October 2015 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH 00/10] Add support for Imagination Technologies audio controllers
by Damien Horsley 04 Nov '15

04 Nov '15

From: "Damien.Horsley" <Damien.Horsley(a)imgtec.com> Add drivers and binding documents for the following Imagination Technologies audio controllers: I2S Input Controller I2S Output Controller Parallel Output Controller SPDIF Input Controller SPDIF Output Controller These controllers are used in Pistachio SoC Damien.Horsley (10): ASoC: img: Add binding document for I2S input controller ASoC: img: Add driver for I2S input controller ASoC: img: Add binding document for I2S output controller ASoC: img: Add driver for I2S output controller ASoC: img: Add binding document for parallel output controller ASoC: img: Add driver for parallel output controller ASoC: img: Add binding document for SPDIF input controller ASoC: img: Add driver for SPDIF input controller ASoC: img: Add binding document for SPDIF output controller ASoC: img: Add driver for SPDIF output controller .../devicetree/bindings/sound/img,i2s-in.txt | 45 ++ .../devicetree/bindings/sound/img,i2s-out.txt | 49 ++ .../devicetree/bindings/sound/img,parallel-out.txt | 44 ++ .../devicetree/bindings/sound/img,spdif-in.txt | 41 ++ .../devicetree/bindings/sound/img,spdif-out.txt | 44 ++ sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/img/Kconfig | 44 ++ sound/soc/img/Makefile | 5 + sound/soc/img/img-i2s-in.c | 506 +++++++++++++ sound/soc/img/img-i2s-out.c | 555 ++++++++++++++ sound/soc/img/img-parallel-out.c | 372 ++++++++++ sound/soc/img/img-spdif-in.c | 797 +++++++++++++++++++++ sound/soc/img/img-spdif-out.c | 432 +++++++++++ 14 files changed, 2936 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/img,i2s-in.txt create mode 100644 Documentation/devicetree/bindings/sound/img,i2s-out.txt create mode 100644 Documentation/devicetree/bindings/sound/img,parallel-out.txt create mode 100644 Documentation/devicetree/bindings/sound/img,spdif-in.txt create mode 100644 Documentation/devicetree/bindings/sound/img,spdif-out.txt create mode 100644 sound/soc/img/Kconfig create mode 100644 sound/soc/img/Makefile create mode 100644 sound/soc/img/img-i2s-in.c create mode 100644 sound/soc/img/img-i2s-out.c create mode 100644 sound/soc/img/img-parallel-out.c create mode 100644 sound/soc/img/img-spdif-in.c create mode 100644 sound/soc/img/img-spdif-out.c -- 2.1.4

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[alsa-devel] [PATCH MC Next Gen v2 0/3] Add mixer and control interface media entities
by Shuah Khan 03 Nov '15

03 Nov '15

This patch v2 series adds the following: A fix to media_stream_delete() to remove intf devnode. Add support for creating MEDIA_ENT_F_AUDIO_MIXER entity for each mixer and a MEDIA_INTF_T_ALSA_CONTROL control interface entity that links to mixer entities. MEDIA_INTF_T_ALSA_CONTROL entity corresponds to the control device for the card. Change au0828_create_media_graph() to create pad link between MEDIA_ENT_F_AUDIO_MIXER entity and decoder's AU8522_PAD_AUDIO_OUT. With mixer entity now linked to decoder, change to link MEDIA_ENT_F_AUDIO_CAPTURE to mixer's source pad Changes since v1: 1. Included the fix to media_stream_delete() which was by mistake not included in the patch v1. 2. Fixed mixer to decoder link and mider to capture node links based on Mauro Chehapb's review comments on the media graph. This patch series is dependent on an earlier patch series: Update ALSA, and au0828 drivers to use Managed Media Controller API: https://www.mail-archive.com/linux-media@vger.kernel.org/msg92752.html Please find the media graphs for ALSA work with mixer and control interface nodes: https://drive.google.com/folderview?id=0B0NIL0BQg-AlLWE3SzAxazBJWm8&usp=sha… Shuah Khan (3): sound/usb: Fix media_stream_delete() to remove intf devnode sound/usb: Create media mixer function and control interface entities media: au0828 create link between ALSA Mixer and decoder drivers/media/usb/au0828/au0828-core.c | 17 +++++-- drivers/media/usb/au0828/au0828.h | 1 + sound/usb/card.c | 5 ++ sound/usb/media.c | 90 ++++++++++++++++++++++++++++++++++ sound/usb/media.h | 20 ++++++++ sound/usb/mixer.h | 1 + sound/usb/usbaudio.h | 1 + 7 files changed, 132 insertions(+), 3 deletions(-) -- 2.1.4

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[alsa-devel] [RFT][PATCH 1/2] ASoC: rt286: Fix run time error while modifying const data
by Axel Lin 03 Nov '15

03 Nov '15

Make a copy of memory for index_cache rather than directly use the rt286_index_def to avoid run time error. Fixes: c418a84a8c8f ("ASoC: Constify reg_default tables") Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/codecs/rt286.c | 8 ++++++-- 1 file changed, 6 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/rt286.c b/sound/soc/codecs/rt286.c index 1fbdb4f..af2ed77 100644 --- a/sound/soc/codecs/rt286.c +++ b/sound/soc/codecs/rt286.c @@ -37,7 +37,7 @@ #define RT288_VENDOR_ID 0x10ec0288 struct rt286_priv { - const struct reg_default *index_cache; + struct reg_default *index_cache; int index_cache_size; struct regmap *regmap; struct snd_soc_codec *codec; @@ -1160,7 +1160,11 @@ static int rt286_i2c_probe(struct i2c_client *i2c, return -ENODEV; } - rt286->index_cache = rt286_index_def; + rt286->index_cache = devm_kmemdup(&i2c->dev, rt286_index_def, + sizeof(rt286_index_def), GFP_KERNEL); + if (!rt286->index_cache) + return -ENOMEM; + rt286->index_cache_size = INDEX_CACHE_SIZE; rt286->i2c = i2c; i2c_set_clientdata(i2c, rt286); -- 2.1.4

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[alsa-devel] [PATCH 1/2] ASoC: rt298: fix wrong setting of gpio2_en
by Bard Liao 03 Nov '15

03 Nov '15

The register value to enable gpio2 was incorrect. So fix it. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- sound/soc/codecs/rt298.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt298.c b/sound/soc/codecs/rt298.c index 56e5e1a..b3f795c 100644 --- a/sound/soc/codecs/rt298.c +++ b/sound/soc/codecs/rt298.c @@ -1217,7 +1217,7 @@ static int rt298_i2c_probe(struct i2c_client *i2c, mdelay(10); if (!rt298->pdata.gpio2_en) - regmap_write(rt298->regmap, RT298_SET_DMIC2_DEFAULT, 0x4000); + regmap_write(rt298->regmap, RT298_SET_DMIC2_DEFAULT, 0x40); else regmap_write(rt298->regmap, RT298_SET_DMIC2_DEFAULT, 0); -- 1.8.1.1.439.g50a6b54

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[alsa-devel] Noise in tlv320aic3104-based audio system
by Rick Mann 02 Nov '15

02 Nov '15

I have a custom cape on a Beaglebone Black that uses a tlv320aic3104, connected to a tpa3140d2 audio power amp. I currently manually control the /shudown pins on the two ICs, and just turn them on before doing anything else. The system works. But I notice that when I open the output device for writing (via libao), I immediately hear hiss, even before I send any samples to the device. After I close the libao audio device, a few seconds later Linux sends some i2c commands to the tlv320 and it's once again silent. The perceived amplitude of this hiss does not seem to be affected by "amixer set PCM <0 - 127>" calls. Does anyone have any idea where this noise might be coming from? Is it just thermal or other noise in the routing of audio within the tlv320? Is there some configuration or routing I might be able to change to avoid it? Did I screw up somewhere in my circuit design or layout that only manifests when linux configures the chip for operation? Any ideas are appreciated. Thanks. -- Rick Mann rmann(a)latencyzero.com

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[alsa-devel] fsl_ssi.c: Getting channel slips with fsl_ssi.c in TDM (network) mode.
by Caleb Crome 02 Nov '15

02 Nov '15

Hello Arnaud, all, I'm trying to get the i.MX6 SSI working in 16-channel TDM mode. FYI, I'm working with the stock 4.3 kernel at https://github.com/torvalds/linux. When I apply the patch below, the SSI does all configure properly and even starts streaming properly with all the bits in the right place on the SSI pins. However, given a little bit of time and/or IO to the SD card, the bit-stream slips by 1 slot, causing all of the channels to be misaligned. My changes to the SSI driver are very minimal (shown below), and amount to forcing it into network (TDM) mode, and changing the maximum channels, and setting the STCCR DC mask. There is no indication from user space that anything has slipped, so the data stream just continues on shifted by 1 slot. You (Arnaud) mentioned in a previous thread ("Multiple codecs on one sound card for multi-channel sound card"), that I should just have to set channels_max (and presumably the other changes I mentioned), and it'll work mostly. However, it's very unreliable at the moment. Any thoughts to how I can diagnose this problem would be greatly appreciated! So, what happens is this: The SSI Starts sending data like this: SLOT 0 1 2 3 4 5 6 7 8 9 10 11 12 13 14 15 DATA 0 1 2 3 4 5 6 7 8 9 10 11 12 13 14 15 But then after time, something slips and without warning it goes to: SLOT 0 1 2 3 4 5 6 7 8 9 10 11 12 13 14 15 DATA 15 0 1 2 3 4 5 6 7 8 9 10 11 12 13 14 diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 37c5cd4..73778c2 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -749,7 +749,10 @@ static int fsl_ssi_hw_params(struct snd_pcm_substream *substream, CCSR_SSI_SCR_NET | CCSR_SSI_SCR_I2S_MODE_MASK, channels == 1 ? 0 : i2smode); } - + ssi_private->i2s_mode = CCSR_SSI_SCR_I2S_MODE_NORMAL | CCSR_SSI_SCR_NET; + regmap_update_bits(regs, CCSR_SSI_SCR, + CCSR_SSI_SCR_NET | CCSR_SSI_SCR_I2S_MODE_MASK, + ssi_private->i2s_mode); /* * FIXME: The documentation says that SxCCR[WL] should not be * modified while the SSI is enabled. The only time this can @@ -863,6 +866,15 @@ static int _fsl_ssi_set_dai_fmt(struct device *dev, return -EINVAL; } scr |= ssi_private->i2s_mode; + // Set to 16 slots/frame + regmap_update_bits(regs, CCSR_SSI_STCCR, + CCSR_SSI_SxCCR_DC_MASK, + CCSR_SSI_SxCCR_DC(16)); + + regmap_update_bits(regs, CCSR_SSI_SRCCR, + CCSR_SSI_SxCCR_DC_MASK, + CCSR_SSI_SxCCR_DC(16)); + /* DAI clock inversion */ switch (fmt & SND_SOC_DAIFMT_INV_MASK) { @@ -1084,14 +1099,14 @@ static struct snd_soc_dai_driver fsl_ssi_dai_template = { .playback = { .stream_name = "CPU-Playback", .channels_min = 1, - .channels_max = 2, + .channels_max = 16, .rates = FSLSSI_I2S_RATES, .formats = FSLSSI_I2S_FORMATS, }, .capture = { .stream_name = "CPU-Capture", .channels_min = 1, - .channels_max = 2, + .channels_max = 16, .rates = FSLSSI_I2S_RATES, .formats = FSLSSI_I2S_FORMATS, }, Another thing I have tried is changing the watermark level for the fifo to give the DMA interrupt some extra time. The problem still happens, but seems to be a bit better. The fifo watermark change is this: diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 73778c2..7c2e4b0 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -54,6 +54,8 @@ #include "fsl_ssi.h" #include "imx-pcm.h" +#define WATERMARK 8 + /** * FSLSSI_I2S_RATES: sample rates supported by the I2S * @@ -943,7 +950,7 @@ static int _fsl_ssi_set_dai_fmt(struct device *dev, * size. */ if (ssi_private->use_dma) - wm = ssi_private->fifo_depth - 2; + wm = ssi_private->fifo_depth - WATERMARK; else wm = ssi_private->fifo_depth; @@ -1260,8 +1267,8 @@ static int fsl_ssi_imx_probe(struct platform_device *pdev, * We have burstsize be "fifo_depth - 2" to match the SSI * watermark setting in fsl_ssi_startup(). */ - ssi_private->dma_params_tx.maxburst = ssi_private->fifo_depth - 2; - ssi_private->dma_params_rx.maxburst = ssi_private->fifo_depth - 2; + ssi_private->dma_params_tx.maxburst = ssi_private->fifo_depth - WATERMARK; + ssi_private->dma_params_rx.maxburst = ssi_private->fifo_depth - WATERMARK; ssi_private->dma_params_tx.addr = ssi_private->ssi_phys + CCSR_SSI_STX0; ssi_private->dma_params_rx.addr = ssi_private->ssi_phys + CCSR_SSI_SRX0; Thanks, -Caleb

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[alsa-devel] [PATCH] ASoC: da7219: Use logical instead of bitwise OR for boolean expression
by Axel Lin 02 Nov '15

02 Nov '15

Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/codecs/da7219.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/da7219.c b/sound/soc/codecs/da7219.c index abba4b3..86c528a 100644 --- a/sound/soc/codecs/da7219.c +++ b/sound/soc/codecs/da7219.c @@ -1306,7 +1306,7 @@ static int da7219_hw_params(struct snd_pcm_substream *substream, } channels = params_channels(params); - if ((channels < 1) | (channels > DA7219_DAI_CH_NUM_MAX)) { + if ((channels < 1) || (channels > DA7219_DAI_CH_NUM_MAX)) { dev_err(codec->dev, "Invalid number of channels, only 1 to %d supported\n", DA7219_DAI_CH_NUM_MAX); -- 2.1.4

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[alsa-devel] ALSA: USB-audio: Add quirk for Zoom R16/24 playback
by Alain 02 Nov '15

02 Nov '15

Hello, First of all, sorry for my english I'm French speaking. Some time ago I was planning to buy a ZOOM R24 but after some research on internet I saw that machine was not fully functional under linux. till some nice people took care with that problem. As I can see, but i'm not sure, the problem is fixed Can you confirm me the ZOOM R16/24 is ok for 8 inputs/2 outputs ? Is the next kernel version will include the modifications? or should I do the modifications (compliling kernel is a big chalenge for me) ? Thanks for your job Alain

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[alsa-devel] [Minutes] ELCE Audio mini conf.
by Liam Girdwood 01 Nov '15

01 Nov '15

I've written up the minutes here below (including Vinod's minutes after I left for my flight). Please do feel free to correct or add any missing items. If you intend to reply and discuss a topic in detail then it would probably be helpful to include the topic in the $SUBJECT (so that it's easy to track multiple topics that may arise from the minutes). Jaroslav, I have some PDF slides from a few delegates. Can I post them on to the Wiki ? This failed last time I tried after the Edinburgh audio conference (Wiki could not upload PDFs ??). Soundwire: Pierre ================= - Slides to be posted. - Shadow registers required for regamp. Mark: need to make sure we can commit changes, and rules for witting. Reuse of codec drivers since reg mapping should be similar to i2c regmaps. - Some devices wont support run time clock changes. Need compatible devices on bus. - Each device has unique ID set by HW. Device enumeration can be auto discovered. Can detect device but not capabilities. - Device capabilities can come from hard coded driver data or ACPI/DT data. Windows may use inf files for capabilities. - Define bus api for master "controller". - Grouped triggers can be used for start duplex streams for similtanious capture and playback. - DAI is physical connection, logical links need to be worked out with Routing config Mark: Redefining of ports/slots at runtime i.e. Leave up to userspace for handling supported configurations of ports. Energy efficiency: Alexander (Minutes also supplied by Alexander). ================================================================== Power consumption of a SandyBridge-based Sony laptop has been measured with a workload that does only audio playback. To ensure a reproducible workload, the whole system has been constructed by a script using debootstrap, and placed into an initramfs. Various versions of PulseAudio were compared to CRAS and to the use of ALSA hw device, at latencies 7 ms, 28 ms and 448 ms. Even in the best case (ALSA), going from 448 to 28 ms means using extra 300 mW. For PulseAudio, the overhead of using low latency is even higher, presumably because of a chatty protocol. In version 7, there was some work (srbchannel) with the aim to reduce this overhead, but there is still more way to go. There was a suggestion to use perf to analyze the problem further. It should be noted that latencies as high as 448 ms are not usable without the ability to rewind, which PulseAudio posesses (at the cost of more complex, and actually incorrect, code) and CRAS doesn't. So CRAS decision to never do rewinds costs it 300 mW (448 ms latency is usable only for benchmarking purposes). Note: at least two people in the audience misunderstood this statement as originally formulated (thinko?), and got it as "PulseAudio decision to do rewinds costs it 300 mW", which is of course wrong. The cost of rewinds is in code complexity, real user-reported bugs and "how to test this code path" questions from developers, which, in the opinion of Alexander Patrakov, may outweigh the gains in energy efficiency (Arun Raghavan disagrees). A proposal was made by P.-L. Bossart to introduce a hw param that the application would set if it intends to never do rewinds, with the implication that a SkyLake driver can use this information for codec-side power saving (by prefetching audio data and powering the link off). Arun Raghavan (presumably due to the above misunderstanding) disagreed and instead proposed to make the device non-rewindable if this means power savings. Alexander is in favor of the hw param, because it would be useful for CRAS, but harmful for the current PulseAudio (it would force abandoning high latency). A future version of PulseAudio could, though, use a separate DSP stream for a music application (if there is only one), and, because a new application never appears, and the hardware volume is precise enough, there would never be a need to rewind. This, though, would need eliminating the support for rewinds from the client API (the last two parameters for pa_stream_write()), which can be done by ignoring these parameters. Documentation: Takashi ====================== - Patches that introduce new functionality should also include documentation. - Caretaker required for docs. Who ? - Uploading materials, in central location, ask kernel.org. - Charles and Rakesh voluntered to write some docs. Alsa-lib: release schedule: Vinod. ================================== - Request for more frequent release. Release inline with 1 - 2 kernel releases. - Integrate tinycompress into alsa release. Testing/QA: Liam, Takashi, Mark, Pierre ======================================= - BAT tool: Now upstream, WiP for review comments docs and new features. - Mark doing brute force tool for kcontrol testing. Kcontrol status can be watched via debubFS. Multiple threads. - Takashi: Need unit testing codec drivers. Some codec vendors have tools for this but something generic would be good. QA unit test could use topology readback to check topology in userspace. - Pierre: Latency testing, USB device to toggle GPIO (to indicate stream start) and audio latency measured from this point. Available on 01.org latency tools. To be extended to BAT. HDA/gfx: relevant people not in room. ===================================== - Pierre: Hotplug DP over USB C required. Needs input from GFX folks. BATCH flag for USB: Arun. ========================= - Flag does not respond to reality, lets deprecate it. no users. - Dylan: need to know transfer size for CRAS (uses extra samples for buffering). - BATCH flag means period size transfers, applications that use new granularity API can ignore batch flag. Pierre: to implement. Splitting out controls: Takashi =============================== - Restricted access. Consensus to restrict access to some controls due to possibility of breaking HW at kernel level. i.e. prevent feeding digital Mic into HP amp to prevent speaker over heating. - Some clamping APIs already exist, but these should be extended to include DAPM. Simple card: ACPI support. Vinod ================================ - Clock drivers required alongside helper functions to help initialise DAI links etc. - Vinod to publish spec as RFC in Q4. - Doubts of using DMI names for differentiation. Topology: Liam, Mengdong ======================== - Proposals to use debugFS/ioctl/media controller for exporting topology data to userspace. Drawbacks with all three approaches, but are aimed at different end users i.e. debugFS aimed at kernel audio developers/integrators/testers and media aimed at userspace developers and end users. Will probably support both debugFS (or other kernel file) and media API. Mark to discuss at Media team in Seoul. ASoC internals to gather data for debugFS and Media API will be the same code so nothing blocking initial development. Tiny-fication. Keyon, Liam ========================== - IoT devices have tight memory footprints so audio kernel and userspace can be very limited. - Userspace is fine with tinyalsa. - Currently Keyon has worked on some IoT device options to disable ALSA functions. e.g. code saving of 13k for refinement, but some IoT options can break existing userpsace. - Suggestion is to research tinycompress API for general PCM usage. - Look at removing DAPM strings (will need topology dump tool to automate this though). - Some structures can also be shrunk. Android configuration consolidation =================================== - different configuration formats XML, ucm, This tool takes Android XML and converts to UCM - uses Sony Xperia - generates UCM file from XML - Android: Try to converge to a HAL, but diverges quickly. Quirks for modem etc cant be represented well. - Arun will send the link to his work HDA Restructuring: Vinod & Rakesh ================================= - Discussed HW and SW overview - Future HDA work and codec conversion discussion <dont have detailed notes here as I was talking> DPMST: Mengdong ================ - One DVI port can be used to transmit multiple independent streams. - Three display pipelines in HW, so at most 3 streams can be supported across all ports - Need to modify representation to have stream on device rather than per pin. - While moving the monitor can be incompatible so should be modified - display should tell us unplug/plug on move. Sound server reacts to it. - This notification thru Component fwk. - Various options are discussed. Needs more offline discussion on approach and to solve this issue ALSA Core Challenges: ====================== - ALSA Core locking is complicated. Core code is quite difficult to understand. - PCM linking makes things complex - Add documentation for locks. - Controls can be hidden in UI tools through iface_cards. - DPCM hidden PCMs should not be shown in usermode, hide them from Usermode Multi Channel ASoC Controls; ============================= - channel maps dont make sense for ASoC as asoc paths dont have channel information - conclusion: add channel maps on HDMI edge. - we can add multi-channel controls to the asoc Liam

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[alsa-devel] Applied "ASoC: topology: ABI - Rename dai_elems to pcm_elems in manifest" to the asoc tree
by Mark Brown 31 Oct '15

31 Oct '15

The patch ASoC: topology: ABI - Rename dai_elems to pcm_elems in manifest has been applied to the asoc tree at git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From bbb64300469d207f62c08ea7358c839991a52877 Mon Sep 17 00:00:00 2001 From: Mengdong Lin <mengdong.lin(a)intel.com> Date: Fri, 30 Oct 2015 15:13:16 +0800 Subject: [PATCH] ASoC: topology: ABI - Rename dai_elems to pcm_elems in manifest This field is the number of PCM objects (a pair of FE DAI and DAI link). Signed-off-by: Lin Mengdong <mengdong.lin(a)linux.intel.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- include/uapi/sound/asoc.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/include/uapi/sound/asoc.h b/include/uapi/sound/asoc.h index 26539a7..c4cc1e4 100644 --- a/include/uapi/sound/asoc.h +++ b/include/uapi/sound/asoc.h @@ -243,7 +243,7 @@ struct snd_soc_tplg_manifest { __le32 control_elems; /* number of control elements */ __le32 widget_elems; /* number of widget elements */ __le32 graph_elems; /* number of graph elements */ - __le32 dai_elems; /* number of DAI elements */ + __le32 pcm_elems; /* number of PCM elements */ __le32 dai_link_elems; /* number of DAI link elements */ struct snd_soc_tplg_private priv; } __attribute__((packed)); -- 2.6.1

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