm */ - ret = soc_new_pcm(rtd, rtd->num); + ret = soc_new_pcm(rtd, num); if (ret < 0) { dev_err(card->dev, "ASoC: can't create pcm %s :%d\n", dai_link->stream_name, ret); @@ -1846,6 +1870,74 @@ int snd_soc_set_dmi_name(struct snd_soc_card *card, const char *flavour) EXPORT_SYMBOL_GPL(snd_soc_set_dmi_name); #endif /* CONFIG_DMI */ +static void soc_check_tplg_fes(struct snd_soc_card *card) +{ + struct snd_soc_component *component; + const struct snd_soc_component_driver *comp_drv; + struct snd_soc_dai_link *dai_link; + int i; + + list_for_each_entry(component, &component_list, list) { + + /* does this component override FEs ? */ + if (!component->driver->ignore_machine) + continue; + + /* for this machine ? */ + if (strcmp(component->driver->ignore_machine, + card->dev->driver->name)) + continue; + + /* machine matches, so override the rtd data */ + for (i = 0; i < card->num_links; i++) { + + dai_link = &card->dai_link[i]; + + /* ignore this FE */ + if (dai_link->dynamic) { + dai_link->ignore = true; + continue; + } + + dev_info(card->dev, "info: override FE DAI link %s\n", + card->dai_link[i].name); + + /* override platform component */ + dai_link->platform_name = component->name; + + /* convert non BE into BE */ + dai_link->no_pcm = 1; + + /* override any BE fixups */ + dai_link->be_hw_params_fixup = + component->driver->be_hw_params_fixup; + + /* most BE links don't set stream name, so set it to + * dai link name if it's NULL to help bind widgets. + */ + if (!dai_link->stream_name) + dai_link->stream_name = dai_link->name; + } + + /* Inform userspace we are using alternate topology */ + if (component->driver->topology_name_prefix) { + + /* topology shortname created ? */ + if (!card->topology_shortname_created) { + comp_drv = component->driver; + + snprintf(card->topology_shortname, 32, "%s-%s", + comp_drv->topology_name_prefix, + card->name); + card->topology_shortname_created = true; + } + + /* use topology shortname */ + card->name = card->topology_shortname; + } + } +} + static int snd_soc_instantiate_card(struct snd_soc_card *card) { struct snd_soc_pcm_runtime *rtd; @@ -1855,6 +1947,9 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card) mutex_lock(&client_mutex); mutex_lock_nested(&card->mutex, SND_SOC_CARD_CLASS_INIT); + /* check whether any platform is ignore machine FE and using topology */ + soc_check_tplg_fes(card); + /* bind DAIs */ for (i = 0; i < card->num_links; i++) { ret = soc_bind_dai_link(card, &card->dai_link[i]); diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 153218aa4909..db51849ae9bd 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -865,8 +865,20 @@ int soc_dai_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { + struct snd_soc_pcm_runtime *rtd = substream->private_data; int ret; + /* perform any topology hw_params fixups before DAI */ + if (rtd->dai_link->be_hw_params_fixup) { + ret = rtd->dai_link->be_hw_params_fixup(rtd, params); + if (ret < 0) { + dev_err(rtd->dev, + "ASoC: hw_params topology fixup failed %d\n", + ret); + return ret; + } + } + if (dai->driver->ops->hw_params) { ret = dai->driver->ops->hw_params(substream, params, dai); if (ret < 0) { -- 2.17.1

4 4

[alsa-devel] [PATCH] Headset, microphone and internal speaker support for Tuxedo XC 1509
by Richard Sailer 03 May '19

03 May '19

This adds a SND_PCI_QUIRK(...) line for the Tuxedo XC 1509. The Tuxedo XC 1509 and the System76 oryp5 are the same barebone notebooks manufactured by Clevo. To name the fixups both use after the actual underlying hardware, this patch also changes System76_orpy5 to clevo_pb51ed in 2 enum symbols and one function name, matching the other pci_quirk entries which are also named after the device ODM. Signed-off-by: Richard Sailer <rs(a)tuxedocomputers.com> --- sound/pci/hda/patch_realtek.c | 19 ++++++++++--------- 1 file changed, 10 insertions(+), 9 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 715ab2342151..9df868ea96ee 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -1864,8 +1864,8 @@ enum { ALC887_FIXUP_BASS_CHMAP, ALC1220_FIXUP_GB_DUAL_CODECS, ALC1220_FIXUP_CLEVO_P950, - ALC1220_FIXUP_SYSTEM76_ORYP5, - ALC1220_FIXUP_SYSTEM76_ORYP5_PINS, + ALC1220_FIXUP_CLEVO_PB51ED, + ALC1220_FIXUP_CLEVO_PB51ED_PINS, }; static void alc889_fixup_coef(struct hda_codec *codec, @@ -2070,7 +2070,7 @@ static void alc1220_fixup_clevo_p950(struct hda_codec *codec, static void alc_fixup_headset_mode_no_hp_mic(struct hda_codec *codec, const struct hda_fixup *fix, int action); -static void alc1220_fixup_system76_oryp5(struct hda_codec *codec, +static void alc1220_fixup_clevo_pb51ed(struct hda_codec *codec, const struct hda_fixup *fix, int action) { @@ -2322,18 +2322,18 @@ static const struct hda_fixup alc882_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = alc1220_fixup_clevo_p950, }, - [ALC1220_FIXUP_SYSTEM76_ORYP5] = { + [ALC1220_FIXUP_CLEVO_PB51ED] = { .type = HDA_FIXUP_FUNC, - .v.func = alc1220_fixup_system76_oryp5, + .v.func = alc1220_fixup_clevo_pb51ed, }, - [ALC1220_FIXUP_SYSTEM76_ORYP5_PINS] = { + [ALC1220_FIXUP_CLEVO_PB51ED_PINS] = { .type = HDA_FIXUP_PINS, .v.pins = (const struct hda_pintbl[]) { { 0x19, 0x01a1913c }, /* use as headset mic, without its own jack detect */ {} }, .chained = true, - .chain_id = ALC1220_FIXUP_SYSTEM76_ORYP5, + .chain_id = ALC1220_FIXUP_CLEVO_PB51ED, }, }; @@ -2411,8 +2411,9 @@ static const struct snd_pci_quirk alc882_fixup_tbl[] = { SND_PCI_QUIRK(0x1558, 0x9501, "Clevo P950HR", ALC1220_FIXUP_CLEVO_P950), SND_PCI_QUIRK(0x1558, 0x95e1, "Clevo P95xER", ALC1220_FIXUP_CLEVO_P950), SND_PCI_QUIRK(0x1558, 0x95e2, "Clevo P950ER", ALC1220_FIXUP_CLEVO_P950), - SND_PCI_QUIRK(0x1558, 0x96e1, "System76 Oryx Pro (oryp5)", ALC1220_FIXUP_SYSTEM76_ORYP5_PINS), - SND_PCI_QUIRK(0x1558, 0x97e1, "System76 Oryx Pro (oryp5)", ALC1220_FIXUP_SYSTEM76_ORYP5_PINS), + SND_PCI_QUIRK(0x1558, 0x96e1, "System76 Oryx Pro (oryp5)", ALC1220_FIXUP_CLEVO_PB51ED_PINS), + SND_PCI_QUIRK(0x1558, 0x97e1, "System76 Oryx Pro (oryp5)", ALC1220_FIXUP_CLEVO_PB51ED_PINS), + SND_PCI_QUIRK(0x1558, 0x65d1, "Tuxedo Book XC1509", ALC1220_FIXUP_CLEVO_PB51ED_PINS), SND_PCI_QUIRK_VENDOR(0x1558, "Clevo laptop", ALC882_FIXUP_EAPD), SND_PCI_QUIRK(0x161f, 0x2054, "Medion laptop", ALC883_FIXUP_EAPD), SND_PCI_QUIRK(0x17aa, 0x3a0d, "Lenovo Y530", ALC882_FIXUP_LENOVO_Y530), -- 2.17.1

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[alsa-devel] [PATCH v2 0/5] Allwinner H6 SPDIF support
by Clément Péron 03 May '19

03 May '19

*H6 DMA support IS REQUIRED* Allwinner H6 SoC has a SPDIF controller called One Wire Audio (OWA) which is different from the previous H3 generation and not compatible. Difference are an increase of fifo sizes, some memory mapping are different and there is now the possibility to output the master clock on a pin. Also this series require the DMA working on H6, a first version has been submitted by Jernej Škrabec but is not yet accepted (as this moment): https://patchwork.kernel.org/project/linux-arm-kernel/list/?series=89011 This series has been tested on Beelink GS1 on top of sunxi/for-next with Jernej DMA support patch. Clément Péron (5): dt-bindings: sound: sun4i-spdif: Add Allwinner H6 compatible ASoC: sun4i-spdif: Add support for H6 SoC arm64: dts: allwinner: Add SPDIF node for Allwinner H6 arm64: dts: allwinner: h6: Enable SPDIF for Beelink GS1 arm64: defconfig: Enable Sun4i SPDIF module .../bindings/sound/sunxi,sun4i-spdif.txt | 3 +- .../dts/allwinner/sun50i-h6-beelink-gs1.dts | 4 ++ arch/arm64/boot/dts/allwinner/sun50i-h6.dtsi | 38 +++++++++++++++++ arch/arm64/configs/defconfig | 1 + sound/soc/sunxi/sun4i-spdif.c | 42 ++++++++++++++++--- 5 files changed, 81 insertions(+), 7 deletions(-) -- 2.17.1

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Re: [alsa-devel] [PATCH 00/19] ASoC: SOF: improvements for ABI checks and Intel code
by Pierre-Louis Bossart 03 May '19

03 May '19

On 5/3/19 12:40 AM, Mark Brown wrote: > On Tue, Apr 30, 2019 at 06:09:15PM -0500, Pierre-Louis Bossart wrote: >> This series is a set of relatively small SOF changes after the big >> batch merged last week (Thanks!). Since we are very close to the merge >> window and in May where most of the world takes time off, it'd be >> perfectly understandable if those patches were queued for 5.3, after >> feedback and corrections as needed. > > The headline thing I see when this pops up in my inbox is yet another 20 > patch series for these DSPs, I'm not going to notice the size of the > patches on a first pass. As I think you've noticed there's a lot of > reviewer fatigue setting in with this stuff, one thing that'd really > help here is if there were some help from Intel people with review for > the DPCM code. I can certainly understand reviewer fatigue, i've had to put a time limit on daily reviews to keep my sanity, but I don't get your last point. These patches were submitted and reviewed by Intel people on GitHub, what you see here is the result of multiple iterations precisely to make sure the patches are acceptable for upstream. we've set the goal of having two Intel aprovers for each patch. Can you elaborate on how we can make your life simpler? FYI we are going to have a number of fixes coming in the next 2 months for HDaudio support, changes to IPC/power flows to support D0ix states and SoundWire, 3rd party firmware module support, etc, so I really want to make sure you are comfortable with SOF-related patches, their level of maturity and the submission rate. Thanks! -Pierre

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[alsa-devel] [PATCH v2 0/2] soundwire: fix Kconfig select/depend issues
by Pierre-Louis Bossart 03 May '19

03 May '19

0-day/Kbuild starts complaining about missed module dependencies and compilation issues. Since codecs and soc drivers need to be compilable independently, let's fix this using the following model: SOF_INTEL_XYZ --- select ----> SOUNDWIRE_INTEL ---> select ---- | v SOUNDWIRE_BUS ^ | CODEC_ABC ------- select -----> REGMAP_SOUNDWIRE ---> select --- The SOUNDWIRE option remains a top-level boolean selector without impact on compilation, as recommended by Linus [1] and done for all Intel ASoC stuff. [1] https://lkml.org/lkml/2017/11/15/554 Changes since v1: Removed useless depend in REGMAP_SOUNDWIRE (select will ignore dependencies) Reordered patches to avoid circular dependencies with git bisect Pierre-Louis Bossart (2): soundwire: fix SOUNDWIRE_BUS option regmap: soundwire: fix Kconfig select/depend issue drivers/base/regmap/Kconfig | 2 +- drivers/soundwire/Kconfig | 1 - 2 files changed, 1 insertion(+), 2 deletions(-) -- 2.17.1

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[alsa-devel] [PATCH] MAINTAINERS: update git tree for sound entries
by Ross Zwisler 03 May '19

03 May '19

Several sound related entries in MAINTAINERS refer to the old git tree at "git://git.alsa-project.org/alsa-kernel.git". This is no longer used for development, and Takashi Iwai's kernel.org tree is used instead. Signed-off-by: Ross Zwisler <zwisler(a)google.com> --- MAINTAINERS | 13 ++++++------- 1 file changed, 6 insertions(+), 7 deletions(-) diff --git a/MAINTAINERS b/MAINTAINERS index e17ebf70b5480..d373d976a9317 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -3351,7 +3351,7 @@ F: include/uapi/linux/bsg.h BT87X AUDIO DRIVER M: Clemens Ladisch <clemens(a)ladisch.de> L: alsa-devel(a)alsa-project.org (moderated for non-subscribers) -T: git git://git.alsa-project.org/alsa-kernel.git +T: git git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git S: Maintained F: Documentation/sound/cards/bt87x.rst F: sound/pci/bt87x.c @@ -3404,7 +3404,7 @@ F: drivers/scsi/FlashPoint.* C-MEDIA CMI8788 DRIVER M: Clemens Ladisch <clemens(a)ladisch.de> L: alsa-devel(a)alsa-project.org (moderated for non-subscribers) -T: git git://git.alsa-project.org/alsa-kernel.git +T: git git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git S: Maintained F: sound/pci/oxygen/ @@ -5696,7 +5696,7 @@ F: drivers/edac/qcom_edac.c EDIROL UA-101/UA-1000 DRIVER M: Clemens Ladisch <clemens(a)ladisch.de> L: alsa-devel(a)alsa-project.org (moderated for non-subscribers) -T: git git://git.alsa-project.org/alsa-kernel.git +T: git git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git S: Maintained F: sound/usb/misc/ua101.c @@ -6036,7 +6036,7 @@ F: include/linux/f75375s.h FIREWIRE AUDIO DRIVERS M: Clemens Ladisch <clemens(a)ladisch.de> L: alsa-devel(a)alsa-project.org (moderated for non-subscribers) -T: git git://git.alsa-project.org/alsa-kernel.git +T: git git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git S: Maintained F: sound/firewire/ @@ -11593,7 +11593,7 @@ F: Documentation/devicetree/bindings/opp/ OPL4 DRIVER M: Clemens Ladisch <clemens(a)ladisch.de> L: alsa-devel(a)alsa-project.org (moderated for non-subscribers) -T: git git://git.alsa-project.org/alsa-kernel.git +T: git git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git S: Maintained F: sound/drivers/opl4/ @@ -14490,7 +14490,6 @@ M: Takashi Iwai <tiwai(a)suse.com> L: alsa-devel(a)alsa-project.org (moderated for non-subscribers) W: http://www.alsa-project.org/ T: git git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git -T: git git://git.alsa-project.org/alsa-kernel.git Q: http://patchwork.kernel.org/project/alsa-devel/list/ S: Maintained F: Documentation/sound/ @@ -16100,7 +16099,7 @@ F: drivers/usb/storage/ USB MIDI DRIVER M: Clemens Ladisch <clemens(a)ladisch.de> L: alsa-devel(a)alsa-project.org (moderated for non-subscribers) -T: git git://git.alsa-project.org/alsa-kernel.git +T: git git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git S: Maintained F: sound/usb/midi.* -- 2.21.0.593.g511ec345e18-goog

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