June 2018 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH v3 00/14] ARM: pxa: switch to DMA slave maps
by Robert Jarzmik 28 Jun '18

28 Jun '18

As I gathered almost all the required acks, this is an information only post before queuing to the PXA tree. The only missing part is for netdev, for which I'd like an ack from netdev people for patches 0007 and 0008, but that won't prevent me from queuing all the other patches (excepting the patch 0012 which can only be applied once 0007 and 0008 are queued). Cheers. -- Robert Robert Jarzmik (14): dmaengine: pxa: use a dma slave map ARM: pxa: add dma slave map dmaengine: pxa: add a default requestor policy mmc: pxamci: remove the dmaengine compat need media: pxa_camera: remove the dmaengine compat need mtd: rawnand: marvell: remove the dmaengine compat need net: smc911x: remove the dmaengine compat need net: smc91x: remove the dmaengine compat need ASoC: pxa: remove the dmaengine compat need ata: pata_pxa: remove the dmaengine compat need dmaengine: pxa: document pxad_param dmaengine: pxa: make the filter function internal ARM: pxa: remove the DMA IO resources ARM: pxa: change SSP DMA channels allocation arch/arm/mach-pxa/devices.c | 148 +--------------------------------- arch/arm/mach-pxa/devices.h | 6 +- arch/arm/mach-pxa/pxa25x.c | 38 ++++++++- arch/arm/mach-pxa/pxa27x.c | 39 ++++++++- arch/arm/mach-pxa/pxa3xx.c | 41 +++++++++- arch/arm/plat-pxa/ssp.c | 47 ----------- drivers/ata/pata_pxa.c | 10 +-- drivers/dma/pxa_dma.c | 18 ++++- drivers/media/platform/pxa_camera.c | 22 +---- drivers/mmc/host/pxamci.c | 29 +------ drivers/mtd/nand/raw/marvell_nand.c | 17 +--- drivers/net/ethernet/smsc/smc911x.c | 13 +-- drivers/net/ethernet/smsc/smc91x.c | 9 +-- drivers/net/ethernet/smsc/smc91x.h | 1 - include/linux/dma/pxa-dma.h | 20 +++-- include/linux/platform_data/mmp_dma.h | 4 + include/linux/pxa2xx_ssp.h | 2 - sound/arm/pxa2xx-ac97.c | 14 +--- sound/arm/pxa2xx-pcm-lib.c | 6 +- sound/soc/pxa/pxa-ssp.c | 5 +- sound/soc/pxa/pxa2xx-ac97.c | 32 ++------ sound/soc/pxa/pxa2xx-i2s.c | 6 +- 22 files changed, 176 insertions(+), 351 deletions(-) -- 2.11.0

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[alsa-devel] [PATCH v2] ASoC: dpcm: improve runtime update predictability
by Jerome Brunet 28 Jun '18

28 Jun '18

As it is, dpcm_runtime_update() performs the old path and new path update of a frontend before going on to the next frontend DAI. Depending the order of the FEs within the rtd list, the result of the update might be different. For example: * Frontend A connected to backend C, with a 48kHz playback * Frontend B connected to backend D, with a 44.1kHz playback * FE A appears before FE B in the rtd list of the card. If we reparent BE C to FE B (disconnecting BE D): * old path update of FE A will run first, and BE C will get hw_free() and shutdown() * new path update of FE B will run after and BE C, which is stopped, so it will be configured at 44.1kHz, as expected If we reparent BE D to FE A (disconnecting BE C): * new path update of FE A will run first but since BE D is still running at 44.1kHz, it won't be reconfigured (no call to startup() or hw_params()) * old path update of FE B runs after, nothing happens * In this case, we end up with a BE playing at 44.1kHz a stream which is supposed to be played at 48Khz (too slow) To improve this situation, this patch performs all the FE old paths update before going on to update the new paths. With this, the result should no longer depend on the order of the FE within the card rtd list. Please note that there might be a small performance penalty since dpcm_process_paths() is called twice per stream direction. Signed-off-by: Jerome Brunet <jbrunet(a)baylibre.com> --- Changes since v1 [0]: * Add missing static as reported by kbuild robot. [0]: https://lkml.kernel.org/r/20180625082923.25636-1-jbrunet@baylibre.com sound/soc/soc-pcm.c | 165 +++++++++++++++++++++++++++------------------------- 1 file changed, 86 insertions(+), 79 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 5e7ae47a9658..4eda18cfe895 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -2543,106 +2543,113 @@ static int dpcm_run_old_update(struct snd_soc_pcm_runtime *fe, int stream) return ret; } -/* Called by DAPM mixer/mux changes to update audio routing between PCMs and - * any DAI links. - */ -int soc_dpcm_runtime_update(struct snd_soc_card *card) +static int soc_dpcm_fe_runtime_update(struct snd_soc_pcm_runtime *fe, int new) { - struct snd_soc_pcm_runtime *fe; - int old, new, paths; + struct snd_soc_dapm_widget_list *list; + int count, paths; - mutex_lock_nested(&card->mutex, SND_SOC_CARD_CLASS_RUNTIME); - list_for_each_entry(fe, &card->rtd_list, list) { - struct snd_soc_dapm_widget_list *list; + if (!fe->dai_link->dynamic) + return 0; - /* make sure link is FE */ - if (!fe->dai_link->dynamic) - continue; + /* only check active links */ + if (!fe->cpu_dai->active) + return 0; - /* only check active links */ - if (!fe->cpu_dai->active) - continue; + /* DAPM sync will call this to update DSP paths */ + dev_dbg(fe->dev, "ASoC: DPCM %s runtime update for FE %s\n", + new ? "new" : "old", fe->dai_link->name); - /* DAPM sync will call this to update DSP paths */ - dev_dbg(fe->dev, "ASoC: DPCM runtime update for FE %s\n", - fe->dai_link->name); + /* skip if FE doesn't have playback capability */ + if (!fe->cpu_dai->driver->playback.channels_min || + !fe->codec_dai->driver->playback.channels_min) + goto capture; - /* skip if FE doesn't have playback capability */ - if (!fe->cpu_dai->driver->playback.channels_min - || !fe->codec_dai->driver->playback.channels_min) - goto capture; - - /* skip if FE isn't currently playing */ - if (!fe->cpu_dai->playback_active - || !fe->codec_dai->playback_active) - goto capture; - - paths = dpcm_path_get(fe, SNDRV_PCM_STREAM_PLAYBACK, &list); - if (paths < 0) { - dev_warn(fe->dev, "ASoC: %s no valid %s path\n", - fe->dai_link->name, "playback"); - mutex_unlock(&card->mutex); - return paths; - } + /* skip if FE isn't currently playing */ + if (!fe->cpu_dai->playback_active || !fe->codec_dai->playback_active) + goto capture; - /* update any new playback paths */ - new = dpcm_process_paths(fe, SNDRV_PCM_STREAM_PLAYBACK, &list, 1); - if (new) { - dpcm_run_new_update(fe, SNDRV_PCM_STREAM_PLAYBACK); - dpcm_clear_pending_state(fe, SNDRV_PCM_STREAM_PLAYBACK); - dpcm_be_disconnect(fe, SNDRV_PCM_STREAM_PLAYBACK); - } + paths = dpcm_path_get(fe, SNDRV_PCM_STREAM_PLAYBACK, &list); + if (paths < 0) { + dev_warn(fe->dev, "ASoC: %s no valid %s path\n", + fe->dai_link->name, "playback"); + return paths; + } - /* update any old playback paths */ - old = dpcm_process_paths(fe, SNDRV_PCM_STREAM_PLAYBACK, &list, 0); - if (old) { + /* update any playback paths */ + count = dpcm_process_paths(fe, SNDRV_PCM_STREAM_PLAYBACK, &list, new); + if (count) { + if (new) + dpcm_run_new_update(fe, SNDRV_PCM_STREAM_PLAYBACK); + else dpcm_run_old_update(fe, SNDRV_PCM_STREAM_PLAYBACK); - dpcm_clear_pending_state(fe, SNDRV_PCM_STREAM_PLAYBACK); - dpcm_be_disconnect(fe, SNDRV_PCM_STREAM_PLAYBACK); - } - dpcm_path_put(&list); + dpcm_clear_pending_state(fe, SNDRV_PCM_STREAM_PLAYBACK); + dpcm_be_disconnect(fe, SNDRV_PCM_STREAM_PLAYBACK); + } + + dpcm_path_put(&list); + capture: - /* skip if FE doesn't have capture capability */ - if (!fe->cpu_dai->driver->capture.channels_min - || !fe->codec_dai->driver->capture.channels_min) - continue; + /* skip if FE doesn't have capture capability */ + if (!fe->cpu_dai->driver->capture.channels_min || + !fe->codec_dai->driver->capture.channels_min) + return 0; - /* skip if FE isn't currently capturing */ - if (!fe->cpu_dai->capture_active - || !fe->codec_dai->capture_active) - continue; + /* skip if FE isn't currently capturing */ + if (!fe->cpu_dai->capture_active || !fe->codec_dai->capture_active) + return 0; - paths = dpcm_path_get(fe, SNDRV_PCM_STREAM_CAPTURE, &list); - if (paths < 0) { - dev_warn(fe->dev, "ASoC: %s no valid %s path\n", - fe->dai_link->name, "capture"); - mutex_unlock(&card->mutex); - return paths; - } + paths = dpcm_path_get(fe, SNDRV_PCM_STREAM_CAPTURE, &list); + if (paths < 0) { + dev_warn(fe->dev, "ASoC: %s no valid %s path\n", + fe->dai_link->name, "capture"); + return paths; + } - /* update any new capture paths */ - new = dpcm_process_paths(fe, SNDRV_PCM_STREAM_CAPTURE, &list, 1); - if (new) { + /* update any old capture paths */ + count = dpcm_process_paths(fe, SNDRV_PCM_STREAM_CAPTURE, &list, new); + if (count) { + if (new) dpcm_run_new_update(fe, SNDRV_PCM_STREAM_CAPTURE); - dpcm_clear_pending_state(fe, SNDRV_PCM_STREAM_CAPTURE); - dpcm_be_disconnect(fe, SNDRV_PCM_STREAM_CAPTURE); - } - - /* update any old capture paths */ - old = dpcm_process_paths(fe, SNDRV_PCM_STREAM_CAPTURE, &list, 0); - if (old) { + else dpcm_run_old_update(fe, SNDRV_PCM_STREAM_CAPTURE); - dpcm_clear_pending_state(fe, SNDRV_PCM_STREAM_CAPTURE); - dpcm_be_disconnect(fe, SNDRV_PCM_STREAM_CAPTURE); - } - dpcm_path_put(&list); + dpcm_clear_pending_state(fe, SNDRV_PCM_STREAM_CAPTURE); + dpcm_be_disconnect(fe, SNDRV_PCM_STREAM_CAPTURE); } - mutex_unlock(&card->mutex); + dpcm_path_put(&list); + return 0; } + +/* Called by DAPM mixer/mux changes to update audio routing between PCMs and + * any DAI links. + */ +int soc_dpcm_runtime_update(struct snd_soc_card *card) +{ + struct snd_soc_pcm_runtime *fe; + int ret = 0; + + mutex_lock_nested(&card->mutex, SND_SOC_CARD_CLASS_RUNTIME); + /* shutdown all old paths first */ + list_for_each_entry(fe, &card->rtd_list, list) { + ret = soc_dpcm_fe_runtime_update(fe, 0); + if (ret) + goto out; + } + + /* bring new paths up */ + list_for_each_entry(fe, &card->rtd_list, list) { + ret = soc_dpcm_fe_runtime_update(fe, 1); + if (ret) + goto out; + } + +out: + mutex_unlock(&card->mutex); + return ret; +} int soc_dpcm_be_digital_mute(struct snd_soc_pcm_runtime *fe, int mute) { struct snd_soc_dpcm *dpcm; -- 2.14.4

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[alsa-devel] [PATCH] ASoC: qdsp6: qdafe: fix some off by one bugs
by Dan Carpenter 28 Jun '18

28 Jun '18

The > should be >= or we could read one element beyond the end of the port_maps[] array. Fixes: 7fa2d70f9766 ("ASoC: qdsp6: q6afe: Add q6afe driver") Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/soc/qcom/qdsp6/q6afe.c b/sound/soc/qcom/qdsp6/q6afe.c index 01f43218984b..69a7896cb713 100644 --- a/sound/soc/qcom/qdsp6/q6afe.c +++ b/sound/soc/qcom/qdsp6/q6afe.c @@ -777,7 +777,7 @@ static int q6afe_callback(struct apr_device *adev, struct apr_resp_pkt *data) */ int q6afe_get_port_id(int index) { - if (index < 0 || index > AFE_PORT_MAX) + if (index < 0 || index >= AFE_PORT_MAX) return -EINVAL; return port_maps[index].port_id; @@ -1014,7 +1014,7 @@ int q6afe_port_stop(struct q6afe_port *port) port_id = port->id; index = port->token; - if (index < 0 || index > AFE_PORT_MAX) { + if (index < 0 || index >= AFE_PORT_MAX) { dev_err(afe->dev, "AFE port index[%d] invalid!\n", index); return -EINVAL; } @@ -1355,7 +1355,7 @@ struct q6afe_port *q6afe_port_get_from_id(struct device *dev, int id) unsigned long flags; int cfg_type; - if (id < 0 || id > AFE_PORT_MAX) { + if (id < 0 || id >= AFE_PORT_MAX) { dev_err(dev, "AFE port token[%d] invalid!\n", id); return ERR_PTR(-EINVAL); }

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[alsa-devel] [PATCH] ASoC: dpcm: don't merge format from invalid codec dai
by Jerome Brunet 28 Jun '18

28 Jun '18

When merging codec formats, dpcm_runtime_base_format() should skip the codecs which are not supporting the current stream direction. At the moment, if a BE link has more than one codec, and only one of these codecs has no capture DAI, it becomes impossible to start a capture stream because the merged format would be 0. Skipping invalid codec DAI solves the problem. Fixes: b073ed4e2126 ("ASoC: soc-pcm: DPCM cares BE format") Signed-off-by: Jerome Brunet <jbrunet(a)baylibre.com> --- sound/soc/soc-pcm.c | 8 ++++++++ 1 file changed, 8 insertions(+) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 5141409ccaed..a58d8e7bf52b 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -1694,6 +1694,14 @@ static u64 dpcm_runtime_base_format(struct snd_pcm_substream *substream) int i; for (i = 0; i < be->num_codecs; i++) { + /* + * Skip CODECs which don't support the current stream + * type. See soc_pcm_init_runtime_hw() for more details + */ + if (!snd_soc_dai_stream_valid(be->codec_dais[i], + stream)) + continue; + codec_dai_drv = be->codec_dais[i]->driver; if (stream == SNDRV_PCM_STREAM_PLAYBACK) codec_stream = &codec_dai_drv->playback; -- 2.14.4

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[alsa-devel] [PATCH 0/3] ALSA: PM-related fixes/cleanups
by Takashi Iwai 28 Jun '18

28 Jun '18

Hi, this is a patch set addressing some minor issues found in PM-related stuff, as well as the Intel HDMI fallback issue. The first patch covers the unusual error paths from runtime PM, the second one is a code refactoring and the last one fixes the superfluous (rather harmful) binding with the generic HDMI driver as fallback of i915 component binding. Takashi === Takashi Iwai (3): ALSA: hda - Handle error from snd_hda_power_up*() ALSA: hda - Move in_pm accessors to HDA core ALSA: hda/hdmi - Don't fall back to generic when i915 binding fails include/sound/hdaudio.h | 27 +++++++++++++++++ sound/pci/hda/hda_beep.c | 3 +- sound/pci/hda/hda_codec.c | 25 ++++++---------- sound/pci/hda/hda_controller.c | 4 ++- sound/pci/hda/hda_proc.c | 3 +- sound/pci/hda/hda_sysfs.c | 4 ++- sound/pci/hda/patch_ca0132.c | 53 +++++++++++++++++++++++++++------- sound/pci/hda/patch_hdmi.c | 4 ++- sound/pci/hda/patch_realtek.c | 3 +- 9 files changed, 93 insertions(+), 33 deletions(-) -- 2.17.1

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[alsa-devel] [PATCH] ALSA: hda: Add AZX_DCAPS_PM_RUNTIME for AMD Raven Ridge
by Kai-Heng Feng 28 Jun '18

28 Jun '18

This patch can make audio controller in AMD Raven Ridge gets runtime suspended to D3, to save ~1W power when it's not in use. Cc: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> Signed-off-by: Kai-Heng Feng <kai.heng.feng(a)canonical.com> --- sound/pci/hda/hda_intel.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 1ae1850b3bfd..a9b55d65f2bd 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2535,7 +2535,8 @@ static const struct pci_device_id azx_ids[] = { .driver_data = AZX_DRIVER_GENERIC | AZX_DCAPS_PRESET_ATI_SB }, /* AMD Raven */ { PCI_DEVICE(0x1022, 0x15e3), - .driver_data = AZX_DRIVER_GENERIC | AZX_DCAPS_PRESET_ATI_SB }, + .driver_data = AZX_DRIVER_GENERIC | AZX_DCAPS_PRESET_ATI_SB | + AZX_DCAPS_PM_RUNTIME }, /* ATI HDMI */ { PCI_DEVICE(0x1002, 0x0002), .driver_data = AZX_DRIVER_ATIHDMI_NS | AZX_DCAPS_PRESET_ATI_HDMI_NS }, -- 2.17.1

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[alsa-devel] dsd formats in ALSA plugins
by slava 28 Jun '18

28 Jun '18

Sorry, don't receive my own post, so can't reply Find in alsa-lib-1.1.6 file src/pcm/pcm-params.c line 1945 static const snd_mask_t refine_masks[SND_PCM_HW_PARAM_LAST_MASK - SND_PCM_HW_PARAM_FIRST_MASK + 1] = { [SND_PCM_HW_PARAM_ACCESS - SND_PCM_HW_PARAM_FIRST_MASK] = { .bits = { 0x1f }, }, [SND_PCM_HW_PARAM_FORMAT - SND_PCM_HW_PARAM_FIRST_MASK] = { .bits = { 0x81ffffff, 0xfff}, Change last 0xfff to 0xfffff and now can use DSD formats How i understand, in version 1.0.27 last format = 43 and when new format was added but mask not changed May be i absolutely wrong and what i do has side effects or needed any else? Thanks

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[alsa-devel] [PATCH] Fixed typo on line 733.
by Guy Chronister 28 Jun '18

28 Jun '18

Signed-off-by: Guy Chronister <guylovesbritt(a)gmail.com> --- sound/pci/oxygen/oxygen_mixer.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/oxygen/oxygen_mixer.c b/sound/pci/oxygen/oxygen_mixer.c index 81af21a..ba87213 100644 --- a/sound/pci/oxygen/oxygen_mixer.c +++ b/sound/pci/oxygen/oxygen_mixer.c @@ -730,7 +730,7 @@ static int ac97_fp_rec_volume_put(struct snd_kcontrol *ctl, oldreg = oxygen_read_ac97(chip, 1, AC97_REC_GAIN); newreg = oldreg & ~0x0707; newreg = newreg | (value->value.integer.value[0] & 7); - newreg = newreg | ((value->value.integer.value[0] & 7) << 8); + newreg = newreg | ((value->value.integer.value[1] & 7) << 8); change = newreg != oldreg; if (change) oxygen_write_ac97(chip, 1, AC97_REC_GAIN, newreg); -- 2.7.4

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[alsa-devel] [PATCH v2] ALSA: hda/hdmi - Don't fall back to generic when i915 binding fails
by Takashi Iwai 28 Jun '18

28 Jun '18

When i915 component binding fails, it means that HDMI isn't applicable anyway. Although the probe with the generic HDMI parser would still work, it's essentially useless, hence better to be left unbound. This patch mimics the probe_id field at failing the i915 component binding so that the generic HDMI won't be bound after that. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- v1->v2: use the dummy probe_id value to assure failing the further binding sound/pci/hda/hda_codec.h | 1 + sound/pci/hda/patch_hdmi.c | 2 ++ 2 files changed, 3 insertions(+) diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index a8b1b31f161c..53af948dacea 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -84,6 +84,7 @@ struct hda_bus { */ typedef int (*hda_codec_patch_t)(struct hda_codec *); +#define HDA_CODEC_ID_SKIP_PROBE 0x00000001 #define HDA_CODEC_ID_GENERIC_HDMI 0x00000101 #define HDA_CODEC_ID_GENERIC 0x00000201 diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index c182e619ad58..31f092abf299 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -2542,6 +2542,8 @@ static int alloc_intel_hdmi(struct hda_codec *codec) /* requires i915 binding */ if (!codec->bus->core.audio_component) { codec_info(codec, "No i915 binding for Intel HDMI/DP codec\n"); + /* set probe_id here to prevent generic fallback binding */ + codec->probe_id = HDA_CODEC_ID_SKIP_PROBE; return -ENODEV; } -- 2.16.4

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[alsa-devel] [PATCH] ALSA: hda - Handle pm failure during hotplug
by Chris Wilson 27 Jun '18

27 Jun '18

Obtaining the runtime pm wakeref can fail, especially in a hotplug scenario where i915.ko has been unloaded. If we do not catch the failure, we end up with an unbalanced pm. References: 222bde03881c ("ALSA: hda - Fix mutex deadlock at HDMI/DP hotplug") Signed-off-by: Chris Wilson <chris(a)chris-wilson.co.uk> Cc: Ville Syrjälä <ville.syrjala(a)linux.intel.com> Cc: Takashi Iwai <tiwai(a)suse.de> Cc: Vinod Koul <vinod.koul(a)intel.com> --- sound/pci/hda/patch_hdmi.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 7d7eb1354eee..3ca98c9fcaee 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -1632,8 +1632,8 @@ static bool hdmi_present_sense(struct hdmi_spec_per_pin *per_pin, int repoll) int ret; /* no temporary power up/down needed for component notifier */ - if (!codec_has_acomp(codec)) - snd_hda_power_up_pm(codec); + if (!codec_has_acomp(codec) && snd_hda_power_up_pm(codec)) + return false; mutex_lock(&spec->pcm_lock); if (codec_has_acomp(codec)) { -- 2.18.0

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