April 2020 - Alsa-devel - mailman.alsa-project.org

[PATCH 1/2] ALSA: usb-audio: Improve frames size computation
by Alexander Tsoy 25 Apr '20

25 Apr '20

For computation of the the next frame size current value of fs/fps and accumulated fractional parts of fs/fps are used, where values are stored in Q16.16 format. This is quite natural for computing frame size for asynchronous endpoints driven by explicit feedback, since in this case fs/fps is a value provided by the feedback endpoint and it's already in the Q format. If an error is accumulated over time, the device can adjust fs/fps value to prevent buffer overruns/underruns. But for synchronous endpoints the accuracy provided by these computations is not enough. Due to accumulated error the driver periodically produces frames with incorrect size (+/- 1 audio sample). This patch fixes this issue by implementing a different algorithm for frame size computation. It is based on accumulating of the remainders from division fs/fps and it doesn't accumulate errors over time. This new method is enabled for synchronous and adaptive playback endpoints. Signed-off-by: Alexander Tsoy <alexander(a)tsoy.me> --- sound/usb/card.h | 4 ++++ sound/usb/endpoint.c | 43 ++++++++++++++++++++++++++++++++++++++----- sound/usb/endpoint.h | 1 + sound/usb/pcm.c | 2 ++ 4 files changed, 45 insertions(+), 5 deletions(-) diff --git a/sound/usb/card.h b/sound/usb/card.h index 395403a2d33f..820e564656ed 100644 --- a/sound/usb/card.h +++ b/sound/usb/card.h @@ -84,6 +84,10 @@ struct snd_usb_endpoint { dma_addr_t sync_dma; /* DMA address of syncbuf */ unsigned int pipe; /* the data i/o pipe */ + unsigned int framesize[2]; /* small/large frame sizes in samples */ + unsigned int sample_rem; /* remainder from division fs/fps */ + unsigned int sample_accum; /* sample accumulator */ + unsigned int fps; /* frames per second */ unsigned int freqn; /* nominal sampling rate in fs/fps in Q16.16 format */ unsigned int freqm; /* momentary sampling rate in fs/fps in Q16.16 format */ int freqshift; /* how much to shift the feedback value to get Q16.16 */ diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c index 4a9a2f6ef5a4..d8dc7cb56d43 100644 --- a/sound/usb/endpoint.c +++ b/sound/usb/endpoint.c @@ -124,12 +124,12 @@ int snd_usb_endpoint_implicit_feedback_sink(struct snd_usb_endpoint *ep) /* * For streaming based on information derived from sync endpoints, - * prepare_outbound_urb_sizes() will call next_packet_size() to + * prepare_outbound_urb_sizes() will call slave_next_packet_size() to * determine the number of samples to be sent in the next packet. * - * For implicit feedback, next_packet_size() is unused. + * For implicit feedback, slave_next_packet_size() is unused. */ -int snd_usb_endpoint_next_packet_size(struct snd_usb_endpoint *ep) +int snd_usb_endpoint_slave_next_packet_size(struct snd_usb_endpoint *ep) { unsigned long flags; int ret; @@ -146,6 +146,29 @@ int snd_usb_endpoint_next_packet_size(struct snd_usb_endpoint *ep) return ret; } +/* + * For adaptive and synchronous endpoints, prepare_outbound_urb_sizes() + * will call next_packet_size() to determine the number of samples to be + * sent in the next packet. + */ +int snd_usb_endpoint_next_packet_size(struct snd_usb_endpoint *ep) +{ + int ret; + + if (ep->fill_max) + return ep->maxframesize; + + ep->sample_accum += ep->sample_rem; + if (ep->sample_accum >= ep->fps) { + ep->sample_accum -= ep->fps; + ret = ep->framesize[1]; + } else { + ret = ep->framesize[0]; + } + + return ret; +} + static void retire_outbound_urb(struct snd_usb_endpoint *ep, struct snd_urb_ctx *urb_ctx) { @@ -190,6 +213,8 @@ static void prepare_silent_urb(struct snd_usb_endpoint *ep, if (ctx->packet_size[i]) counts = ctx->packet_size[i]; + else if (ep->sync_master) + counts = snd_usb_endpoint_slave_next_packet_size(ep); else counts = snd_usb_endpoint_next_packet_size(ep); @@ -874,10 +899,17 @@ int snd_usb_endpoint_set_params(struct snd_usb_endpoint *ep, ep->maxpacksize = fmt->maxpacksize; ep->fill_max = !!(fmt->attributes & UAC_EP_CS_ATTR_FILL_MAX); - if (snd_usb_get_speed(ep->chip->dev) == USB_SPEED_FULL) + if (snd_usb_get_speed(ep->chip->dev) == USB_SPEED_FULL) { ep->freqn = get_usb_full_speed_rate(rate); - else + ep->fps = 1000; + } else { ep->freqn = get_usb_high_speed_rate(rate); + ep->fps = 8000; + } + + ep->sample_rem = rate % ep->fps; + ep->framesize[0] = rate / ep->fps; + ep->framesize[1] = (rate + (ep->fps - 1)) / ep->fps; /* calculate the frequency in 16.16 format */ ep->freqm = ep->freqn; @@ -936,6 +968,7 @@ int snd_usb_endpoint_start(struct snd_usb_endpoint *ep) ep->active_mask = 0; ep->unlink_mask = 0; ep->phase = 0; + ep->sample_accum = 0; snd_usb_endpoint_start_quirk(ep); diff --git a/sound/usb/endpoint.h b/sound/usb/endpoint.h index 63a39d4fa8d8..d23fa0a8c11b 100644 --- a/sound/usb/endpoint.h +++ b/sound/usb/endpoint.h @@ -28,6 +28,7 @@ void snd_usb_endpoint_release(struct snd_usb_endpoint *ep); void snd_usb_endpoint_free(struct snd_usb_endpoint *ep); int snd_usb_endpoint_implicit_feedback_sink(struct snd_usb_endpoint *ep); +int snd_usb_endpoint_slave_next_packet_size(struct snd_usb_endpoint *ep); int snd_usb_endpoint_next_packet_size(struct snd_usb_endpoint *ep); void snd_usb_handle_sync_urb(struct snd_usb_endpoint *ep, diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index a4e4064f9aee..b50965ab3b3a 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -1579,6 +1579,8 @@ static void prepare_playback_urb(struct snd_usb_substream *subs, for (i = 0; i < ctx->packets; i++) { if (ctx->packet_size[i]) counts = ctx->packet_size[i]; + else if (ep->sync_master) + counts = snd_usb_endpoint_slave_next_packet_size(ep); else counts = snd_usb_endpoint_next_packet_size(ep); -- 2.26.2

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[PATCH 00/11] ASoC: QCOM: Add support for SC7180 lpass variant
by Ajit Pandey 25 Apr '20

25 Apr '20

We've common ASoC cpu driver for QCOM LPASS soc varaints. This patch chain add support for new variant SC7180 soc by doing the required modification in existing common lpass-cpu driver. Below is a brief summary of patch series: PATCH 01 ... 04: Update lpass-cpu driver files to make it more generic PATCH 05 ... 08: Add changes to support SC7180 specific configuration PATCH 09 ... 11: Add new drivers and documentation for SC7180 soc lpass Ajit Pandey (11): Documentation: device-tree: sound: Update lpass-cpu driver binding ASoC: qcom: lpass: Add struct lpass_dai to store dai clocks pointer ASoC: qcom: Add common array to initialize soc based core clocks ASoC: qcom: lpass-cpu: Make "ahbix-clk" an optional clock. ASoC: qcom: lpass: Add support for newer lpass version dt-bindings: sound: Add bindings related to lpass-cpu configuration Documentation: dt-bindings: sound: Add details for new dai properties ASoC: qcom : lpass: Add support to configure dai's connection mode device-tree: bindings: sound: lpass-cpu: Add new compatible soc ASoC: qcom: lpass-sc7180: Add platform driver for lpass audio ASoC: qcom: lpass-platform: Replace card->dev with component->dev .../devicetree/bindings/sound/qcom,lpass-cpu.txt | 158 ------------- .../devicetree/bindings/sound/qcom,lpass-cpu.yaml | 159 +++++++++++++ include/dt-bindings/sound/qcom,lpass.h | 31 +++ sound/soc/qcom/Kconfig | 5 + sound/soc/qcom/Makefile | 2 + sound/soc/qcom/lpass-apq8016.c | 39 ++- sound/soc/qcom/lpass-cpu.c | 263 ++++++++++++++------- sound/soc/qcom/lpass-lpaif-reg.h | 182 +++++++++++++- sound/soc/qcom/lpass-platform.c | 36 +-- sound/soc/qcom/lpass-sc7180.c | 192 +++++++++++++++ sound/soc/qcom/lpass.h | 40 +++- 11 files changed, 808 insertions(+), 299 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/qcom,lpass-cpu.txt create mode 100644 Documentation/devicetree/bindings/sound/qcom,lpass-cpu.yaml create mode 100644 include/dt-bindings/sound/qcom,lpass.h create mode 100644 sound/soc/qcom/lpass-sc7180.c -- 1.9.1

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[Help] Are there tools to change Audio Mixer other than HDA Analyzer?
by andythe_great 24 Apr '20

24 Apr '20

Hello, I could not find appropriate mailing list for this question. I think HDA Analyzer is retired. https://www.alsa-project.org/main/index.php/HDA_Analyzer So I am finding other tool to change Audio Mixer like how HDA Analyzer do it. I attached a screenshot to show what Audio Mixer is in the HDA Analyzer which I try to figure out where it is in hdajackretask. Thanks Andy

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[PATCH] soundwire: intel: Remove unused function
by Sudip Mukherjee 24 Apr '20

24 Apr '20

The function sdw_intel_init() is not used anywhere, remove it for now. Signed-off-by: Sudip Mukherjee <sudipm.mukherjee(a)gmail.com> --- drivers/soundwire/intel_init.c | 23 ----------------------- 1 file changed, 23 deletions(-) diff --git a/drivers/soundwire/intel_init.c b/drivers/soundwire/intel_init.c index ad7053463889..2f37dad06321 100644 --- a/drivers/soundwire/intel_init.c +++ b/drivers/soundwire/intel_init.c @@ -183,29 +183,6 @@ static acpi_status sdw_intel_acpi_cb(acpi_handle handle, u32 level, } /** - * sdw_intel_init() - SoundWire Intel init routine - * @parent_handle: ACPI parent handle - * @res: resource data - * - * This scans the namespace and creates SoundWire link controller devices - * based on the info queried. - */ -static void *sdw_intel_init(acpi_handle *parent_handle, - struct sdw_intel_res *res) -{ - acpi_status status; - - status = acpi_walk_namespace(ACPI_TYPE_DEVICE, - parent_handle, 1, - sdw_intel_acpi_cb, - NULL, res, NULL); - if (ACPI_FAILURE(status)) - return NULL; - - return sdw_intel_add_controller(res); -} - -/** * sdw_intel_exit() - SoundWire Intel exit * @arg: callback context * -- 2.11.0

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[PATCH] ALSA: pcm: oss: Place the plugin buffer overflow checks correctly (for 5.7)
by Takashi Iwai 24 Apr '20

24 Apr '20

[ This is again a forward-port of the fix applied for 5.6-base code (commit 4285de0725b1) to 5.7-base, hence neither Fixes nor Cc-to-stable tags are included here -- tiwai ] The checks of the plugin buffer overflow in the previous fix by commit f2ecf903ef06 ("ALSA: pcm: oss: Avoid plugin buffer overflow") are put in the wrong places mistakenly, which leads to the expected (repeated) sound when the rate plugin is involved. Fix in the right places. Also, at those right places, the zero check is needed for the termination node, so added there as well, and let's get it done, finally. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/oss/pcm_plugin.c | 10 ++++++---- 1 file changed, 6 insertions(+), 4 deletions(-) diff --git a/sound/core/oss/pcm_plugin.c b/sound/core/oss/pcm_plugin.c index 59d62f05658f..1545f8fdb4db 100644 --- a/sound/core/oss/pcm_plugin.c +++ b/sound/core/oss/pcm_plugin.c @@ -205,13 +205,14 @@ static snd_pcm_sframes_t calc_dst_frames(struct snd_pcm_substream *plug, plugin = snd_pcm_plug_first(plug); while (plugin && frames > 0) { plugin_next = plugin->next; + if (check_size && plugin->buf_frames && + frames > plugin->buf_frames) + frames = plugin->buf_frames; if (plugin->dst_frames) { frames = plugin->dst_frames(plugin, frames); if (frames < 0) return frames; } - if (check_size && frames > plugin->buf_frames) - frames = plugin->buf_frames; plugin = plugin_next; } return frames; @@ -225,14 +226,15 @@ static snd_pcm_sframes_t calc_src_frames(struct snd_pcm_substream *plug, plugin = snd_pcm_plug_last(plug); while (plugin && frames > 0) { - if (check_size && frames > plugin->buf_frames) - frames = plugin->buf_frames; plugin_prev = plugin->prev; if (plugin->src_frames) { frames = plugin->src_frames(plugin, frames); if (frames < 0) return frames; } + if (check_size && plugin->buf_frames && + frames > plugin->buf_frames) + frames = plugin->buf_frames; plugin = plugin_prev; } return frames; -- 2.16.4

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[PATCH] ALSA: pcm: oss: Place the plugin buffer overflow checks correctly
by Takashi Iwai 24 Apr '20

24 Apr '20

The checks of the plugin buffer overflow in the previous fix by commit f2ecf903ef06 ("ALSA: pcm: oss: Avoid plugin buffer overflow") are put in the wrong places mistakenly, which leads to the expected (repeated) sound when the rate plugin is involved. Fix in the right places. Also, at those right places, the zero check is needed for the termination node, so added there as well, and let's get it done, finally. Fixes: f2ecf903ef06 ("ALSA: pcm: oss: Avoid plugin buffer overflow") Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/oss/pcm_plugin.c | 20 ++++++++++++-------- 1 file changed, 12 insertions(+), 8 deletions(-) diff --git a/sound/core/oss/pcm_plugin.c b/sound/core/oss/pcm_plugin.c index 3ecc070738e8..c1315ce98b54 100644 --- a/sound/core/oss/pcm_plugin.c +++ b/sound/core/oss/pcm_plugin.c @@ -211,21 +211,23 @@ static snd_pcm_sframes_t plug_client_size(struct snd_pcm_substream *plug, if (stream == SNDRV_PCM_STREAM_PLAYBACK) { plugin = snd_pcm_plug_last(plug); while (plugin && drv_frames > 0) { - if (check_size && drv_frames > plugin->buf_frames) - drv_frames = plugin->buf_frames; plugin_prev = plugin->prev; if (plugin->src_frames) drv_frames = plugin->src_frames(plugin, drv_frames); + if (check_size && plugin->buf_frames && + drv_frames > plugin->buf_frames) + drv_frames = plugin->buf_frames; plugin = plugin_prev; } } else if (stream == SNDRV_PCM_STREAM_CAPTURE) { plugin = snd_pcm_plug_first(plug); while (plugin && drv_frames > 0) { plugin_next = plugin->next; + if (check_size && plugin->buf_frames && + drv_frames > plugin->buf_frames) + drv_frames = plugin->buf_frames; if (plugin->dst_frames) drv_frames = plugin->dst_frames(plugin, drv_frames); - if (check_size && drv_frames > plugin->buf_frames) - drv_frames = plugin->buf_frames; plugin = plugin_next; } } else @@ -251,26 +253,28 @@ static snd_pcm_sframes_t plug_slave_size(struct snd_pcm_substream *plug, plugin = snd_pcm_plug_first(plug); while (plugin && frames > 0) { plugin_next = plugin->next; + if (check_size && plugin->buf_frames && + frames > plugin->buf_frames) + frames = plugin->buf_frames; if (plugin->dst_frames) { frames = plugin->dst_frames(plugin, frames); if (frames < 0) return frames; } - if (check_size && frames > plugin->buf_frames) - frames = plugin->buf_frames; plugin = plugin_next; } } else if (stream == SNDRV_PCM_STREAM_CAPTURE) { plugin = snd_pcm_plug_last(plug); while (plugin) { - if (check_size && frames > plugin->buf_frames) - frames = plugin->buf_frames; plugin_prev = plugin->prev; if (plugin->src_frames) { frames = plugin->src_frames(plugin, frames); if (frames < 0) return frames; } + if (check_size && plugin->buf_frames && + frames > plugin->buf_frames) + frames = plugin->buf_frames; plugin = plugin_prev; } } else -- 2.16.4

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ASoC driver names
by Jaroslav Kysela 24 Apr '20

24 Apr '20

Hi all, I am fighting actually with the ASoC driver names. The current situation, where each hardware variant (card name) is mapped to the driver name is not very practical (and not correct). The driver name should describe the common part (usually the controller - like 'HDA-Intel' or bus 'USB-Audio'). I am talking about the situation, where the ASoC card name is set via the device tree (snd_soc_of_parse_card_name) like in [1], but the change may be considered for other drivers like Intel SST: Card driver name: Lenovo-YOGA-C63 Card name: Lenovo-YOGA-C630-13Q50 Card long name: LENOVO-81JL-LenovoYOGAC630_13Q50-LNVNB161216 [1] https://github.com/alsa-project/alsa-tests/pull/3/commits/1b5a552cff06c6039… My question is, can we change / add the more apropriate driver names conditionally with a new kernel CONFIG option? Will you accept this change? The goal is to group the related UCM2 configurations and do the required split inside the UCM2 top-level configuration file based on card components string or other card identificators (related to the driver). Thanks, Jaroslav -- Jaroslav Kysela <perex(a)perex.cz> Linux Sound Maintainer; ALSA Project; Red Hat, Inc.

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[PATCH] ASoC: snd-sof-intel-hda-common - add hda_model parameter and pass it to HDA codec driver
by Jaroslav Kysela 24 Apr '20

24 Apr '20

It may be useful to pass the specific model to the generic HDA codec routines like the legacy HDA driver (snd-hda-intel) allows. The model name "sofbus" is tricky anyway. Original proposal: https://lore.kernel.org/alsa-devel/20191203161908.7496-1-perex@perex.cz/ Signed-off-by: Jaroslav Kysela <perex(a)perex.cz> Cc: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Cc: Mark Brown <broonie(a)kernel.org> --- sound/soc/sof/intel/hda.c | 6 +++++- 1 file changed, 5 insertions(+), 1 deletion(-) diff --git a/sound/soc/sof/intel/hda.c b/sound/soc/sof/intel/hda.c index 211e91e79eae..ea0189ee8939 100644 --- a/sound/soc/sof/intel/hda.c +++ b/sound/soc/sof/intel/hda.c @@ -282,6 +282,10 @@ module_param_named(use_msi, hda_use_msi, bool, 0444); MODULE_PARM_DESC(use_msi, "SOF HDA use PCI MSI mode"); #endif +static char *hda_model; +module_param(hda_model, charp, 0444); +MODULE_PARM_DESC(hda_model, "Use the given HDA board model."); + #if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA) static int hda_dmic_num = -1; module_param_named(dmic_num, hda_dmic_num, int, 0444); @@ -503,7 +507,7 @@ static int hda_init(struct snd_sof_dev *sdev) mutex_init(&hbus->prepare_mutex); hbus->pci = pci; hbus->mixer_assigned = -1; - hbus->modelname = "sofbus"; + hbus->modelname = hda_model; /* initialise hdac bus */ bus->addr = pci_resource_start(pci, 0); -- 2.25.3

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Re: [PATCH] ASoC: fsl_easrc: Check for null pointer before dereferencing “ctx” in fsl_easrc_hw_free()
by Markus Elfring 24 Apr '20

24 Apr '20

> The patch 955ac624058f: "ASoC: fsl_easrc: Add EASRC ASoC CPU DAI > drivers" from Apr 16, 2020, leads to the following Smatch complaint: > > sound/soc/fsl/fsl_easrc.c:1529 fsl_easrc_hw_free() > warn: variable dereferenced before check 'ctx' (see line 1527) Please avoid a typo in the final commit subject. Regards, Markus

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[PATCH v3 0/3] Add support for different DMIC configurations
by Mateusz Gorski 24 Apr '20

24 Apr '20

Set of patches to enable DMIC capture on different hardware configurations. Information about supported DMIC configuration is read from NHLT and correct pipeline configuration is selected automatically. Also, adding additional option for topology binary name which is based on used machine driver. Changes in v2: - removed patch 1/4 swapping machine device and platform device registration order - alt_tplg_name creation now uses different field to read machine driver name - including of <sound/soc-acpi.h> moved to different patch Changes in v3: - cosmetic changes in skl-topology.c file Mateusz Gorski (3): ASoC: Intel: Skylake: Add alternative topology binary name ASoC: Intel: Multiple I/O PCM format support for pipe ASoC: Intel: Skylake: Automatic DMIC format configuration according to information from NHLT include/uapi/sound/skl-tplg-interface.h | 2 + sound/soc/intel/skylake/skl-topology.c | 188 +++++++++++++++++++++++- sound/soc/intel/skylake/skl-topology.h | 1 + 3 files changed, 186 insertions(+), 5 deletions(-) -- 2.17.1

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