From 8d01c3c73cefbb5bacffe804427daed3e6051435 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Fri, 10 May 2019 11:38:03 -0500 Subject: ASoC: SOF: nocodec: fix undefined reference The nocodec option can be selected individually, leading to the following issue: sound/soc/sof/core.o: In function `snd_sof_device_probe': core.c:(.text+0x4af): undefined reference to `sof_nocodec_setup' Fix by selecting the SND_SOF_NOCODEC option as needed. Reported-by: Hulk Robot Reported-by: YueHaibing Signed-off-by: Pierre-Louis Bossart Signed-off-by: Mark Brown --- sound/soc/sof/Kconfig | 8 ++++++-- 1 file changed, 6 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/sof/Kconfig b/sound/soc/sof/Kconfig index b204c65698f9..a9a1d502daae 100644 --- a/sound/soc/sof/Kconfig +++ b/sound/soc/sof/Kconfig @@ -44,7 +44,10 @@ config SND_SOC_SOF_OPTIONS if SND_SOC_SOF_OPTIONS config SND_SOC_SOF_NOCODEC - tristate "SOF nocodec mode Support" + tristate + +config SND_SOC_SOF_NOCODEC_SUPPORT + bool "SOF nocodec mode support" help This adds support for a dummy/nocodec machine driver fallback option if no known codec is detected. This is typically only @@ -80,7 +83,7 @@ if SND_SOC_SOF_DEBUG config SND_SOC_SOF_FORCE_NOCODEC_MODE bool "SOF force nocodec Mode" - depends on SND_SOC_SOF_NOCODEC + depends on SND_SOC_SOF_NOCODEC_SUPPORT help This forces SOF to use dummy/nocodec as machine driver, even though there is a codec detected on the real platform. This is @@ -135,6 +138,7 @@ endif ## SND_SOC_SOF_OPTIONS config SND_SOC_SOF tristate select SND_SOC_TOPOLOGY + select SND_SOC_SOF_NOCODEC if SND_SOC_SOF_NOCODEC_SUPPORT help This option is not user-selectable but automagically handled by 'select' statements at a higher level -- cgit v1.2.3 From a69270d8bfeb35fc89d047ea6db803cd75a74f12 Mon Sep 17 00:00:00 2001 From: Kai Vehmanen Date: Thu, 9 May 2019 15:10:23 -0500 Subject: ASoC: SOF: fix race in FW boot timeout handling A race condition exists in handling firmware boot timeout. If FW sends FW_READY just after boot timeout has expired in driver, a kernel exception will result as FW_READY handler will be run while the state is still being cleaned up in snd_sof_run_firmware(). Avoid the race by setting boot_complete also in the error case. Signed-off-by: Kai Vehmanen Signed-off-by: Pierre-Louis Bossart Signed-off-by: Mark Brown --- sound/soc/sof/loader.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound') diff --git a/sound/soc/sof/loader.c b/sound/soc/sof/loader.c index 81c7452aae17..628fae552442 100644 --- a/sound/soc/sof/loader.c +++ b/sound/soc/sof/loader.c @@ -372,6 +372,8 @@ int snd_sof_run_firmware(struct snd_sof_dev *sdev) msecs_to_jiffies(sdev->boot_timeout)); if (ret == 0) { dev_err(sdev->dev, "error: firmware boot failure\n"); + /* after this point FW_READY msg should be ignored */ + sdev->boot_complete = true; snd_sof_dsp_dbg_dump(sdev, SOF_DBG_REGS | SOF_DBG_MBOX | SOF_DBG_TEXT | SOF_DBG_PCI); return -EIO; -- cgit v1.2.3 From 8199a12037892f01e2cf5bedf5fbf08dff11b282 Mon Sep 17 00:00:00 2001 From: Ranjani Sridharan Date: Thu, 9 May 2019 15:10:25 -0500 Subject: ASoC: SOF: fix error in verbose ipc command parsing Remove the erroneous addition of "SET_VALUE" to the GLB IPC command string. Signed-off-by: Ranjani Sridharan Signed-off-by: Pierre-Louis Bossart Signed-off-by: Mark Brown --- sound/soc/sof/ipc.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/sof/ipc.c b/sound/soc/sof/ipc.c index f0b9d3c53f6f..894e68cbd69d 100644 --- a/sound/soc/sof/ipc.c +++ b/sound/soc/sof/ipc.c @@ -115,7 +115,7 @@ static void ipc_log_header(struct device *dev, u8 *text, u32 cmd) } break; case SOF_IPC_GLB_COMP_MSG: - str = "GLB_COMP_MSG: SET_VALUE"; + str = "GLB_COMP_MSG"; switch (type) { case SOF_IPC_COMP_SET_VALUE: str2 = "SET_VALUE"; break; -- cgit v1.2.3 From f3df05c805983427319eddc2411a2105ee1757cf Mon Sep 17 00:00:00 2001 From: Matt Flax Date: Wed, 8 May 2019 16:33:13 +1000 Subject: ASoC : cs4265 : readable register too low 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 Reviewed-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/cs4265.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/cs4265.c b/sound/soc/codecs/cs4265.c index ab27d2b94d02..c0190ec59e74 100644 --- a/sound/soc/codecs/cs4265.c +++ b/sound/soc/codecs/cs4265.c @@ -60,7 +60,7 @@ static const struct reg_default cs4265_reg_defaults[] = { static bool cs4265_readable_register(struct device *dev, unsigned int reg) { switch (reg) { - case CS4265_CHIP_ID ... CS4265_SPDIF_CTL2: + case CS4265_CHIP_ID ... CS4265_MAX_REGISTER: return true; default: return false; -- cgit v1.2.3 From a8dee20d792432740509237943700fbcfc230bad Mon Sep 17 00:00:00 2001 From: Viorel Suman Date: Thu, 9 May 2019 13:30:36 +0000 Subject: ASoC: ak4458: add return value for ak4458_probe AK4458 is probed successfully even if AK4458 is not present - this is caused by probe function returning no error on i2c access failure. Return an error on probe if i2c access has failed. Signed-off-by: Shengjiu Wang Signed-off-by: Viorel Suman Signed-off-by: Mark Brown --- sound/soc/codecs/ak4458.c | 13 +++++++------ 1 file changed, 7 insertions(+), 6 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/ak4458.c b/sound/soc/codecs/ak4458.c index eab7c76cfcd9..4c5c3ec92609 100644 --- a/sound/soc/codecs/ak4458.c +++ b/sound/soc/codecs/ak4458.c @@ -536,9 +536,10 @@ static void ak4458_power_on(struct ak4458_priv *ak4458) } } -static void ak4458_init(struct snd_soc_component *component) +static int ak4458_init(struct snd_soc_component *component) { struct ak4458_priv *ak4458 = snd_soc_component_get_drvdata(component); + int ret; /* External Mute ON */ if (ak4458->mute_gpiod) @@ -546,21 +547,21 @@ static void ak4458_init(struct snd_soc_component *component) ak4458_power_on(ak4458); - snd_soc_component_update_bits(component, AK4458_00_CONTROL1, + ret = snd_soc_component_update_bits(component, AK4458_00_CONTROL1, 0x80, 0x80); /* ACKS bit = 1; 10000000 */ + if (ret < 0) + return ret; - ak4458_rstn_control(component, 1); + return ak4458_rstn_control(component, 1); } static int ak4458_probe(struct snd_soc_component *component) { struct ak4458_priv *ak4458 = snd_soc_component_get_drvdata(component); - ak4458_init(component); - ak4458->fs = 48000; - return 0; + return ak4458_init(component); } static void ak4458_remove(struct snd_soc_component *component) -- cgit v1.2.3 From 5087a8f17df868601cd7568299e91c28086d2b45 Mon Sep 17 00:00:00 2001 From: Libin Yang Date: Wed, 8 May 2019 10:32:41 +0800 Subject: ASoC: soc-pcm: BE dai needs prepare when pause release after resume 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 Signed-off-by: Mark Brown --- sound/soc/soc-pcm.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 74695355c1f8..7347e6f99248 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -2471,7 +2471,8 @@ int dpcm_be_dai_prepare(struct snd_soc_pcm_runtime *fe, int stream) if ((be->dpcm[stream].state != SND_SOC_DPCM_STATE_HW_PARAMS) && (be->dpcm[stream].state != SND_SOC_DPCM_STATE_STOP) && - (be->dpcm[stream].state != SND_SOC_DPCM_STATE_SUSPEND)) + (be->dpcm[stream].state != SND_SOC_DPCM_STATE_SUSPEND) && + (be->dpcm[stream].state != SND_SOC_DPCM_STATE_PAUSED)) continue; dev_dbg(be->dev, "ASoC: prepare BE %s\n", -- cgit v1.2.3 From 176a11834b65ec35e3b7a953f87fb9cc41309497 Mon Sep 17 00:00:00 2001 From: Viorel Suman Date: Mon, 13 May 2019 10:02:42 +0000 Subject: ASoC: ak4458: rstn_control - return a non-zero on error only snd_soc_component_update_bits() may return 1 if operation was successful and the value of the register changed. Return a non-zero in ak4458_rstn_control for an error only. Signed-off-by: Shengjiu Wang Signed-off-by: Viorel Suman Reviewed-by: Daniel Baluta Signed-off-by: Mark Brown --- sound/soc/codecs/ak4458.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/ak4458.c b/sound/soc/codecs/ak4458.c index 4c5c3ec92609..71562154c0b1 100644 --- a/sound/soc/codecs/ak4458.c +++ b/sound/soc/codecs/ak4458.c @@ -304,7 +304,10 @@ static int ak4458_rstn_control(struct snd_soc_component *component, int bit) AK4458_00_CONTROL1, AK4458_RSTN_MASK, 0x0); - return ret; + if (ret < 0) + return ret; + + return 0; } static int ak4458_hw_params(struct snd_pcm_substream *substream, -- cgit v1.2.3 From b06c58c2a1eed571ea2a6640fdb85b7b00196b1e Mon Sep 17 00:00:00 2001 From: S.j. Wang Date: Wed, 15 May 2019 06:42:18 +0000 Subject: ASoC: fsl_asrc: Fix the issue about unsupported rate 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: Signed-off-by: Shengjiu Wang Acked-by: Nicolin Chen Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_asrc.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/fsl/fsl_asrc.c b/sound/soc/fsl/fsl_asrc.c index 0b937924d2e4..ea035c12a325 100644 --- a/sound/soc/fsl/fsl_asrc.c +++ b/sound/soc/fsl/fsl_asrc.c @@ -282,8 +282,8 @@ static int fsl_asrc_config_pair(struct fsl_asrc_pair *pair) return -EINVAL; } - if ((outrate > 8000 && outrate < 30000) && - (outrate/inrate > 24 || inrate/outrate > 8)) { + if ((outrate >= 8000 && outrate <= 30000) && + (outrate > 24 * inrate || inrate > 8 * outrate)) { pair_err("exceed supported ratio range [1/24, 8] for \ inrate/outrate: %d/%d\n", inrate, outrate); return -EINVAL; -- cgit v1.2.3 From ad6eecbfc01c987e0253371f274c3872042e4350 Mon Sep 17 00:00:00 2001 From: S.j. Wang Date: Thu, 16 May 2019 06:04:29 +0000 Subject: ASoC: cs42xx8: Add regcache mask dirty 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: Signed-off-by: Shengjiu Wang Signed-off-by: Mark Brown --- sound/soc/codecs/cs42xx8.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/soc/codecs/cs42xx8.c b/sound/soc/codecs/cs42xx8.c index ebb9e0cf8364..28a4ac36c4f8 100644 --- a/sound/soc/codecs/cs42xx8.c +++ b/sound/soc/codecs/cs42xx8.c @@ -558,6 +558,7 @@ static int cs42xx8_runtime_resume(struct device *dev) msleep(5); regcache_cache_only(cs42xx8->regmap, false); + regcache_mark_dirty(cs42xx8->regmap); ret = regcache_sync(cs42xx8->regmap); if (ret) { -- cgit v1.2.3 From f7c4842abfa1a219554a3ffd8c317e8fdd979bec Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Fri, 17 May 2019 10:21:12 +0900 Subject: ASoC: soc-dpm: fixup DAI active unbalance snd_soc_dai_link_event() is updating snd_soc_dai :: active, but it is unbalance. It counts up if it has startup callback. case SND_SOC_DAPM_PRE_PMU: ... snd_soc_dapm_widget_for_each_source_path(w, path) { ... if (source->driver->ops->startup) { ... => source->active++; } ... } ... But, always counts down case SND_SOC_DAPM_PRE_PMD: ... snd_soc_dapm_widget_for_each_source_path(w, path) { ... => source->active--; ... } This patch always counts up when SND_SOC_DAPM_PRE_PMD. Signed-off-by: Kuninori Morimoto Reviewed-by: Vinod Koul Signed-off-by: Mark Brown --- sound/soc/soc-dapm.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 65ee0bb5dd0b..62e27defce56 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -3828,8 +3828,8 @@ static int snd_soc_dai_link_event(struct snd_soc_dapm_widget *w, ret); goto out; } - source->active++; } + source->active++; ret = soc_dai_hw_params(&substream, params, source); if (ret < 0) goto out; @@ -3850,8 +3850,8 @@ static int snd_soc_dai_link_event(struct snd_soc_dapm_widget *w, ret); goto out; } - sink->active++; } + sink->active++; ret = soc_dai_hw_params(&substream, params, sink); if (ret < 0) goto out; -- cgit v1.2.3 From 7b8164c1a29ce8ef91672c50ceac5c14475f5601 Mon Sep 17 00:00:00 2001 From: Curtis Malainey Date: Thu, 16 May 2019 18:43:40 -0700 Subject: ASoC: rt5677-spi: Handle over reading when flipping bytes There is a case when a we want to read a large number of bytes that require a burst but is not a multiple of the word size (8). When this happens rt5677_spi_reverse will run off the end of the buffer. The solution is to tell spi_reverse the actual size of the destination and stop if we reach it even if we have data left that we read. Cc: Ben Zhang Signed-off-by: Curtis Malainey Signed-off-by: Mark Brown --- sound/soc/codecs/rt5677-spi.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/rt5677-spi.c b/sound/soc/codecs/rt5677-spi.c index 84b6bd8b50e1..a4dfa0345c6e 100644 --- a/sound/soc/codecs/rt5677-spi.c +++ b/sound/soc/codecs/rt5677-spi.c @@ -101,7 +101,7 @@ static void rt5677_spi_reverse(u8 *dst, u32 dstlen, const u8 *src, u32 srclen) u32 word_size = min_t(u32, dstlen, 8); for (w = 0; w < dstlen; w += word_size) { - for (i = 0; i < word_size; i++) { + for (i = 0; i < word_size && i + w < dstlen; i++) { si = w + word_size - i - 1; dst[w + i] = si < srclen ? src[si] : 0; } @@ -152,8 +152,9 @@ int rt5677_spi_read(u32 addr, void *rxbuf, size_t len) status |= spi_sync(g_spi, &m); mutex_unlock(&spi_mutex); + /* Copy data back to caller buffer */ - rt5677_spi_reverse(cb + offset, t[1].len, body, t[1].len); + rt5677_spi_reverse(cb + offset, len - offset, body, t[1].len); } return status; } -- cgit v1.2.3 From 30d9d4ff53532087bc13ed29d7715df868794b5e Mon Sep 17 00:00:00 2001 From: Sathya Prakash M R Date: Sat, 18 May 2019 13:30:08 -0500 Subject: ASoC: Intel: soc-acpi: Fix machine selection order The selection order of m/c in match table is corrected to use common codec as last in the list. Signed-off-by: Sathya Prakash M R Signed-off-by: Pierre-Louis Bossart Signed-off-by: Mark Brown --- sound/soc/intel/common/soc-acpi-intel-cnl-match.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/soc/intel/common/soc-acpi-intel-cnl-match.c b/sound/soc/intel/common/soc-acpi-intel-cnl-match.c index df7c52cad5c3..c36c0aa4f683 100644 --- a/sound/soc/intel/common/soc-acpi-intel-cnl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-cnl-match.c @@ -29,17 +29,17 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_cnl_machines[] = { .sof_tplg_filename = "sof-cnl-rt274.tplg", }, { - .id = "10EC5682", + .id = "MX98357A", .drv_name = "sof_rt5682", + .quirk_data = &cml_codecs, .sof_fw_filename = "sof-cnl.ri", - .sof_tplg_filename = "sof-cml-rt5682.tplg", + .sof_tplg_filename = "sof-cml-rt5682-max98357a.tplg", }, { - .id = "MX98357A", + .id = "10EC5682", .drv_name = "sof_rt5682", - .quirk_data = &cml_codecs, .sof_fw_filename = "sof-cnl.ri", - .sof_tplg_filename = "sof-cml-rt5682-max98357a.tplg", + .sof_tplg_filename = "sof-cml-rt5682.tplg", }, {}, -- cgit v1.2.3 From 069d037aea98ffa64c26d4b1dc958fb8f39f5c2b Mon Sep 17 00:00:00 2001 From: Jon Hunter Date: Thu, 16 May 2019 18:51:26 +0100 Subject: ASoC: simple-card: Fix configuration of DAI format When configuring a codec to be both bit-clock and frame-master, it was found that the codec was always configured as bit-clock and frame-slave. Looking at the simple_dai_link_of() function there appears to be two problems with the configuration of the DAI format, which are ... 1. The function asoc_simple_parse_daifmt() is called before the function asoc_simple_parse_codec() and this means that the device-tree node for the codec has not been parsed yet, which is needed by the function asoc_simple_parse_daifmt() to determine who is the codec. 2. The phandle passed to asoc_simple_parse_daifmt() is the phandle to the 'codec' node and not the phandle of the actual codec defined by the 'sound-dai' property under the 'codec' node. Fix the above by moving the call to asoc_simple_parse_daifmt() after the the call to asoc_simple_parse_codec() and pass the phandle for the codec to asoc_simple_parse_daifmt(). Signed-off-by: Jon Hunter Signed-off-by: Mark Brown --- sound/soc/generic/simple-card.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index 9b568f578bcd..c2c8dcbcf795 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -283,11 +283,6 @@ static int simple_dai_link_of(struct asoc_simple_priv *priv, codec_dai = dai_props->codec_dai = &priv->dais[li->dais++]; - ret = asoc_simple_parse_daifmt(dev, node, codec, - prefix, &dai_link->dai_fmt); - if (ret < 0) - goto dai_link_of_err; - simple_parse_mclk_fs(top, cpu, codec, dai_props, prefix); ret = asoc_simple_parse_cpu(cpu, dai_link, &single_cpu); @@ -298,6 +293,11 @@ static int simple_dai_link_of(struct asoc_simple_priv *priv, if (ret < 0) goto dai_link_of_err; + ret = asoc_simple_parse_daifmt(dev, node, dai_link->codecs->of_node, + prefix, &dai_link->dai_fmt); + if (ret < 0) + goto dai_link_of_err; + ret = asoc_simple_parse_platform(plat, dai_link); if (ret < 0) goto dai_link_of_err; -- cgit v1.2.3 From d5952f34ade5e6034e5eca3617fb77d4395bf492 Mon Sep 17 00:00:00 2001 From: Sathya Prakash M R Date: Mon, 20 May 2019 14:46:41 -0500 Subject: ASoC: Intel: sof-rt5682: fix for codec button mapping The RT5682 codec button mapping, initially copied from the DA7219 one, needs to be corrected. Signed-off-by: Sathya Prakash M R Signed-off-by: Pierre-Louis Bossart Signed-off-by: Mark Brown --- sound/soc/intel/boards/sof_rt5682.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/intel/boards/sof_rt5682.c b/sound/soc/intel/boards/sof_rt5682.c index f28fb98cc306..39ddefec4991 100644 --- a/sound/soc/intel/boards/sof_rt5682.c +++ b/sound/soc/intel/boards/sof_rt5682.c @@ -144,9 +144,9 @@ static int sof_rt5682_codec_init(struct snd_soc_pcm_runtime *rtd) jack = &ctx->sof_headset; snd_jack_set_key(jack->jack, SND_JACK_BTN_0, KEY_PLAYPAUSE); - snd_jack_set_key(jack->jack, SND_JACK_BTN_1, KEY_VOLUMEUP); - snd_jack_set_key(jack->jack, SND_JACK_BTN_2, KEY_VOLUMEDOWN); - snd_jack_set_key(jack->jack, SND_JACK_BTN_3, KEY_VOICECOMMAND); + snd_jack_set_key(jack->jack, SND_JACK_BTN_1, KEY_VOICECOMMAND); + snd_jack_set_key(jack->jack, SND_JACK_BTN_2, KEY_VOLUMEUP); + snd_jack_set_key(jack->jack, SND_JACK_BTN_3, KEY_VOLUMEDOWN); ret = snd_soc_component_set_jack(component, jack, NULL); if (ret) { -- cgit v1.2.3 From df9366131a452296d040a7a496d93108f1fc240c Mon Sep 17 00:00:00 2001 From: Sathya Prakash M R Date: Mon, 20 May 2019 14:46:42 -0500 Subject: ASoC: Intel: sof-rt5682: fix AMP quirk support The use of BIT/GENMASK was incorrect, fix. Signed-off-by: Sathya Prakash M R Signed-off-by: Pierre-Louis Bossart Signed-off-by: Mark Brown --- sound/soc/intel/boards/sof_rt5682.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/intel/boards/sof_rt5682.c b/sound/soc/intel/boards/sof_rt5682.c index 39ddefec4991..3343dbcd506f 100644 --- a/sound/soc/intel/boards/sof_rt5682.c +++ b/sound/soc/intel/boards/sof_rt5682.c @@ -29,9 +29,10 @@ #define SOF_RT5682_MCLK_EN BIT(3) #define SOF_RT5682_MCLK_24MHZ BIT(4) #define SOF_SPEAKER_AMP_PRESENT BIT(5) -#define SOF_RT5682_SSP_AMP(quirk) ((quirk) & GENMASK(8, 6)) -#define SOF_RT5682_SSP_AMP_MASK (GENMASK(8, 6)) #define SOF_RT5682_SSP_AMP_SHIFT 6 +#define SOF_RT5682_SSP_AMP_MASK (GENMASK(8, 6)) +#define SOF_RT5682_SSP_AMP(quirk) \ + (((quirk) << SOF_RT5682_SSP_AMP_SHIFT) & SOF_RT5682_SSP_AMP_MASK) /* Default: MCLK on, MCLK 19.2M, SSP0 */ static unsigned long sof_rt5682_quirk = SOF_RT5682_MCLK_EN | -- cgit v1.2.3