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2019-11-10ASoc: rockchip: i2s: Fix RPM imbalanceRobin Murphy
[ Upstream commit b1e620e7d32f5aad5353cc3cfc13ed99fea65d3a ] If rockchip_pcm_platform_register() fails, e.g. upon deferring to wait for an absent DMA channel, we return without disabling RPM, which makes subsequent re-probe attempts scream with errors about the unbalanced enable. Don't do that. Fixes: ebb75c0bdba2 ("ASoC: rockchip: i2s: Adjust devm usage") Signed-off-by: Robin Murphy <robin.murphy@arm.com> Link: https://lore.kernel.org/r/bcb12a849a05437fb18372bc7536c649b94bdf07.1570029862.git.robin.murphy@arm.com Signed-off-by: Mark Brown <broonie@kernel.org> Signed-off-by: Sasha Levin <sashal@kernel.org>
2019-11-10ASoC: wm_adsp: Don't generate kcontrols without READ flagsStuart Henderson
[ Upstream commit 3ae7359c0e39f42a96284d6798fc669acff38140 ] User space always expects to be able to read ALSA controls, so ensure no kcontrols are generated without an appropriate READ flag. In the case of a read of such a control zeros will be returned. Signed-off-by: Stuart Henderson <stuarth@opensource.cirrus.com> Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com> Link: https://lore.kernel.org/r/20191002084240.21589-1-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown <broonie@kernel.org> Signed-off-by: Sasha Levin <sashal@kernel.org>
2019-10-29ASoC: rsnd: Reinitialize bit clock inversion flag for every format settingJunya Monden
commit 22e58665a01006d05f0239621f7d41cacca96cc4 upstream. Unlike other format-related DAI parameters, rdai->bit_clk_inv flag is not properly re-initialized when setting format for new stream processing. The inversion, if requested, is then applied not to default, but to a previous value, which leads to SCKP bit in SSICR register being set incorrectly. Fix this by re-setting the flag to its initial value, determined by format. Fixes: 1a7889ca8aba3 ("ASoC: rsnd: fixup SND_SOC_DAIFMT_xB_xF behavior") Cc: Andrew Gabbasov <andrew_gabbasov@mentor.com> Cc: Jiada Wang <jiada_wang@mentor.com> Cc: Timo Wischer <twischer@de.adit-jv.com> Cc: stable@vger.kernel.org # v3.17+ Signed-off-by: Junya Monden <jmonden@jp.adit-jv.com> Signed-off-by: Eugeniu Rosca <erosca@de.adit-jv.com> Acked-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Link: https://lore.kernel.org/r/20191016124255.7442-1-erosca@de.adit-jv.com Signed-off-by: Mark Brown <broonie@kernel.org> Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
2019-10-11ASoC: sgtl5000: Improve VAG power and mute controlOleksandr Suvorov
[ Upstream commit b1f373a11d25fc9a5f7679c9b85799fe09b0dc4a ] VAG power control is improved to fit the manual [1]. This patch fixes as minimum one bug: if customer muxes Headphone to Line-In right after boot, the VAG power remains off that leads to poor sound quality from line-in. I.e. after boot: - Connect sound source to Line-In jack; - Connect headphone to HP jack; - Run following commands: $ amixer set 'Headphone' 80% $ amixer set 'Headphone Mux' LINE_IN Change VAG power on/off control according to the following algorithm: - turn VAG power ON on the 1st incoming event. - keep it ON if there is any active VAG consumer (ADC/DAC/HP/Line-In). - turn VAG power OFF when there is the latest consumer's pre-down event come. - always delay after VAG power OFF to avoid pop. - delay after VAG power ON if the initiative consumer is Line-In, this prevents pop during line-in muxing. According to the data sheet [1], to avoid any pops/clicks, the outputs should be muted during input/output routing changes. [1] https://www.nxp.com/docs/en/data-sheet/SGTL5000.pdf Cc: stable@vger.kernel.org Fixes: 9b34e6cc3bc2 ("ASoC: Add Freescale SGTL5000 codec support") Signed-off-by: Oleksandr Suvorov <oleksandr.suvorov@toradex.com> Reviewed-by: Marcel Ziswiler <marcel.ziswiler@toradex.com> Reviewed-by: Fabio Estevam <festevam@gmail.com> Reviewed-by: Cezary Rojewski <cezary.rojewski@intel.com> Link: https://lore.kernel.org/r/20190719100524.23300-3-oleksandr.suvorov@toradex.com Signed-off-by: Mark Brown <broonie@kernel.org> Signed-off-by: Sasha Levin <sashal@kernel.org>
2019-10-05ASoC: Intel: Fix use of potentially uninitialized variableAmadeusz Sławiński
commit 810f3b860850148788fc1ed8a6f5f807199fed65 upstream. If ipc->ops.reply_msg_match is NULL, we may end up using uninitialized mask value. reported by smatch: sound/soc/intel/common/sst-ipc.c:266 sst_ipc_reply_find_msg() error: uninitialized symbol 'mask'. Signed-off-by: Amadeusz Sławiński <amadeuszx.slawinski@intel.com> Link: https://lore.kernel.org/r/20190827141712.21015-3-amadeuszx.slawinski@linux.intel.com Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Signed-off-by: Mark Brown <broonie@kernel.org> Cc: stable@vger.kernel.org Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
2019-10-05ASoC: Intel: Skylake: Use correct function to access iomem spaceAmadeusz Sławiński
commit 17d29ff98fd4b70e9ccdac5e95e18a087e2737ef upstream. For copying from __iomem, we should use __ioread32_copy. reported by sparse: sound/soc/intel/skylake/skl-debug.c:437:34: warning: incorrect type in argument 1 (different address spaces) sound/soc/intel/skylake/skl-debug.c:437:34: expected void [noderef] <asn:2> *to sound/soc/intel/skylake/skl-debug.c:437:34: got unsigned char * sound/soc/intel/skylake/skl-debug.c:437:51: warning: incorrect type in argument 2 (different address spaces) sound/soc/intel/skylake/skl-debug.c:437:51: expected void const *from sound/soc/intel/skylake/skl-debug.c:437:51: got void [noderef] <asn:2> *[assigned] fw_reg_addr Signed-off-by: Amadeusz Sławiński <amadeuszx.slawinski@intel.com> Link: https://lore.kernel.org/r/20190827141712.21015-2-amadeuszx.slawinski@linux.intel.com Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Signed-off-by: Mark Brown <broonie@kernel.org> Cc: stable@vger.kernel.org Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
2019-10-05ASoC: Intel: NHLT: Fix debug print formatAmadeusz Sławiński
commit 855a06da37a773fd073d51023ac9d07988c87da8 upstream. oem_table_id is 8 chars long, so we need to limit it, otherwise it may print some unprintable characters into dmesg. Signed-off-by: Amadeusz Sławiński <amadeuszx.slawinski@intel.com> Link: https://lore.kernel.org/r/20190827141712.21015-7-amadeuszx.slawinski@linux.intel.com Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Signed-off-by: Mark Brown <broonie@kernel.org> Cc: stable@vger.kernel.org Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
2019-10-05ASoC: dmaengine: Make the pcm->name equal to pcm->id if the name is not setPeter Ujfalusi
[ Upstream commit 2ec42f3147e1610716f184b02e65d7f493eed925 ] Some tools use the snd_pcm_info_get_name() to try to identify PCMs or for other purposes. Currently it is left empty with the dmaengine-pcm, in this case copy the pcm->id string as pcm->name. For example IGT is using this to find the HDMI PCM for testing audio on it. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com> Reported-by: Arthur She <arthur.she@linaro.org> Link: https://lore.kernel.org/r/20190906055524.7393-1-peter.ujfalusi@ti.com Signed-off-by: Mark Brown <broonie@kernel.org> Signed-off-by: Sasha Levin <sashal@kernel.org>
2019-10-05ASoC: es8316: fix headphone mixer volume tableKatsuhiro Suzuki
[ Upstream commit f972d02fee2496024cfd6f59021c9d89d54922a6 ] This patch fix setting table of Headphone mixer volume. Current code uses 4 ... 7 values but these values are prohibited. Correct settings are the following: 0000 -12dB 0001 -10.5dB 0010 -9dB 0011 -7.5dB 0100 -6dB 1000 -4.5dB 1001 -3dB 1010 -1.5dB 1011 0dB Signed-off-by: Katsuhiro Suzuki <katsuhiro@katsuster.net> Reviewed-by: Daniel Drake <drake@endlessm.com> Link: https://lore.kernel.org/r/20190826153900.25969-1-katsuhiro@katsuster.net Signed-off-by: Mark Brown <broonie@kernel.org> Signed-off-by: Sasha Levin <sashal@kernel.org>
2019-10-05ASoC: rsnd: don't call clk_get_rate() under atomic contextKuninori Morimoto
[ Upstream commit 06e8f5c842f2dbb232897ba967ea7b422745c271 ] ADG is using clk_get_rate() under atomic context, thus, we might have scheduling issue. To avoid this issue, we need to get/keep clk rate under non atomic context. We need to handle ADG as special device at Renesas Sound driver. From SW point of view, we want to impletent it as rsnd_mod_ops :: prepare, but it makes code just complicate. To avoid complicated code/patch, this patch adds new clk_rate[] array, and keep clk IN rate when rsnd_adg_clk_enable() was called. Reported-by: Leon Kong <Leon.KONG@cn.bosch.com> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Tested-by: Leon Kong <Leon.KONG@cn.bosch.com> Link: https://lore.kernel.org/r/87v9vb0xkp.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown <broonie@kernel.org> Signed-off-by: Sasha Levin <sashal@kernel.org>
2019-10-05ASoC: sgtl5000: Fix charge pump source assignmentOleksandr Suvorov
[ Upstream commit b6319b061ba279577fd7030a9848fbd6a17151e3 ] If VDDA != VDDIO and any of them is greater than 3.1V, charge pump source can be assigned automatically [1]. [1] https://www.nxp.com/docs/en/data-sheet/SGTL5000.pdf Signed-off-by: Oleksandr Suvorov <oleksandr.suvorov@toradex.com> Reviewed-by: Marcel Ziswiler <marcel.ziswiler@toradex.com> Reviewed-by: Igor Opaniuk <igor.opaniuk@toradex.com> Reviewed-by: Fabio Estevam <festevam@gmail.com> Link: https://lore.kernel.org/r/20190719100524.23300-7-oleksandr.suvorov@toradex.com Signed-off-by: Mark Brown <broonie@kernel.org> Signed-off-by: Sasha Levin <sashal@kernel.org>
2019-10-05ASoC: fsl: Fix of-node refcount unbalance in fsl_ssi_probe_from_dt()Takashi Iwai
commit 2757970f6d0d0a112247600b23d38c0c728ceeb3 upstream. The node obtained from of_find_node_by_path() has to be unreferenced after the use, but we forgot it for the root node. Fixes: f0fba2ad1b6b ("ASoC: multi-component - ASoC Multi-Component Support") Cc: Timur Tabi <timur@kernel.org> Cc: Nicolin Chen <nicoleotsuka@gmail.com> Cc: Xiubo Li <Xiubo.Lee@gmail.com> Cc: Fabio Estevam <festevam@gmail.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Acked-by: Nicolin Chen <nicoleotsuka@gmail.com> Signed-off-by: Mark Brown <broonie@kernel.org> Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
2019-09-06Revert "ASoC: Fail card instantiation if DAI format setup fails"Greg Kroah-Hartman
This reverts commit a56c79efe4dc8ae560eb2c9a4b755d57c8c2e8f4 which is commit 40aa5383e393d72f6aa3943a4e7b1aae25a1e43b upstream. Mark Brown writes: I nacked this patch when Sasha posted it - it only improves diagnostics and might make systems that worked by accident break since it turns things into a hard failure, it won't make anything that didn't work previously work. Reported-by: Mark Brown <broonie@kernel.org> Cc: Ricard Wanderlof <ricardw@axis.com> Cc: Sasha Levin <sashal@kernel.org> Link: https://lore.kernel.org/lkml/20190904181027.GG4348@sirena.co.uk Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
2019-08-29ASoC: ti: davinci-mcasp: Correct slot_width posed constraintPeter Ujfalusi
[ Upstream commit 1e112c35e3c96db7c8ca6ddaa96574f00c06e7db ] The slot_width is a property for the bus while the constraint for SNDRV_PCM_HW_PARAM_SAMPLE_BITS is for the in memory format. Applying slot_width constraint to sample_bits works most of the time, but it will blacklist valid formats in some cases. With slot_width 24 we can support S24_3LE and S24_LE formats as they both look the same on the bus, but a a 24 constraint on sample_bits would not allow S24_LE as it is stored in 32bits in memory. Implement a simple hw_rule function to allow all formats which require less or equal number of bits on the bus as slot_width (if configured). Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com> Link: https://lore.kernel.org/r/20190726064244.3762-2-peter.ujfalusi@ti.com Signed-off-by: Mark Brown <broonie@kernel.org> Signed-off-by: Sasha Levin <sashal@kernel.org>
2019-08-29ASoC: Fail card instantiation if DAI format setup failsRicard Wanderlof
[ Upstream commit 40aa5383e393d72f6aa3943a4e7b1aae25a1e43b ] If the DAI format setup fails, there is no valid communication format between CPU and CODEC, so fail card instantiation, rather than continue with a card that will most likely not function properly. Signed-off-by: Ricard Wanderlof <ricardw@axis.com> Link: https://lore.kernel.org/r/alpine.DEB.2.20.1907241132350.6338@lnxricardw1.se.axis.com Signed-off-by: Mark Brown <broonie@kernel.org> Signed-off-by: Sasha Levin <sashal@kernel.org>
2019-08-29ASoC: dapm: Fix handling of custom_stop_condition on DAPM graph walksCharles Keepax
[ Upstream commit 8dd26dff00c0636b1d8621acaeef3f6f3a39dd77 ] DPCM uses snd_soc_dapm_dai_get_connected_widgets to build a list of the widgets connected to a specific front end DAI so it can search through this list for available back end DAIs. The custom_stop_condition was added to is_connected_ep to facilitate this list not containing more widgets than is necessary. Doing so both speeds up the DPCM handling as less widgets need to be searched and avoids issues with CODEC to CODEC links as these would be confused with back end DAIs if they appeared in the list of available widgets. custom_stop_condition was implemented by aborting the graph walk when the condition is triggered, however there is an issue with this approach. Whilst walking the graph is_connected_ep should update the endpoints cache on each widget, if the walk is aborted the number of attached end points is unknown for that sub-graph. When the stop condition triggered, the original patch ignored the triggering widget and returned zero connected end points; a later patch updated this to set the triggering widget's cache to 1 and return that. Both of these approaches result in inaccurate values being stored in various end point caches as the values propagate back through the graph, which can result in later issues with widgets powering/not powering unexpectedly. As the original goal was to reduce the size of the widget list passed to the DPCM code, the simplest solution is to limit the functionality of the custom_stop_condition to the widget list. This means the rest of the graph will still be processed resulting in correct end point caches, but only widgets up to the stop condition will be added to the returned widget list. Fixes: 6742064aef7f ("ASoC: dapm: support user-defined stop condition in dai_get_connected_widgets") Fixes: 5fdd022c2026 ("ASoC: dpcm: play nice with CODEC<->CODEC links") Fixes: 09464974eaa8 ("ASoC: dapm: Fix to return correct path list in is_connected_ep.") Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com> Link: https://lore.kernel.org/r/20190718084333.15598-1-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown <broonie@kernel.org> Signed-off-by: Sasha Levin <sashal@kernel.org>
2019-07-31ASoC: dapm: Adapt for debugfs API changeMark Brown
commit ceaea851b9ea75f9ea2bbefb53ff0d4b27cd5a6e upstream. Back in ff9fb72bc07705c (debugfs: return error values, not NULL) the debugfs APIs were changed to return error pointers rather than NULL pointers on error, breaking the error checking in ASoC. Update the code to use IS_ERR() and log the codes that are returned as part of the error messages. Fixes: ff9fb72bc07705c (debugfs: return error values, not NULL) Signed-off-by: Mark Brown <broonie@kernel.org> Cc: stable@vger.kernel.org Signed-off-by: Mark Brown <broonie@kernel.org> Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
2019-07-10SoC: rt274: Fix internal jack assignment in set_jack callbackAmadeusz Sławiński
[ Upstream commit 04268bf2757a125616b6c2140e6250f43b7b737a ] When we call snd_soc_component_set_jack(component, NULL, NULL) we should set rt274->jack to passed jack, so when interrupt is triggered it calls snd_soc_jack_report(rt274->jack, ...) with proper value. This fixes problem in machine where in register, we call snd_soc_register(component, &headset, NULL), which just calls rt274_mic_detect via callback. Now when machine driver is removed "headset" will be gone, so we need to tell codec driver that it's gone with: snd_soc_register(component, NULL, NULL), but we also need to be able to handle NULL jack argument here gracefully. If we don't set it to NULL, next time the rt274_irq runs it will call snd_soc_jack_report with first argument being invalid pointer and there will be Oops. Signed-off-by: Amadeusz Sławiński <amadeuszx.slawinski@linux.intel.com> Signed-off-by: Mark Brown <broonie@kernel.org> Signed-off-by: Sasha Levin <sashal@kernel.org>
2019-07-10ASoC: sun4i-i2s: Add offset to RX channel selectMarcus Cooper
[ Upstream commit f9927000cb35f250051f0f1878db12ee2626eea1 ] Whilst testing the capture functionality of the i2s on the newer SoCs it was noticed that the recording was somewhat distorted. This was due to the offset not being set correctly on the receiver side. Signed-off-by: Marcus Cooper <codekipper@gmail.com> Acked-by: Maxime Ripard <maxime.ripard@bootlin.com> Acked-by: Chen-Yu Tsai <wens@csie.org> Signed-off-by: Mark Brown <broonie@kernel.org> Signed-off-by: Sasha Levin <sashal@kernel.org>
2019-07-10ASoC: sun4i-i2s: Fix sun8i tx channel offset maskMarcus Cooper
[ Upstream commit 7e46169a5f35762f335898a75d1b8a242f2ae0f5 ] Although not causing any noticeable issues, the mask for the channel offset is covering too many bits. Signed-off-by: Marcus Cooper <codekipper@gmail.com> Acked-by: Maxime Ripard <maxime.ripard@bootlin.com> Acked-by: Chen-Yu Tsai <wens@csie.org> Signed-off-by: Mark Brown <broonie@kernel.org> Signed-off-by: Sasha Levin <sashal@kernel.org>
2019-07-10ASoC: max98090: remove 24-bit format support if RJ is 0Yu-Hsuan Hsu
[ Upstream commit 5628c8979642a076f91ee86c3bae5ad251639af0 ] The supported formats are S16_LE and S24_LE now. However, by datasheet of max98090, S24_LE is only supported when it is in the right justified mode. We should remove 24-bit format if it is not in that mode to avoid triggering error. Signed-off-by: Yu-Hsuan Hsu <yuhsuan@chromium.org> Signed-off-by: Mark Brown <broonie@kernel.org> Signed-off-by: Sasha Levin <sashal@kernel.org>
2019-07-10ASoC: soc-pcm: BE dai needs prepare when pause release after resumeLibin Yang
[ Upstream commit 5087a8f17df868601cd7568299e91c28086d2b45 ] If playback/capture is paused and system enters S3, after system returns from suspend, BE dai needs to call prepare() callback when playback/capture is released from pause if RESUME_INFO flag is not set. Currently, the dpcm_be_dai_prepare() function will block calling prepare() if the pcm is in SND_SOC_DPCM_STATE_PAUSED state. This will cause the following test case fail if the pcm uses BE: playback -> pause -> S3 suspend -> S3 resume -> pause release The playback may exit abnormally when pause is released because the BE dai prepare() is not called. This patch allows dpcm_be_dai_prepare() to call dai prepare() callback in SND_SOC_DPCM_STATE_PAUSED state. Signed-off-by: Libin Yang <libin.yang@intel.com> Signed-off-by: Mark Brown <broonie@kernel.org> Signed-off-by: Sasha Levin <sashal@kernel.org>
2019-07-10ASoC : cs4265 : readable register too lowMatt Flax
[ Upstream commit f3df05c805983427319eddc2411a2105ee1757cf ] The cs4265_readable_register function stopped short of the maximum register. An example bug is taken from : https://github.com/Audio-Injector/Ultra/issues/25 Where alsactl store fails with : Cannot read control '2,0,0,C Data Buffer,0': Input/output error This patch fixes the bug by setting the cs4265 to have readable registers up to the maximum hardware register CS4265_MAX_REGISTER. Signed-off-by: Matt Flax <flatmax@flatmax.org> Reviewed-by: Charles Keepax <ckeepax@opensource.cirrus.com> Signed-off-by: Mark Brown <broonie@kernel.org> Signed-off-by: Sasha Levin <sashal@kernel.org>
2019-06-19ASoC: fsl_asrc: Fix the issue about unsupported rateS.j. Wang
commit b06c58c2a1eed571ea2a6640fdb85b7b00196b1e upstream. When the output sample rate is [8kHz, 30kHz], the limitation of the supported ratio range is [1/24, 8]. In the driver we use (8kHz, 30kHz) instead of [8kHz, 30kHz]. So this patch is to fix this issue and the potential rounding issue with divider. Fixes: fff6e03c7b65 ("ASoC: fsl_asrc: add support for 8-30kHz output sample rate") Cc: <stable@vger.kernel.org> Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com> Acked-by: Nicolin Chen <nicoleotsuka@gmail.com> Signed-off-by: Mark Brown <broonie@kernel.org> Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
2019-06-19ASoC: cs42xx8: Add regcache mask dirtyS.j. Wang
commit ad6eecbfc01c987e0253371f274c3872042e4350 upstream. Add regcache_mark_dirty before regcache_sync for power of codec may be lost at suspend, then all the register need to be reconfigured. Fixes: 0c516b4ff85c ("ASoC: cs42xx8: Add codec driver support for CS42448/CS42888") Cc: <stable@vger.kernel.org> Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com> Signed-off-by: Mark Brown <broonie@kernel.org> Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
2019-05-31ASoC: davinci-mcasp: Fix clang warning without CONFIG_PMArnd Bergmann
[ Upstream commit 8ca5104715cfd14254ea5aecc390ae583b707607 ] Building with clang shows a variable that is only used by the suspend/resume functions but defined outside of their #ifdef block: sound/soc/ti/davinci-mcasp.c:48:12: error: variable 'context_regs' is not needed and will not be emitted We commonly fix these by marking the PM functions as __maybe_unused, but here that would grow the davinci_mcasp structure, so instead add another #ifdef here. Fixes: 1cc0c054f380 ("ASoC: davinci-mcasp: Convert the context save/restore to use array") Signed-off-by: Arnd Bergmann <arnd@arndb.de> Acked-by: Peter Ujfalusi <peter.ujfalusi@ti.com> Reviewed-by: Nathan Chancellor <natechancellor@gmail.com> Signed-off-by: Mark Brown <broonie@kernel.org> Signed-off-by: Sasha Levin <sashal@kernel.org>
2019-05-31ASoC: fsl_utils: fix a leaked reference by adding missing of_node_putWen Yang
[ Upstream commit c705247136a523488eac806bd357c3e5d79a7acd ] The call to of_parse_phandle returns a node pointer with refcount incremented thus it must be explicitly decremented after the last usage. Detected by coccinelle with the following warnings: ./sound/soc/fsl/fsl_utils.c:74:2-8: ERROR: missing of_node_put; acquired a node pointer with refcount incremented on line 38, but without a corresponding object release within this function. Signed-off-by: Wen Yang <wen.yang99@zte.com.cn> Cc: Timur Tabi <timur@kernel.org> Cc: Nicolin Chen <nicoleotsuka@gmail.com> Cc: Xiubo Li <Xiubo.Lee@gmail.com> Cc: Fabio Estevam <festevam@gmail.com> Cc: Liam Girdwood <lgirdwood@gmail.com> Cc: Mark Brown <broonie@kernel.org> Cc: Jaroslav Kysela <perex@perex.cz> Cc: Takashi Iwai <tiwai@suse.com> Cc: alsa-devel@alsa-project.org Cc: linuxppc-dev@lists.ozlabs.org Cc: linux-kernel@vger.kernel.org Signed-off-by: Mark Brown <broonie@kernel.org> Signed-off-by: Sasha Levin <sashal@kernel.org>
2019-05-31ASoC: eukrea-tlv320: fix a leaked reference by adding missing of_node_putWen Yang
[ Upstream commit b820d52e7eed7b30b2dfef5f4213a2bc3cbea6f3 ] The call to of_parse_phandle returns a node pointer with refcount incremented thus it must be explicitly decremented after the last usage. Detected by coccinelle with the following warnings: ./sound/soc/fsl/eukrea-tlv320.c:121:3-9: ERROR: missing of_node_put; acquired a node pointer with refcount incremented on line 102, but without a correspo nding object release within this function. ./sound/soc/fsl/eukrea-tlv320.c:127:3-9: ERROR: missing of_node_put; acquired a node pointer with refcount incremented on line 102, but without a correspo nding object release within this function. Signed-off-by: Wen Yang <wen.yang99@zte.com.cn> Cc: Liam Girdwood <lgirdwood@gmail.com> Cc: Mark Brown <broonie@kernel.org> Cc: Jaroslav Kysela <perex@perex.cz> Cc: Takashi Iwai <tiwai@suse.com> Cc: alsa-devel@alsa-project.org Cc: linux-kernel@vger.kernel.org Signed-off-by: Mark Brown <broonie@kernel.org> Signed-off-by: Sasha Levin <sashal@kernel.org>
2019-05-31ASoC: fsl_sai: Update is_slave_mode with correct valueDaniel Baluta
[ Upstream commit ddb351145a967ee791a0fb0156852ec2fcb746ba ] is_slave_mode defaults to false because sai structure that contains it is kzalloc'ed. Anyhow, if we decide to set the following configuration SAI slave -> SAI master, is_slave_mode will remain set on true although SAI being master it should be set to false. Fix this by updating is_slave_mode for each call of fsl_sai_set_dai_fmt. Signed-off-by: Daniel Baluta <daniel.baluta@nxp.com> Acked-by: Nicolin Chen <nicoleotsuka@gmail.com> Signed-off-by: Mark Brown <broonie@kernel.org> Signed-off-by: Sasha Levin <sashal@kernel.org>
2019-05-31ASoC: imx: fix fiq dependenciesArnd Bergmann
[ Upstream commit ea751227c813ab833609afecfeedaf0aa26f327e ] During randconfig builds, I occasionally run into an invalid configuration of the freescale FIQ sound support: WARNING: unmet direct dependencies detected for SND_SOC_IMX_PCM_FIQ Depends on [m]: SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && SND_IMX_SOC [=m] Selected by [y]: - SND_SOC_FSL_SPDIF [=y] && SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && SND_IMX_SOC [=m]!=n && (MXC_TZIC [=n] || MXC_AVIC [=y]) sound/soc/fsl/imx-ssi.o: In function `imx_ssi_remove': imx-ssi.c:(.text+0x28): undefined reference to `imx_pcm_fiq_exit' sound/soc/fsl/imx-ssi.o: In function `imx_ssi_probe': imx-ssi.c:(.text+0xa64): undefined reference to `imx_pcm_fiq_init' The Kconfig warning is a result of the symbol being defined inside of the "if SND_IMX_SOC" block, and is otherwise harmless. The link error is more tricky and happens with SND_SOC_IMX_SSI=y, which may or may not imply FIQ support. However, if SND_SOC_FSL_SSI is set to =m at the same time, that selects SND_SOC_IMX_PCM_FIQ as a loadable module dependency, which then causes a link failure from imx-ssi. The solution here is to make SND_SOC_IMX_PCM_FIQ built-in whenever one of its potential users is built-in. Fixes: ff40260f79dc ("ASoC: fsl: refine DMA/FIQ dependencies") Signed-off-by: Arnd Bergmann <arnd@arndb.de> Signed-off-by: Mark Brown <broonie@kernel.org> Signed-off-by: Sasha Levin <sashal@kernel.org>
2019-05-31ASoC: hdmi-codec: unlock the device on startup errorsJerome Brunet
[ Upstream commit 30180e8436046344b12813dc954b2e01dfdcd22d ] If the hdmi codec startup fails, it should clear the current_substream pointer to free the device. This is properly done for the audio_startup() callback but for snd_pcm_hw_constraint_eld(). Make sure the pointer cleared if an error is reported. Signed-off-by: Jerome Brunet <jbrunet@baylibre.com> Signed-off-by: Mark Brown <broonie@kernel.org> Signed-off-by: Sasha Levin <sashal@kernel.org>
2019-05-21ASoC: RT5677-SPI: Disable 16Bit SPI TransfersCurtis Malainey
commit a46eb523220e242affb9a6bc9bb8efc05f4f7459 upstream. The current algorithm allows 3 types of transfers, 16bit, 32bit and burst. According to Realtek, 16bit transfers have a special restriction in that it is restricted to the memory region of 0x18020000 ~ 0x18021000. This region is the memory location of the I2C registers. The current algorithm does not uphold this restriction and therefore fails to complete writes. Since this has been broken for some time it likely no one is using it. Better to simply disable the 16 bit writes. This will allow users to properly load firmware over SPI without data corruption. Signed-off-by: Curtis Malainey <cujomalainey@chromium.org> Reviewed-by: Ben Zhang <benzh@chromium.org> Signed-off-by: Mark Brown <broonie@kernel.org> Cc: stable@vger.kernel.org Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
2019-05-21ASoC: max98090: Fix restore of DAPM MuxesJon Hunter
commit ecb2795c08bc825ebd604997e5be440b060c5b18 upstream. The max98090 driver defines 3 DAPM muxes; one for the right line output (LINMOD Mux), one for the left headphone mixer source (MIXHPLSEL Mux) and one for the right headphone mixer source (MIXHPRSEL Mux). The same bit is used for the mux as well as the DAPM enable, and although the mux can be correctly configured, after playback has completed, the mux will be reset during the disable phase. This is preventing the state of these muxes from being saved and restored correctly on system reboot. Fix this by marking these muxes as SND_SOC_NOPM. Note this has been verified this on the Tegra124 Nyan Big which features the MAX98090 codec. Signed-off-by: Jon Hunter <jonathanh@nvidia.com> Signed-off-by: Mark Brown <broonie@kernel.org> Cc: stable@vger.kernel.org Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
2019-05-10ASoC: Intel: avoid Oops if DMA setup failsRoss Zwisler
commit 0efa3334d65b7f421ba12382dfa58f6ff5bf83c4 upstream. Currently in sst_dsp_new() if we get an error return from sst_dma_new() we just print an error message and then still complete the function successfully. This means that we are trying to run without sst->dma properly set up, which will result in NULL pointer dereference when sst->dma is later used. This was happening for me in sst_dsp_dma_get_channel(): struct sst_dma *dma = dsp->dma; ... dma->ch = dma_request_channel(mask, dma_chan_filter, dsp); This resulted in: BUG: unable to handle kernel NULL pointer dereference at 0000000000000018 IP: sst_dsp_dma_get_channel+0x4f/0x125 [snd_soc_sst_firmware] Fix this by adding proper error handling for the case where we fail to set up DMA. This change only affects Haswell and Broadwell systems. Baytrail systems explicilty opt-out of DMA via sst->pdata->resindex_dma_base being set to -1. Signed-off-by: Ross Zwisler <zwisler@google.com> Cc: stable@vger.kernel.org Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Signed-off-by: Mark Brown <broonie@kernel.org> Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
2019-05-10ASoC: Intel: kbl: fix wrong number of channelsTzung-Bi Shih
[ Upstream commit d6ba3f815bc5f3c4249d15c8bc5fbb012651b4a4 ] Fix wrong setting on number of channels. The context wants to set constraint to 2 channels instead of 4. Signed-off-by: Tzung-Bi Shih <tzungbi@google.com> Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Signed-off-by: Mark Brown <broonie@kernel.org> Signed-off-by: Sasha Levin <sashal@kernel.org>
2019-05-10ASoC: rockchip: pdm: fix regmap_ops hang issueSugar Zhang
[ Upstream commit c85064435fe7a216ec0f0238ef2b8f7cd850a450 ] This is because set_fmt ops maybe called when PD is off, and in such case, regmap_ops will lead system hang. enale PD before doing regmap_ops. Signed-off-by: Sugar Zhang <sugar.zhang@rock-chips.com> Signed-off-by: Mark Brown <broonie@kernel.org> Signed-off-by: Sasha Levin <sashal@kernel.org>
2019-05-10ASoC: tlv320aic32x4: Fix Common PinsAnnaliese McDermond
[ Upstream commit c63adb28f6d913310430f14c69f0a2ea55eed0cc ] The common pins were mistakenly not added to the DAPM graph. Adding these pins will allow valid graphs to be created. Signed-off-by: Annaliese McDermond <nh6z@nh6z.net> Signed-off-by: Mark Brown <broonie@kernel.org> Signed-off-by: Sasha Levin <sashal@kernel.org>
2019-05-10ASoC: cs4270: Set auto-increment bit for register writesDaniel Mack
[ Upstream commit f0f2338a9cfaf71db895fa989ea7234e8a9b471d ] The CS4270 does not by default increment the register address on consecutive writes. During normal operation it doesn't matter as all register accesses are done individually. At resume time after suspend, however, the regcache code gathers the biggest possible block of registers to sync and sends them one on one go. To fix this, set the INCR bit in all cases. Signed-off-by: Daniel Mack <daniel@zonque.org> Signed-off-by: Mark Brown <broonie@kernel.org> Signed-off-by: Sasha Levin <sashal@kernel.org>
2019-05-10ASoC: wm_adsp: Add locking to wm_adsp2_bus_errorCharles Keepax
[ Upstream commit a2225a6d155fcb247fe4c6d87f7c91807462966d ] Best to lock across handling the bus error to ensure the DSP doesn't change power state as we are reading the status registers. Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com> Signed-off-by: Mark Brown <broonie@kernel.org> Signed-off-by: Sasha Levin <sashal@kernel.org>
2019-05-10ASoC: samsung: odroid: Fix clock configuration for 44100 sample rateSylwester Nawrocki
[ Upstream commit 2b13bee3884926cba22061efa75bd315e871de24 ] After commit fbeec965b8d1c ("ASoC: samsung: odroid: Fix 32000 sample rate handling") the audio root clock frequency is configured improperly for 44100 sample rate. Due to clock rate rounding it's 20070401 Hz instead of 22579000 Hz. This results in a too low value of the PSR clock divider in the CPU DAI driver and too fast actual sample rate for fs=44100. E.g. 1 kHz tone has actual 1780 Hz frequency (1 kHz * 20070401/22579000 * 2). Fix this by increasing the correction passed to clk_set_rate() to take into account inaccuracy of the EPLL frequency properly. Fixes: fbeec965b8d1c ("ASoC: samsung: odroid: Fix 32000 sample rate handling") Reported-by: JaeChul Lee <jcsing.lee@samsung.com> Signed-off-by: Sylwester Nawrocki <s.nawrocki@samsung.com> Signed-off-by: Mark Brown <broonie@kernel.org> Signed-off-by: Sasha Levin <sashal@kernel.org>
2019-05-10ASoC: nau8810: fix the issue of widget with prefixed nameJohn Hsu
[ Upstream commit 54d1cf78b0f4ba348a7c7fb8b7d0708d71b6cc8a ] The driver changes the stream name of DAC and ADC to avoid the issue of widget with prefixed name. When the machine adds prefixed name for codec, the stream name of DAI may not find the widgets. Signed-off-by: John Hsu <KCHSU0@nuvoton.com> Signed-off-by: Mark Brown <broonie@kernel.org> Signed-off-by: Sasha Levin <sashal@kernel.org>
2019-05-10ASoC: nau8824: fix the issue of the widget with prefix nameJohn Hsu
[ Upstream commit 844a4a362dbec166b44d6b9b3dd45b08cb273703 ] The driver has two issues when machine add prefix name for codec. (1)The stream name of DAI can't find the AIF widgets. (2)The drivr can enable/disalbe the MICBIAS and SAR widgets. The patch will fix these issues caused by prefixed name added. Signed-off-by: John Hsu <KCHSU0@nuvoton.com> Signed-off-by: Mark Brown <broonie@kernel.org> Signed-off-by: Sasha Levin <sashal@kernel.org>
2019-05-10ASoC:soc-pcm:fix a codec fixup issue in TDM caseRander Wang
[ Upstream commit 570f18b6a8d1f0e60e8caf30e66161b6438dcc91 ] On HDaudio platforms, if playback is started when capture is working, there is no audible output. This can be root-caused to the use of the rx|tx_mask to store an HDaudio stream tag. If capture is stared before playback, rx_mask would be non-zero on HDaudio platform, then the channel number of playback, which is in the same codec dai with the capture, would be changed by soc_pcm_codec_params_fixup based on the tx_mask at first, then overwritten by this function based on rx_mask at last. According to the author of tx|rx_mask, tx_mask is for playback and rx_mask is for capture. And stream direction is checked at all other references of tx|rx_mask in ASoC, so here should be an error. This patch checks stream direction for tx|rx_mask for fixup function. This issue would affect not only HDaudio+ASoC, but also I2S codecs if the channel number based on rx_mask is not equal to the one for tx_mask. It could be rarely reproduecd because most drivers in kernel set the same channel number to tx|rx_mask or rx_mask is zero. Tested on all platforms using stream_tag & HDaudio and intel I2S platforms. Signed-off-by: Rander Wang <rander.wang@linux.intel.com> Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Signed-off-by: Mark Brown <broonie@kernel.org> Signed-off-by: Sasha Levin <sashal@kernel.org>
2019-05-10ASoC: hdmi-codec: fix S/PDIF DAIRussell King
[ Upstream commit 2e95f984aae4cf0608d0ba2189c756f2bd50b44a ] When using the S/PDIF DAI, there is no requirement to call snd_soc_dai_set_fmt() as there is no DAI format definition that defines S/PDIF. In any case, S/PDIF does not have separate clocks, this is embedded into the data stream. Consequently, when attempting to use TDA998x in S/PDIF mode, the attempt to configure TDA998x via the hw_params callback fails as the hdmi_codec_daifmt is left initialised to zero. Since the S/PDIF DAI will only be used by S/PDIF, prepare the hdmi_codec_daifmt structure for this format. Signed-off-by: Russell King <rmk+kernel@armlinux.org.uk> Reviewed-by: Jyri Sarha <jsarha@ti.com> Signed-off-by: Mark Brown <broonie@kernel.org> Signed-off-by: Sasha Levin <sashal@kernel.org>
2019-05-08ASoC: stm32: fix sai driver name initialisationArnaud Pouliquen
commit 17d3069ccf06970e2db3f7cbf4335f207524279e upstream. This patch fixes the sai driver structure overwriting which results in a cpu dai name equal NULL. Fixes: 3e086ed ("ASoC: stm32: add SAI driver") Signed-off-by: Arnaud Pouliquen <arnaud.pouliquen@st.com> Signed-off-by: Mark Brown <broonie@kernel.org> Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
2019-04-17ASoC: fsl_esai: fix channel swap issue when stream startsS.j. Wang
commit 0ff4e8c61b794a4bf6c854ab071a1abaaa80f358 upstream. There is very low possibility ( < 0.1% ) that channel swap happened in beginning when multi output/input pin is enabled. The issue is that hardware can't send data to correct pin in the beginning with the normal enable flow. This is hardware issue, but there is no errata, the workaround flow is that: Each time playback/recording, firstly clear the xSMA/xSMB, then enable TE/RE, then enable xSMB and xSMA (xSMB must be enabled before xSMA). Which is to use the xSMA as the trigger start register, previously the xCR_TE or xCR_RE is the bit for starting. Fixes commit 43d24e76b698 ("ASoC: fsl_esai: Add ESAI CPU DAI driver") Cc: <stable@vger.kernel.org> Reviewed-by: Fabio Estevam <festevam@gmail.com> Acked-by: Nicolin Chen <nicoleotsuka@gmail.com> Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com> Signed-off-by: Mark Brown <broonie@kernel.org> Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
2019-04-05ASoC: fsl-asoc-card: fix object reference leaks in fsl_asoc_card_probewen yang
[ Upstream commit 11907e9d3533648615db08140e3045b829d2c141 ] The of_find_device_by_node() takes a reference to the underlying device structure, we should release that reference. Signed-off-by: Wen Yang <yellowriver2010@hotmil.com> Cc: Timur Tabi <timur@kernel.org> Cc: Nicolin Chen <nicoleotsuka@gmail.com> Cc: Xiubo Li <Xiubo.Lee@gmail.com> Cc: Fabio Estevam <festevam@gmail.com> Cc: Liam Girdwood <lgirdwood@gmail.com> Cc: Mark Brown <broonie@kernel.org> Cc: Jaroslav Kysela <perex@perex.cz> Cc: Takashi Iwai <tiwai@suse.com> Cc: alsa-devel@alsa-project.org Cc: linuxppc-dev@lists.ozlabs.org Cc: linux-kernel@vger.kernel.org Signed-off-by: Mark Brown <broonie@kernel.org> Signed-off-by: Sasha Levin <sashal@kernel.org>
2019-04-05SoC: imx-sgtl5000: add missing put_device()Wen Yang
[ Upstream commit 8fa857da9744f513036df1c43ab57f338941ae7d ] The of_find_device_by_node() takes a reference to the underlying device structure, we should release that reference. Detected by coccinelle with the following warnings: ./sound/soc/fsl/imx-sgtl5000.c:169:1-7: ERROR: missing put_device; call of_find_device_by_node on line 105, but without a corresponding object release within this function. ./sound/soc/fsl/imx-sgtl5000.c:177:1-7: ERROR: missing put_device; call of_find_device_by_node on line 105, but without a corresponding object release within this function. Signed-off-by: Wen Yang <yellowriver2010@hotmail.com> Cc: Timur Tabi <timur@kernel.org> Cc: Nicolin Chen <nicoleotsuka@gmail.com> Cc: Xiubo Li <Xiubo.Lee@gmail.com> Cc: Fabio Estevam <festevam@gmail.com> Cc: Liam Girdwood <lgirdwood@gmail.com> Cc: Mark Brown <broonie@kernel.org> Cc: Jaroslav Kysela <perex@perex.cz> Cc: Takashi Iwai <tiwai@suse.com> Cc: Shawn Guo <shawnguo@kernel.org> Cc: Sascha Hauer <s.hauer@pengutronix.de> Cc: Pengutronix Kernel Team <kernel@pengutronix.de> Cc: NXP Linux Team <linux-imx@nxp.com> Cc: alsa-devel@alsa-project.org Cc: linuxppc-dev@lists.ozlabs.org Cc: linux-arm-kernel@lists.infradead.org Cc: linux-kernel@vger.kernel.org Signed-off-by: Mark Brown <broonie@kernel.org> Signed-off-by: Sasha Levin <sashal@kernel.org>
2019-03-23ASoC: topology: free created components in tplg load errorBard liao
[ Upstream commit 304017d31df36fb61eb2ed3ebf65fb6870b3c731 ] Topology resources are no longer needed if any element failed to load. Signed-off-by: Bard liao <yung-chuan.liao@linux.intel.com> Signed-off-by: Mark Brown <broonie@kernel.org> Signed-off-by: Sasha Levin <sashal@kernel.org>
2019-03-23ASoC: rsnd: fixup rsnd_ssi_master_clk_start() user count checkKuninori Morimoto
[ Upstream commit d9111d36024de07784f2e1ba2ccf70b16035f378 ] commit 4d230d1271064 ("ASoC: rsnd: fixup not to call clk_get/set under non-atomic") added new rsnd_ssi_prepare() and moved rsnd_ssi_master_clk_start() to .prepare. But, ssi user count (= ssi->usrcnt) is incremented at .init (= rsnd_ssi_init()). Because of these timing exchange, ssi->usrcnt check at rsnd_ssi_master_clk_start() should be adjusted. Otherwise, 2nd master clock setup will be no check. This patch fixup this issue. Fixes: commit 4d230d1271064 ("ASoC: rsnd: fixup not to call clk_get/set under non-atomic") Reported-by: Yusuke Goda <yusuke.goda.sx@renesas.com> Reported-by: Valentine Barshak <valentine.barshak@cogentembedded.com> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Tested-by: Yusuke Goda <yusuke.goda.sx@renesas.com> Signed-off-by: Mark Brown <broonie@kernel.org> Signed-off-by: Sasha Levin <sashal@kernel.org>