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

[alsa-devel] [PATCH] soundwire: slave: fix scanf format
by Pierre-Louis Bossart 24 Oct '19

24 Oct '19

fix cppcheck warning: [drivers/soundwire/slave.c:145]: (warning) %x in format string (no. 1) requires 'unsigned int *' but the argument type is 'signed int *'. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- drivers/soundwire/slave.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/drivers/soundwire/slave.c b/drivers/soundwire/slave.c index 48a63ca130d2..6473fa602f82 100644 --- a/drivers/soundwire/slave.c +++ b/drivers/soundwire/slave.c @@ -128,7 +128,8 @@ int sdw_of_find_slaves(struct sdw_bus *bus) struct device_node *node; for_each_child_of_node(bus->dev->of_node, node) { - int link_id, sdw_version, ret, len; + int link_id, ret, len; + unsigned int sdw_version; const char *compat = NULL; struct sdw_slave_id id; const __be32 *addr; -- 2.20.1

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[alsa-devel] AC97 audio-routing [was: Recording on WM9712]
by Matej Kupljen 24 Oct '19

24 Oct '19

Hi all, I am not sure how to specify audio-routing for AC97 codec. If I use the similar setting as the UDOO board (which was the only board that uses AC97 codec I could find) I get warnings during booting: [ 3.261369] fsl-asoc-card sound: ac97-hifi <-> 202c000.ssi mapping ok [ 3.267917] fsl-asoc-card sound: ASoC: no sink widget found for Playback [ 3.274642] fsl-asoc-card sound: ASoC: Failed to add route AC97 Playback -> direct -> Playback [ 3.283309] fsl-asoc-card sound: ASoC: no source widget found for Capture [ 3.290141] fsl-asoc-card sound: ASoC: Failed to add route Capture -> direct -> AC97 Capture This was tested on kernel Linux version 4.14.134. However, the card is detected: [ 4.041646] ALSA device list: [ 4.044628] #0: ac97-audio and I can use it for playback normally, but the capture is "slow" as described in my previous e-mail. Then I tried to build and boot the latest kernel from GIT (Linux version 5.4.0-rc4) using the same DTS, I get additional warnings from the kernel: [ 2.639878] fsl-asoc-card sound: ac97-hifi <-> 202c000.ssi mapping ok [ 2.651321] fsl-asoc-card sound: ASoC: source widget AC97 Playback overwritten [ 2.658595] fsl-asoc-card sound: ASoC: no sink widget found for Playback [ 2.665317] fsl-asoc-card sound: ASoC: Failed to add route AC97 Playback -> direct -> Playback [ 2.673974] fsl-asoc-card sound: ASoC: sink widget AC97 Capture overwritten [ 2.680986] fsl-asoc-card sound: ASoC: no source widget found for Capture [ 2.687822] fsl-asoc-card sound: ASoC: Failed to add route Capture -> direct -> AC97 Capture [ 2.718636] fsl-asoc-card sound: snd_soc_register_card failed (-19) and the card is not detected: [ 3.285543] ALSA device list: [ 3.288518] No soundcards found. As you can see, there are additional warnings about widget overwrite, because this check was added to the latest kernel, as seen from the kernel source code. This is the setting from my DTS file: sound { compatible = "fsl,imx-audio-ac97"; model = "fsl,imx6q-ac97"; audio-cpu = <&ssi2>; audio-routing = "RX", "Mic Jack", "Headphone Jack", "TX"; mux-int-port = <2>; mux-ext-port = <5>; }; As you can see, I do not even specify any routing about AC97 Playback/Capture and I still get those warnings. For the test, I commented out the code in sound/soc/soc-core.c in function snd_soc_instantiate_card, to ignore the return values from the call to snd_soc_dapm_add_routes and then I can use the card for playback, but the recording has the issues again. What seems to be wrong with this AC97 audio routing? Any suggestions? I am willing to test this on the latest kernel, or from some other GIT repo if needed. Thanks and BR, Matej

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[alsa-devel] [PATCH] ALSA: usb-audio: Add Gustard U16/X26 device family to quirks for native DSD support
by Fly Pig 24 Oct '19

24 Oct '19

Hi There This patch add native DSD support for Gustard U16/X26 USB Interface. Tested on my unit, working properly. Hence submit the patch. Cheers. Thank you very much. Pig --- a/sound/usb/quirks.c 2019-09-22 21:28:09.757897000 +0000 +++ b/sound/usb/quirks.c 2019-09-22 21:33:36.892274800 +0000 @@ -1369,6 +1369,7 @@ case USB_ID(0x22d9, 0x0461): /* OPPO UDP-205 */ case USB_ID(0x2522, 0x0012): /* LH Labs VI DAC Infinity */ case USB_ID(0x2772, 0x0230): /* Pro-Ject Pre Box S2 Digital */ + case USB_ID(0x292b, 0xc4b3): /* Gustard U16 */ if (fp->altsetting == 2) return SNDRV_PCM_FMTBIT_DSD_U32_BE; break;

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[alsa-devel] [asoc:for-5.5 160/172] sound/soc/codecs/rt5677-spi.c:148:33: sparse: sparse: Using plain integer as NULL pointer
by kbuild test robot 24 Oct '19

24 Oct '19

tree: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-5.5 head: 175fc928198236037174e5c5c066fe3c4691903e commit: a0e0d135427cf699fe2dee77da0924e0b47f3170 [160/172] ASoC: rt5677: Add a PCM device for streaming hotword via SPI reproduce: # apt-get install sparse # sparse version: v0.6.1-dirty git checkout a0e0d135427cf699fe2dee77da0924e0b47f3170 make ARCH=x86_64 allmodconfig make C=1 CF='-fdiagnostic-prefix -D__CHECK_ENDIAN__' If you fix the issue, kindly add following tag Reported-by: kbuild test robot <lkp(a)intel.com> sparse warnings: (new ones prefixed by >>) >> sound/soc/codecs/rt5677-spi.c:148:33: sparse: sparse: Using plain integer as NULL pointer >> sound/soc/codecs/rt5677-spi.c:365:13: sparse: sparse: symbol 'rt5677_spi_pcm_page' was not declared. Should it be static? vim +148 sound/soc/codecs/rt5677-spi.c 139 140 static int rt5677_spi_hw_free( 141 struct snd_soc_component *component, 142 struct snd_pcm_substream *substream) 143 { 144 struct rt5677_dsp *rt5677_dsp = 145 snd_soc_component_get_drvdata(component); 146 147 mutex_lock(&rt5677_dsp->dma_lock); > 148 rt5677_dsp->substream = 0; 149 mutex_unlock(&rt5677_dsp->dma_lock); 150 151 return snd_pcm_lib_free_vmalloc_buffer(substream); 152 } 153 --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

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[alsa-devel] [PATCH 00/18] soundwire: code hardening and suspend-resume support
by Pierre-Louis Bossart 24 Oct '19

24 Oct '19

this patchset applies on top of "[PATCH 00/14] soundwire: intel: implement new ASoC interfaces". It implements a series of improvements for: a) interrupt handling on Intel platforms in MSI mode b) race conditions on codec probe and enumeration c) suspend-resume issues (clock-stop mode not supported for now) d) underflow handling e) updates to the stream state machine which did not support valid ALSA transitions. These patches were tested extensively on 4 different platforms and are viewed as required for any sort of SoundWire-based product. Bard Liao (3): soundwire: intel/cadence: fix timeouts in MSI mode soundwire: stream: only prepare stream when it is configured. soundwire: intel: reinitialize IP+DSP in .prepare() Pierre-Louis Bossart (15): soundwire: fix race between driver probe and update_status callback soundwire: bus: add PM/no-PM versions of read/write functions soundwire: bus: write Slave Device Number without runtime_pm soundwire: intel: add helpers for link power down and shim wake soundwire: intel: Add basic power management support soundwire: intel: add pm_runtime support soundwire: intel: reset pm_runtime status during system resume soundwire: bus: add helper to reset Slave status to UNATTACHED soundwire: intel: call helper to reset Slave states on resume soundwire: bus: check first if Slaves become UNATTACHED soundwire: add enumeration_complete signaling soundwire: intel: disable pm_runtime when removing a master soundwire: bus: disable pm_runtime in sdw_slave_delete soundwire: stream: update state machine and add state checks soundwire: stream: do not update parameters during DISABLED-PREPARED transition Documentation/driver-api/soundwire/stream.rst | 63 +++- drivers/soundwire/bus.c | 157 +++++++-- drivers/soundwire/bus.h | 3 + drivers/soundwire/bus_type.c | 5 + drivers/soundwire/cadence_master.c | 17 +- drivers/soundwire/cadence_master.h | 4 + drivers/soundwire/intel.c | 328 ++++++++++++++++-- drivers/soundwire/intel.h | 2 + drivers/soundwire/intel_init.c | 45 ++- drivers/soundwire/slave.c | 3 + drivers/soundwire/stream.c | 64 +++- include/linux/soundwire/sdw.h | 1 + include/linux/soundwire/sdw_intel.h | 2 + 13 files changed, 623 insertions(+), 71 deletions(-) -- 2.20.1

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Re: [alsa-devel] [PATCH] ASoC: fsl_asrc: refine the setting of internal clock divider
by S.j. Wang 23 Oct '19

23 Oct '19

Hi > > On Thu, Oct 17, 2019 at 02:21:08PM +0800, Shengjiu Wang wrote: > > For P2P output, the output divider should align with the output sample > > I think we should avoid "P2P" (or "M2M") keyword in the mainline code as > we know M2M will never get merged while somebody working with the > mainline and caring about new feature might be confused. Ok. But we still curious that is there a way to upstream m2m? > > > rate, if use ideal sample rate, there will be a lot of overload, which > > would cause underrun. > > If I understand it correctly, setting to ideal ratio provides a faster converting > speed but increases the load of the processor of ASRC. So we choose a > slower converting speed here since real- time playback mode doesn't really > need a faster conversion? Yes. Slower speed is enough for real-time playback > > It makes sense to me, yet I feel that the delay at the beginning of the audio > playback might be longer as a compromise. I am okay with this decision > though... > > > The maximum divider of asrc clock is 1024, but there is no judgement > > for this limitaion in driver, which may cause the divider setting not > > correct. > > > > For non-ideal ratio mode, the clock rate should divide the sample rate > > with no remainder, and the quotient should be less than 1024. > > > > Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> > > --- > > sound/soc/fsl/fsl_asrc.c | 40 > > +++++++++++++++++++++++++++++++--------- > > 1 file changed, 31 insertions(+), 9 deletions(-) > > > > diff --git a/sound/soc/fsl/fsl_asrc.c b/sound/soc/fsl/fsl_asrc.c index > > 0bf91a6f54b9..44d05ec28bd3 100644 > > --- a/sound/soc/fsl/fsl_asrc.c > > +++ b/sound/soc/fsl/fsl_asrc.c > > @@ -260,7 +260,7 @@ static int fsl_asrc_set_ideal_ratio(struct > fsl_asrc_pair *pair, > > * of struct asrc_config which includes in/output sample rate, width, > channel > > * and clock settings. > > */ > > -static int fsl_asrc_config_pair(struct fsl_asrc_pair *pair) > > +static int fsl_asrc_config_pair(struct fsl_asrc_pair *pair, bool p2p) > > { > > struct asrc_config *config = pair->config; > > struct fsl_asrc *asrc_priv = pair->asrc_priv; @@ -268,7 +268,8 > > @@ static int fsl_asrc_config_pair(struct fsl_asrc_pair *pair) > > enum asrc_word_width input_word_width; > > enum asrc_word_width output_word_width; > > u32 inrate, outrate, indiv, outdiv; > > - u32 clk_index[2], div[2]; > > + u32 clk_index[2], div[2], rem[2]; > > + u64 clk_rate; > > int in, out, channels; > > int pre_proc, post_proc; > > struct clk *clk; > > @@ -351,7 +352,9 @@ static int fsl_asrc_config_pair(struct fsl_asrc_pair > *pair) > > /* We only have output clock for ideal ratio mode */ > > clk = asrc_priv->asrck_clk[clk_index[ideal ? OUT : IN]]; > > > > - div[IN] = clk_get_rate(clk) / inrate; > > + clk_rate = clk_get_rate(clk); > > The fsl_asrc.c file has config.inclk being set to INCLK_NONE and this sets the > "ideal" in this function to true. So, although we tend to not use ideal ratio > setting for p2p cases, yet the input clock is still not physically connected, so > we still use output clock for div[IN] calculation? For p2p case, it can be ideal or non-ideal. For non-ideal, we still use Output clock for div calculation. > > I am thinking something simplier: if we decided not to use ideal ratio for > "P2P", instead of adding "bool p2p" with the confusing "ideal" in this > function, could we just set config.inclk to the same clock as the output one > for "P2P"? By doing so, "P2P" won't go through ideal ratio mode while still > having a clock rate from the output clock for div[IN] calculation here. Bool p2p is to force output rate to be sample rate, no impact to ideal Ratio mode. > > > + rem[IN] = do_div(clk_rate, inrate); > > + div[IN] = (u32)clk_rate; > > if (div[IN] == 0) { > > Could we check div[IN] and rem[IN] here? Like: > if (div[IN] == 0 || div[IN] > 1024) { > pair_err(); > goto out; > } > > if (!ideal && rem[IN]) { > pair_err(); > goto out; > } > > According to your commit log, I think the max-1024 limitation should be > applied to all cases, not confined to "!ideal" cases right? And we should > add some comments also, indicating it is limited by hardware. For ideal mode, my test result is the divider not impact the output result. Which means it is ok for ideal mode even divider is not correct... > > > pair_err("failed to support input sample rate %dHz by > asrck_%x\n", > > inrate, clk_index[ideal ? OUT : IN]); @@ > > -360,11 +363,20 @@ static int fsl_asrc_config_pair(struct > > fsl_asrc_pair *pair) > > > > clk = asrc_priv->asrck_clk[clk_index[OUT]]; > > > > - /* Use fixed output rate for Ideal Ratio mode (INCLK_NONE) */ > > - if (ideal) > > - div[OUT] = clk_get_rate(clk) / IDEAL_RATIO_RATE; > > - else > > - div[OUT] = clk_get_rate(clk) / outrate; > > + /* > > + * When P2P mode, output rate should align with the out samplerate. > > + * if set too high output rate, there will be lots of Overload. > > + * When M2M mode, output rate should also need to align with the > > + out > > For this "should", do you actually mean "M2M could also"? Sorry, I'm just > trying to understand everyting here, not intentionally being picky at words. > My understanding is that we still keep the ideal ratio setting because > "M2M" still uses it. We use IDEAL_RATIO_RATE as output rate for m2m mode, it likes a Tricky operation, in order to improve the performance. I think The correct operation is to use the real output rate, but the performance Is bad. So it is a compromise. > > > + * samplerate, but M2M must use less time to achieve good > performance. > > + */ > > + clk_rate = clk_get_rate(clk); > > + if (p2p || !ideal) { > > + rem[OUT] = do_div(clk_rate, outrate); > > + div[OUT] = clk_rate; > > + } else { > > + rem[OUT] = do_div(clk_rate, IDEAL_RATIO_RATE); > > + div[OUT] = clk_rate; > > + } > > > > if (div[OUT] == 0) { > > pair_err("failed to support output sample rate %dHz by > > asrck_%x\n", @@ -372,6 +384,16 @@ static int fsl_asrc_config_pair(struct > fsl_asrc_pair *pair) > > return -EINVAL; > > } > > > > + if (!ideal && (div[IN] > 1024 || div[OUT] > 1024 || > > + rem[IN] != 0 || rem[OUT] != 0)) { > > + if (!ideal && (div[IN] > 1024 || div[OUT] > 1024 || rem[IN] || > > + rem[OUT] != 0)) { > > So for ideal == true, these limitaions are not applied any more? > Remember that the "ideal" is true for "p2p == true" cases here. No, not applied. for ideal, the div don't impact the output result Even it is not accurate. > > > + pair_err("The divider can't be used for non ideal mode\n"); > > + return -EINVAL; > > + } > > + > > + /* Divider range is [1, 1024] */ > > + div[IN] = min_t(u32, 1024, div[IN]); > > + div[OUT] = min_t(u32, 1024, div[OUT]); > > Hmm, this looks like we want to allow ideal ratio cases and p2p cases to > operate any way, even if the divider wasn't within the range to get the > in/out rates from the output clock? Yes. We still allow the p2p = true, ideal = false. Note that p2p is not Equal to ideal. Best regards Wang shengjiu

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[alsa-devel] [PATCH] ASoC: kirkwood: fix external clock probe defer
by Russell King 23 Oct '19

23 Oct '19

When our call to get the external clock fails, we forget to clean up the enabled internal clock correctly. Enable the clock after we have obtained all our resources. Fixes: 84aac6c79bfd ("ASoC: kirkwood: fix loss of external clock at probe time") Signed-off-by: Russell King <rmk+kernel(a)armlinux.org.uk> --- sound/soc/kirkwood/kirkwood-i2s.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/kirkwood/kirkwood-i2s.c b/sound/soc/kirkwood/kirkwood-i2s.c index 3446a113f482..c323ae314b55 100644 --- a/sound/soc/kirkwood/kirkwood-i2s.c +++ b/sound/soc/kirkwood/kirkwood-i2s.c @@ -559,10 +559,6 @@ static int kirkwood_i2s_dev_probe(struct platform_device *pdev) return PTR_ERR(priv->clk); } - err = clk_prepare_enable(priv->clk); - if (err < 0) - return err; - priv->extclk = devm_clk_get(&pdev->dev, "extclk"); if (IS_ERR(priv->extclk)) { if (PTR_ERR(priv->extclk) == -EPROBE_DEFER) @@ -578,6 +574,10 @@ static int kirkwood_i2s_dev_probe(struct platform_device *pdev) } } + err = clk_prepare_enable(priv->clk); + if (err < 0) + return err; + /* Some sensible defaults - this reflects the powerup values */ priv->ctl_play = KIRKWOOD_PLAYCTL_SIZE_24; priv->ctl_rec = KIRKWOOD_RECCTL_SIZE_24; -- 2.20.1

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[alsa-devel] [PATCH] ASoC: mediatek: Check SND_SOC_CROS_EC_CODEC dependency
by Mao Wenan 23 Oct '19

23 Oct '19

If SND_SOC_MT8183_MT6358_TS3A227E_MAX98357A=y, below errors can be seen: sound/soc/codecs/cros_ec_codec.o: In function `send_ec_host_command': cros_ec_codec.c:(.text+0x534): undefined reference to `cros_ec_cmd_xfer_status' cros_ec_codec.c:(.text+0x101c): undefined reference to `cros_ec_get_host_event' This is because it will select SND_SOC_CROS_EC_CODEC after commit 2cc3cd5fdc8b ("ASoC: mediatek: mt8183: support WoV"), but SND_SOC_CROS_EC_CODEC depends on CROS_EC. Fixes: 2cc3cd5fdc8b ("ASoC: mediatek: mt8183: support WoV") Signed-off-by: Mao Wenan <maowenan(a)huawei.com> --- sound/soc/mediatek/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/mediatek/Kconfig b/sound/soc/mediatek/Kconfig index 8b29f39..a656d20 100644 --- a/sound/soc/mediatek/Kconfig +++ b/sound/soc/mediatek/Kconfig @@ -125,7 +125,7 @@ config SND_SOC_MT8183_MT6358_TS3A227E_MAX98357A select SND_SOC_MAX98357A select SND_SOC_BT_SCO select SND_SOC_TS3A227E - select SND_SOC_CROS_EC_CODEC + select SND_SOC_CROS_EC_CODEC if CROS_EC help This adds ASoC driver for Mediatek MT8183 boards with the MT6358 TS3A227E MAX98357A audio codec. -- 2.7.4

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[alsa-devel] [PATCH] ASoC: rsnd: add missing of_node_put()
by Kuninori Morimoto 23 Oct '19

23 Oct '19

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> This patch adds missing of_node_put() for rsnd_parse_tdm_split_mode() rsnd_parse_connect_graph() Reported-by: Pavel Machek <pavel(a)denx.de> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/sh/rcar/core.c | 9 ++++++++- 1 file changed, 8 insertions(+), 1 deletion(-) diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index fe681c5..14e86e4 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -1075,7 +1075,10 @@ static void rsnd_parse_tdm_split_mode(struct rsnd_priv *priv, j++; } + of_node_put(node); } + + of_node_put(ssiu_np); } static void rsnd_parse_connect_simple(struct rsnd_priv *priv, @@ -1093,11 +1096,13 @@ static void rsnd_parse_connect_graph(struct rsnd_priv *priv, struct device_node *endpoint) { struct device *dev = rsnd_priv_to_dev(priv); - struct device_node *remote_node = of_graph_get_remote_port_parent(endpoint); + struct device_node *remote_node; if (!rsnd_io_to_mod_ssi(io)) return; + remote_node = of_graph_get_remote_port_parent(endpoint); + /* HDMI0 */ if (strstr(remote_node->full_name, "hdmi@fead0000")) { rsnd_flags_set(io, RSND_STREAM_HDMI0); @@ -1111,6 +1116,8 @@ static void rsnd_parse_connect_graph(struct rsnd_priv *priv, } rsnd_parse_tdm_split_mode(priv, io, endpoint); + + of_node_put(remote_node); } void rsnd_parse_connect_common(struct rsnd_dai *rdai, -- 2.7.4

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[alsa-devel] [PATCH] ASoC: rsnd: dma: set bus width to data width for monaural data
by Eugeniu Rosca 23 Oct '19

23 Oct '19

From: Jiada Wang <jiada_wang(a)mentor.com> According to R-Car3 HW manual 40.3.3 (Data Format on Audio Local Bus), in case of monaural data writing or reading through Audio-DMAC, it's always in Left Justified format, so both src and dst DMA Bus width should be equal to physical data width. Therefore set src and dst's DMA bus width to: - [monaural case] data width - [non-monaural case] 32bits (as prior applying the patch) Cc: Andrew Gabbasov <andrew_gabbasov(a)mentor.com> Cc: Timo Wischer <twischer(a)de.adit-jv.com> Signed-off-by: Jiada Wang <jiada_wang(a)mentor.com> Signed-off-by: Eugeniu Rosca <erosca(a)de.adit-jv.com> --- sound/soc/sh/rcar/dma.c | 30 ++++++++++++++++++++++++++++-- 1 file changed, 28 insertions(+), 2 deletions(-) diff --git a/sound/soc/sh/rcar/dma.c b/sound/soc/sh/rcar/dma.c index 28f65eba2bb4..95aa26d62e4f 100644 --- a/sound/soc/sh/rcar/dma.c +++ b/sound/soc/sh/rcar/dma.c @@ -165,14 +165,40 @@ static int rsnd_dmaen_start(struct rsnd_mod *mod, struct device *dev = rsnd_priv_to_dev(priv); struct dma_async_tx_descriptor *desc; struct dma_slave_config cfg = {}; + enum dma_slave_buswidth buswidth = DMA_SLAVE_BUSWIDTH_4_BYTES; int is_play = rsnd_io_is_play(io); int ret; + /* + * in case of monaural data writing or reading through Audio-DMAC + * data is always in Left Justified format, so both src and dst + * DMA Bus width need to be set equal to physical data width. + */ + if (rsnd_runtime_channel_original(io) == 1) { + struct snd_pcm_runtime *runtime = rsnd_io_to_runtime(io); + int bits = snd_pcm_format_physical_width(runtime->format); + + switch (bits) { + case 8: + buswidth = DMA_SLAVE_BUSWIDTH_1_BYTE; + break; + case 16: + buswidth = DMA_SLAVE_BUSWIDTH_2_BYTES; + break; + case 32: + buswidth = DMA_SLAVE_BUSWIDTH_4_BYTES; + break; + default: + dev_err(dev, "invalid format width %d\n", bits); + return -EINVAL; + } + } + cfg.direction = is_play ? DMA_MEM_TO_DEV : DMA_DEV_TO_MEM; cfg.src_addr = dma->src_addr; cfg.dst_addr = dma->dst_addr; - cfg.src_addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES; - cfg.dst_addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES; + cfg.src_addr_width = buswidth; + cfg.dst_addr_width = buswidth; dev_dbg(dev, "%s %pad -> %pad\n", rsnd_mod_name(mod), -- 2.23.0

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