From b66c9b911fe6ca188002b342b05c43deab4491a3 Mon Sep 17 00:00:00 2001 From: Fabio Estevam Date: Tue, 26 Jun 2018 08:58:35 -0300 Subject: ASoC: soc-utils: Fix unregistration order The unregistration should happen in the opposite order of the registration, so change it accordingly. No real issue has been noticed, but it is good practice to keep the correct unregistration order. Signed-off-by: Fabio Estevam Signed-off-by: Mark Brown --- sound/soc/soc-utils.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/soc-utils.c b/sound/soc/soc-utils.c index 2d9e98bd1530..a863bb3f66c2 100644 --- a/sound/soc/soc-utils.c +++ b/sound/soc/soc-utils.c @@ -381,6 +381,6 @@ int __init snd_soc_util_init(void) void __exit snd_soc_util_exit(void) { - platform_device_unregister(soc_dummy_dev); platform_driver_unregister(&soc_dummy_driver); + platform_device_unregister(soc_dummy_dev); } -- cgit v1.2.3-70-g09d2 From 4febced15ac8ddb9cf3e603edb111842e4863d9a Mon Sep 17 00:00:00 2001 From: Jerome Brunet Date: Wed, 27 Jun 2018 17:36:38 +0200 Subject: ASoC: dpcm: don't merge format from invalid codec dai When merging codec formats, dpcm_runtime_base_format() should skip the codecs which are not supporting the current stream direction. At the moment, if a BE link has more than one codec, and only one of these codecs has no capture DAI, it becomes impossible to start a capture stream because the merged format would be 0. Skipping invalid codec DAI solves the problem. Fixes: b073ed4e2126 ("ASoC: soc-pcm: DPCM cares BE format") Signed-off-by: Jerome Brunet Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/soc-pcm.c | 8 ++++++++ 1 file changed, 8 insertions(+) (limited to 'sound') diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 5e7ae47a9658..5feae9666822 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -1694,6 +1694,14 @@ static u64 dpcm_runtime_base_format(struct snd_pcm_substream *substream) int i; for (i = 0; i < be->num_codecs; i++) { + /* + * Skip CODECs which don't support the current stream + * type. See soc_pcm_init_runtime_hw() for more details + */ + if (!snd_soc_dai_stream_valid(be->codec_dais[i], + stream)) + continue; + codec_dai_drv = be->codec_dais[i]->driver; if (stream == SNDRV_PCM_STREAM_PLAYBACK) codec_stream = &codec_dai_drv->playback; -- cgit v1.2.3-70-g09d2 From 110743189c863e96dc08a581d56c50b965870a3f Mon Sep 17 00:00:00 2001 From: Srinivas Kandagatla Date: Wed, 4 Jul 2018 10:49:43 +0100 Subject: ASoC: qdsp6: q6afe-dai: do not close port if its not opened afe ports are open as part of prepare, so for use cases like "aplay sample.wav" were sample.wav is not present. This would call port close eventhough port was never opened. DSP would return errors for such use cases. Avoid doing this by checking the port state. Signed-off-by: Srinivas Kandagatla Signed-off-by: Mark Brown --- sound/soc/qcom/qdsp6/q6afe-dai.c | 3 +++ 1 file changed, 3 insertions(+) (limited to 'sound') diff --git a/sound/soc/qcom/qdsp6/q6afe-dai.c b/sound/soc/qcom/qdsp6/q6afe-dai.c index 5002dd05bf27..a373ca5523ff 100644 --- a/sound/soc/qcom/qdsp6/q6afe-dai.c +++ b/sound/soc/qcom/qdsp6/q6afe-dai.c @@ -315,6 +315,9 @@ static void q6afe_dai_shutdown(struct snd_pcm_substream *substream, struct q6afe_dai_data *dai_data = dev_get_drvdata(dai->dev); int rc; + if (!dai_data->is_port_started[dai->id]) + return; + rc = q6afe_port_stop(dai_data->port[dai->id]); if (rc < 0) dev_err(dai->dev, "fail to close AFE port (%d)\n", rc); -- cgit v1.2.3-70-g09d2 From 5dffc1752cabde6396fca28ff8343febfa524512 Mon Sep 17 00:00:00 2001 From: Srinivas Kandagatla Date: Wed, 4 Jul 2018 10:49:44 +0100 Subject: ASoC: qdsp6: q6asm-dai: do not close port if its not opened asm ports are open as part of prepare, so for use cases like "aplay sample.wav" were sample.wav is not present. This would call port close eventhough port was never opened. DSP would return errors for such use cases. Avoid doing this by checking the port state. Signed-off-by: Srinivas Kandagatla Signed-off-by: Mark Brown --- sound/soc/qcom/qdsp6/q6asm-dai.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/qcom/qdsp6/q6asm-dai.c b/sound/soc/qcom/qdsp6/q6asm-dai.c index 349c6a883c63..360936703b3d 100644 --- a/sound/soc/qcom/qdsp6/q6asm-dai.c +++ b/sound/soc/qcom/qdsp6/q6asm-dai.c @@ -390,7 +390,9 @@ static int q6asm_dai_close(struct snd_pcm_substream *substream) struct q6asm_dai_rtd *prtd = runtime->private_data; if (prtd->audio_client) { - q6asm_cmd(prtd->audio_client, CMD_CLOSE); + if (prtd->state) + q6asm_cmd(prtd->audio_client, CMD_CLOSE); + q6asm_unmap_memory_regions(substream->stream, prtd->audio_client); q6asm_audio_client_free(prtd->audio_client); -- cgit v1.2.3-70-g09d2 From b8110a87b75f948d978c06e130cc68026645c4a1 Mon Sep 17 00:00:00 2001 From: Dan Carpenter Date: Fri, 13 Jul 2018 18:05:57 +0300 Subject: ASoC: qdsp6: q6afe-dai: fix a range check in of_q6afe_parse_dai_data() The main thing is that the data->priv[] array has AFE_PORT_MAX elements so the > condition should be >=. But we may as well check for negative values as well just to be safe. Fixes: 24c4cbcfac09 ("ASoC: qdsp6: q6afe: Add q6afe dai driver") Signed-off-by: Dan Carpenter Signed-off-by: Mark Brown --- sound/soc/qcom/qdsp6/q6afe-dai.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/qcom/qdsp6/q6afe-dai.c b/sound/soc/qcom/qdsp6/q6afe-dai.c index a373ca5523ff..9ba95956ada8 100644 --- a/sound/soc/qcom/qdsp6/q6afe-dai.c +++ b/sound/soc/qcom/qdsp6/q6afe-dai.c @@ -1183,7 +1183,7 @@ static void of_q6afe_parse_dai_data(struct device *dev, int id, i, num_lines; ret = of_property_read_u32(node, "reg", &id); - if (ret || id > AFE_PORT_MAX) { + if (ret || id < 0 || id >= AFE_PORT_MAX) { dev_err(dev, "valid dai id not found:%d\n", ret); continue; } -- cgit v1.2.3-70-g09d2 From 090345ce7265dd111299e3a63cdc79c3ef924481 Mon Sep 17 00:00:00 2001 From: Dan Carpenter Date: Fri, 13 Jul 2018 18:11:04 +0300 Subject: ASoC: qdsp6: q6routing: off by one in routing_hw_params() The data->port_data[] array has AFE_MAX_PORTS elements so the check should be >= instead of > or we write one element beyond the end of the array. Fixes: e3a33673e845 ("ASoC: qdsp6: q6routing: Add q6routing driver") Signed-off-by: Dan Carpenter Signed-off-by: Mark Brown --- sound/soc/qcom/qdsp6/q6routing.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/qcom/qdsp6/q6routing.c b/sound/soc/qcom/qdsp6/q6routing.c index 593f66b8622f..7a19d6278406 100644 --- a/sound/soc/qcom/qdsp6/q6routing.c +++ b/sound/soc/qcom/qdsp6/q6routing.c @@ -899,7 +899,7 @@ static int routing_hw_params(struct snd_pcm_substream *substream, else path_type = ADM_PATH_LIVE_REC; - if (be_id > AFE_MAX_PORTS) + if (be_id >= AFE_MAX_PORTS) return -EINVAL; session = &data->port_data[be_id]; -- cgit v1.2.3-70-g09d2 From d30e23d69981a4b665f5ce8711335df986576389 Mon Sep 17 00:00:00 2001 From: Russell King Date: Sat, 14 Jul 2018 16:01:06 +0100 Subject: ASoC: hdmi-codec: fix routing Commit 943fa0228252 ("ASoC: hdmi-codec: Use different name for playback streams") broke hdmi-codec's routing between it's output "TX" widget and the S/PDIF or I2S streams by renaming the streams. Whether an error occurs or not is dependent on whether there is another widget called "Playback" registered by some other component - if there is, that widget will be (incorrectly) bound to the HDMI codec's "TX" output widget. If we end up connecting "TX" incorrectly, it can result in components not being started, causing no audio output. Since the I2S and S/PDIF streams now have different names, we can't use a static route at component level to describe the relationship, so arrange to dynamically create the route when the DAI driver is probed. Fixes: 943fa0228252 ("ASoC: hdmi-codec: Use different name for playback streams") Signed-off-by: Russell King Signed-off-by: Mark Brown --- sound/soc/codecs/hdmi-codec.c | 21 +++++++++++++++------ 1 file changed, 15 insertions(+), 6 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/hdmi-codec.c b/sound/soc/codecs/hdmi-codec.c index 6fa11888672d..3e5b12de71bb 100644 --- a/sound/soc/codecs/hdmi-codec.c +++ b/sound/soc/codecs/hdmi-codec.c @@ -291,10 +291,6 @@ static const struct snd_soc_dapm_widget hdmi_widgets[] = { SND_SOC_DAPM_OUTPUT("TX"), }; -static const struct snd_soc_dapm_route hdmi_routes[] = { - { "TX", NULL, "Playback" }, -}; - enum { DAI_ID_I2S = 0, DAI_ID_SPDIF, @@ -689,9 +685,23 @@ static int hdmi_codec_pcm_new(struct snd_soc_pcm_runtime *rtd, return snd_ctl_add(rtd->card->snd_card, kctl); } +static int hdmi_dai_probe(struct snd_soc_dai *dai) +{ + struct snd_soc_dapm_context *dapm; + struct snd_soc_dapm_route route = { + .sink = "TX", + .source = dai->driver->playback.stream_name, + }; + + dapm = snd_soc_component_get_dapm(dai->component); + + return snd_soc_dapm_add_routes(dapm, &route, 1); +} + static const struct snd_soc_dai_driver hdmi_i2s_dai = { .name = "i2s-hifi", .id = DAI_ID_I2S, + .probe = hdmi_dai_probe, .playback = { .stream_name = "I2S Playback", .channels_min = 2, @@ -707,6 +717,7 @@ static const struct snd_soc_dai_driver hdmi_i2s_dai = { static const struct snd_soc_dai_driver hdmi_spdif_dai = { .name = "spdif-hifi", .id = DAI_ID_SPDIF, + .probe = hdmi_dai_probe, .playback = { .stream_name = "SPDIF Playback", .channels_min = 2, @@ -733,8 +744,6 @@ static int hdmi_of_xlate_dai_id(struct snd_soc_component *component, static const struct snd_soc_component_driver hdmi_driver = { .dapm_widgets = hdmi_widgets, .num_dapm_widgets = ARRAY_SIZE(hdmi_widgets), - .dapm_routes = hdmi_routes, - .num_dapm_routes = ARRAY_SIZE(hdmi_routes), .of_xlate_dai_id = hdmi_of_xlate_dai_id, .idle_bias_on = 1, .use_pmdown_time = 1, -- cgit v1.2.3-70-g09d2 From 868e49a4a00afaca07d2c450a02e49581eaece6c Mon Sep 17 00:00:00 2001 From: Stuart Henderson Date: Thu, 19 Jul 2018 11:50:37 +0100 Subject: ASoC: wm_adsp: Ensure DSP boot work complete before preloader_put return All controls derived from the loaded firmware should be created prior to returning from the preloader's put function, such that they are immediately available to user-space. Signed-off-by: Stuart Henderson Signed-off-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/wm_adsp.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index b7b914963c62..4e7f8525449e 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -2672,6 +2672,8 @@ int wm_adsp2_preloader_put(struct snd_kcontrol *kcontrol, snd_soc_dapm_sync(dapm); + flush_work(&dsp->boot_work); + return 0; } EXPORT_SYMBOL_GPL(wm_adsp2_preloader_put); -- cgit v1.2.3-70-g09d2 From d96f8bd28cd0bae3e6702ae90df593628ef6906f Mon Sep 17 00:00:00 2001 From: Oder Chiou Date: Tue, 24 Jul 2018 15:49:23 +0800 Subject: ASoC: rt5514: Fix the issue of the delay volume applied The patch fixes the issue of the delay volume applied. Signed-off-by: Oder Chiou Signed-off-by: Mark Brown --- sound/soc/codecs/rt5514.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/rt5514.c b/sound/soc/codecs/rt5514.c index 1570b91bf018..dca82dd6e3bf 100644 --- a/sound/soc/codecs/rt5514.c +++ b/sound/soc/codecs/rt5514.c @@ -64,8 +64,8 @@ static const struct reg_sequence rt5514_patch[] = { {RT5514_ANA_CTRL_LDO10, 0x00028604}, {RT5514_ANA_CTRL_ADCFED, 0x00000800}, {RT5514_ASRC_IN_CTRL1, 0x00000003}, - {RT5514_DOWNFILTER0_CTRL3, 0x10000362}, - {RT5514_DOWNFILTER1_CTRL3, 0x10000362}, + {RT5514_DOWNFILTER0_CTRL3, 0x10000352}, + {RT5514_DOWNFILTER1_CTRL3, 0x10000352}, }; static const struct reg_default rt5514_reg[] = { @@ -92,10 +92,10 @@ static const struct reg_default rt5514_reg[] = { {RT5514_ASRC_IN_CTRL1, 0x00000003}, {RT5514_DOWNFILTER0_CTRL1, 0x00020c2f}, {RT5514_DOWNFILTER0_CTRL2, 0x00020c2f}, - {RT5514_DOWNFILTER0_CTRL3, 0x10000362}, + {RT5514_DOWNFILTER0_CTRL3, 0x10000352}, {RT5514_DOWNFILTER1_CTRL1, 0x00020c2f}, {RT5514_DOWNFILTER1_CTRL2, 0x00020c2f}, - {RT5514_DOWNFILTER1_CTRL3, 0x10000362}, + {RT5514_DOWNFILTER1_CTRL3, 0x10000352}, {RT5514_ANA_CTRL_LDO10, 0x00028604}, {RT5514_ANA_CTRL_LDO18_16, 0x02000345}, {RT5514_ANA_CTRL_ADC12, 0x0000a2a8}, -- cgit v1.2.3-70-g09d2 From 279fef50b607f9cee94f10bae84f6730e97ccd7c Mon Sep 17 00:00:00 2001 From: Edward Cragg Date: Fri, 27 Jul 2018 13:59:28 +0100 Subject: ASoC: tegra: i2s: Fix typo/broken macro Fix typo in macro TEGRA30_I2S_SLOT_CTRL_TOTAL_SLOTS_MASK. Signed-off-by: Edward Cragg Signed-off-by: Jorge Sanjuan Reviewed-by: Jon Hunter Signed-off-by: Mark Brown --- sound/soc/tegra/tegra30_i2s.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/tegra/tegra30_i2s.h b/sound/soc/tegra/tegra30_i2s.h index 774fc6ad2026..2e561e946de2 100644 --- a/sound/soc/tegra/tegra30_i2s.h +++ b/sound/soc/tegra/tegra30_i2s.h @@ -173,7 +173,7 @@ /* Number of slots in frame, minus 1 */ #define TEGRA30_I2S_SLOT_CTRL_TOTAL_SLOTS_SHIFT 16 #define TEGRA30_I2S_SLOT_CTRL_TOTAL_SLOTS_MASK_US 7 -#define TEGRA30_I2S_SLOT_CTRL_TOTAL_SLOTS_MASK (TEGRA30_I2S_SLOT_CTRL_TOTAL_SLOT_MASK_US << TEGRA30_I2S_SLOT_CTRL_TOTAL_SLOT_SHIFT) +#define TEGRA30_I2S_SLOT_CTRL_TOTAL_SLOTS_MASK (TEGRA30_I2S_SLOT_CTRL_TOTAL_SLOTS_MASK_US << TEGRA30_I2S_SLOT_CTRL_TOTAL_SLOTS_SHIFT) /* TDM mode slot enable bitmask */ #define TEGRA30_I2S_SLOT_CTRL_RX_SLOT_ENABLES_SHIFT 8 -- cgit v1.2.3-70-g09d2 From c9c9780d8fa526a124933134a7e4041fa09662f6 Mon Sep 17 00:00:00 2001 From: Ladislav Michl Date: Sat, 28 Jul 2018 14:30:17 +0200 Subject: ASoC: wm8988: fix typo in rate constraints Remove duplicated entry and add missing zero in rate constraints. Signed-off-by: Ladislav Michl Signed-off-by: Mark Brown --- sound/soc/codecs/wm8988.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8988.c b/sound/soc/codecs/wm8988.c index 62200117444b..6e52c6a8bab3 100644 --- a/sound/soc/codecs/wm8988.c +++ b/sound/soc/codecs/wm8988.c @@ -522,7 +522,7 @@ static inline int get_coeff(int mclk, int rate) /* The set of rates we can generate from the above for each SYSCLK */ static const unsigned int rates_12288[] = { - 8000, 12000, 16000, 24000, 24000, 32000, 48000, 96000, + 8000, 12000, 16000, 24000, 32000, 48000, 96000, }; static const struct snd_pcm_hw_constraint_list constraints_12288 = { @@ -540,7 +540,7 @@ static const struct snd_pcm_hw_constraint_list constraints_112896 = { }; static const unsigned int rates_12[] = { - 8000, 11025, 12000, 16000, 22050, 2400, 32000, 41100, 48000, + 8000, 11025, 12000, 16000, 22050, 24000, 32000, 41100, 48000, 48000, 88235, 96000, }; -- cgit v1.2.3-70-g09d2 From b1470d4ce77c2d60661c7d5325d4fb8063e15ff8 Mon Sep 17 00:00:00 2001 From: Ajit Pandey Date: Tue, 7 Aug 2018 18:30:42 +0100 Subject: ASoC: wm_adsp: Correct DSP pointer for preloader control The offset of the DSP core needs to be taken into account for the DSP preloader control get and put. Currently the dsp->preloaded variable will only ever be read/updated on the first DSP, whilst this doesn't affect the operation of the control the readback will be incorrect. Signed-off-by: Ajit Pandey Signed-off-by: Charles Keepax Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/codecs/wm_adsp.c | 8 ++++++-- 1 file changed, 6 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 4e7f8525449e..08dc82770273 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -2643,7 +2643,10 @@ int wm_adsp2_preloader_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol); - struct wm_adsp *dsp = snd_soc_component_get_drvdata(component); + struct wm_adsp *dsps = snd_soc_component_get_drvdata(component); + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + struct wm_adsp *dsp = &dsps[mc->shift - 1]; ucontrol->value.integer.value[0] = dsp->preloaded; @@ -2655,10 +2658,11 @@ int wm_adsp2_preloader_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol); - struct wm_adsp *dsp = snd_soc_component_get_drvdata(component); + struct wm_adsp *dsps = snd_soc_component_get_drvdata(component); struct snd_soc_dapm_context *dapm = snd_soc_component_get_dapm(component); struct soc_mixer_control *mc = (struct soc_mixer_control *)kcontrol->private_value; + struct wm_adsp *dsp = &dsps[mc->shift - 1]; char preload[32]; snprintf(preload, ARRAY_SIZE(preload), "DSP%u Preload", mc->shift); -- cgit v1.2.3-70-g09d2 From 0717edbdfed61b4c1e8291140f78882d3a481042 Mon Sep 17 00:00:00 2001 From: Ryan Lee Date: Tue, 7 Aug 2018 20:06:38 -0700 Subject: ASoC: max98373: Added software reset register to readable registers Signed-off-by: Ryan Lee Signed-off-by: Mark Brown --- sound/soc/codecs/max98373.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/soc/codecs/max98373.c b/sound/soc/codecs/max98373.c index a92586106932..92b7125ea169 100644 --- a/sound/soc/codecs/max98373.c +++ b/sound/soc/codecs/max98373.c @@ -488,6 +488,7 @@ static const DECLARE_TLV_DB_RANGE(max98373_bde_gain_tlv, static bool max98373_readable_register(struct device *dev, unsigned int reg) { switch (reg) { + case MAX98373_R2000_SW_RESET: case MAX98373_R2001_INT_RAW1 ... MAX98373_R200C_INT_EN3: case MAX98373_R2010_IRQ_CTRL: case MAX98373_R2014_THERM_WARN_THRESH -- cgit v1.2.3-70-g09d2