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authorLinus Torvalds <torvalds@linux-foundation.org>2020-07-21 08:06:45 -0700
committerLinus Torvalds <torvalds@linux-foundation.org>2020-07-21 08:06:45 -0700
commit8c26c87b05323a7ccdc632820b85253e0bf47fd9 (patch)
treea7eed7f43a921398981afb481d74ef6a379eff92 /sound/soc/sof
parent4fa640dc52302b5e62b01b05c755b055549633ae (diff)
parent568e4e82128aac2c62c2c359ebebb6007fd794f9 (diff)
Merge tag 'sound-5.8-rc7' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound into master
Pull sound fixes from Takashi Iwai: "This became fairly large, containing mostly the collection of ASoC fixes that slipped from the previous request, so I sent now a bit earlier than usual. But all changes look small and mostly device-specific, hence nothing to worry too much. Majority of changes are for x86 based platforms and their CODEC drivers, in order to address some issues hit by their recent tests and fuzzing. The rest are other ASoC device-specific fixes (imx, qcom, wm8974, amd, rockchip) as well as a trivial fix for a kernel WARNING hit by syzkaller" * tag 'sound-5.8-rc7' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (28 commits) ALSA: hda/realtek: Fixed ALC298 sound bug by adding quirk for Samsung Notebook Pen S ALSA: info: Drop WARN_ON() from buffer NULL sanity check ASoC: rt5682: Report the button event in the headset type only ASoC: Intel: bytcht_es8316: Add missed put_device() ASoC: rt5682: Enable Vref2 under using PLL2 ASoC: rt286: fix unexpected interrupt happens ASoC: wm8974: remove unsupported clock mode ASoC: wm8974: fix Boost Mixer Aux Switch ASoC: SOF: core: fix null-ptr-deref bug during device removal ASoc: codecs: max98373: remove Idle_bias_on to let codec suspend ASoC: codecs: max98373: Removed superfluous volume control from chip default ASoC: topology: fix tlvs in error handling for widget_dmixer ASoC: topology: fix kernel oops on route addition error ASoC: SOF: imx: add min/max channels for SAI/ESAI on i.MX8/i.MX8M ASoC: Intel: bdw-rt5677: fix non BE conversion ASoC: soc-dai: set dai_link dpcm_ flags with a helper MAINTAINERS: Add Shengjiu to reviewer list of sound/soc/fsl ASoC: core: Remove only the registered component in devm functions MAINTAINERS: Change Maintainer for some at91 drivers ASoC: dt-bindings: simple-card: Fix 'make dt_binding_check' warnings ...
Diffstat (limited to 'sound/soc/sof')
-rw-r--r--sound/soc/sof/core.c10
-rw-r--r--sound/soc/sof/imx/imx8.c8
-rw-r--r--sound/soc/sof/imx/imx8m.c8
3 files changed, 21 insertions, 5 deletions
diff --git a/sound/soc/sof/core.c b/sound/soc/sof/core.c
index 339c4930b0c0..adc7c37145d6 100644
--- a/sound/soc/sof/core.c
+++ b/sound/soc/sof/core.c
@@ -345,15 +345,15 @@ int snd_sof_device_remove(struct device *dev)
struct snd_sof_pdata *pdata = sdev->pdata;
int ret;
- ret = snd_sof_dsp_power_down_notify(sdev);
- if (ret < 0)
- dev_warn(dev, "error: %d failed to prepare DSP for device removal",
- ret);
-
if (IS_ENABLED(CONFIG_SND_SOC_SOF_PROBE_WORK_QUEUE))
cancel_work_sync(&sdev->probe_work);
if (sdev->fw_state > SOF_FW_BOOT_NOT_STARTED) {
+ ret = snd_sof_dsp_power_down_notify(sdev);
+ if (ret < 0)
+ dev_warn(dev, "error: %d failed to prepare DSP for device removal",
+ ret);
+
snd_sof_fw_unload(sdev);
snd_sof_ipc_free(sdev);
snd_sof_free_debug(sdev);
diff --git a/sound/soc/sof/imx/imx8.c b/sound/soc/sof/imx/imx8.c
index 63f9c20a1bac..a4fa8451d8cb 100644
--- a/sound/soc/sof/imx/imx8.c
+++ b/sound/soc/sof/imx/imx8.c
@@ -375,6 +375,14 @@ static int imx8_ipc_pcm_params(struct snd_sof_dev *sdev,
static struct snd_soc_dai_driver imx8_dai[] = {
{
.name = "esai-port",
+ .playback = {
+ .channels_min = 1,
+ .channels_max = 8,
+ },
+ .capture = {
+ .channels_min = 1,
+ .channels_max = 8,
+ },
},
};
diff --git a/sound/soc/sof/imx/imx8m.c b/sound/soc/sof/imx/imx8m.c
index fa86a9e2990f..287114a37688 100644
--- a/sound/soc/sof/imx/imx8m.c
+++ b/sound/soc/sof/imx/imx8m.c
@@ -240,6 +240,14 @@ static int imx8m_ipc_pcm_params(struct snd_sof_dev *sdev,
static struct snd_soc_dai_driver imx8m_dai[] = {
{
.name = "sai-port",
+ .playback = {
+ .channels_min = 1,
+ .channels_max = 32,
+ },
+ .capture = {
+ .channels_min = 1,
+ .channels_max = 32,
+ },
},
};