From 4623a614e87e4f2df08c83b5b9f68af394951dc9 Mon Sep 17 00:00:00 2001 From: Koro Chen Date: Tue, 23 Jun 2015 19:01:20 +0800 Subject: ASoC: mediatek: Fix unbalanced calls to runtime suspend/resume This adds call to runtime suspend in dev remove. It fixs the problem that suspend is not called in the case of CONFIG_PM=n. It also fixs build warning when CONFIG_PM=n. Signed-off-by: Koro Chen Signed-off-by: Mark Brown --- sound/soc/mediatek/mtk-afe-pcm.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound') diff --git a/sound/soc/mediatek/mtk-afe-pcm.c b/sound/soc/mediatek/mtk-afe-pcm.c index cc228db5fb76..9863da73dfe0 100644 --- a/sound/soc/mediatek/mtk-afe-pcm.c +++ b/sound/soc/mediatek/mtk-afe-pcm.c @@ -1199,6 +1199,8 @@ err_pm_disable: static int mtk_afe_pcm_dev_remove(struct platform_device *pdev) { pm_runtime_disable(&pdev->dev); + if (!pm_runtime_status_suspended(&pdev->dev)) + mtk_afe_runtime_suspend(&pdev->dev); snd_soc_unregister_component(&pdev->dev); snd_soc_unregister_platform(&pdev->dev); return 0; -- cgit v1.2.3 From ffacb48e5a4665d3d7286babb38a5af855a36bc0 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Fri, 26 Jun 2015 10:39:43 +0100 Subject: ASoC: dapm: Fix deadlock on auto-disable mux controls The commit 02aa78abec6e ("ASoC: DAPM: Add APIs to create individual DAPM controls.") added locking to the snd_soc_dapm_new_control function but did not update the call to snd_soc_dapm_new_control in the auto-disable mux code, this appears to be because the patches were sent at fairly similar times. This patch change the call in the auto-disable mux code to use the new snd_soc_dapm_new_control_unlocked function instead. Signed-off-by: Charles Keepax 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 aa327c92480c..a47a8ce7a5ea 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -389,8 +389,8 @@ static int dapm_kcontrol_data_alloc(struct snd_soc_dapm_widget *widget, data->value = template.on_val; - data->widget = snd_soc_dapm_new_control(widget->dapm, - &template); + data->widget = snd_soc_dapm_new_control_unlocked( + widget->dapm, &template); if (!data->widget) { ret = -ENOMEM; goto err_name; -- cgit v1.2.3 From 7084ffbff494669b06ceb457150c38887e26d2a3 Mon Sep 17 00:00:00 2001 From: Liam Girdwood Date: Mon, 29 Jun 2015 17:36:43 +0100 Subject: ASoC: topology: Fix TLV size calculation. TLV size calculation was incorrectly calculated. Fix this according to include/sound/tlv.h Signed-off-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/soc-topology.c | 21 ++++++++++----------- 1 file changed, 10 insertions(+), 11 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index d0960683c409..7a19df313fe8 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -580,27 +580,26 @@ static int soc_tplg_init_kcontrol(struct soc_tplg *tplg, } static int soc_tplg_create_tlv(struct soc_tplg *tplg, - struct snd_kcontrol_new *kc, u32 tlv_size) + struct snd_kcontrol_new *kc, struct snd_soc_tplg_ctl_tlv *tplg_tlv) { - struct snd_soc_tplg_ctl_tlv *tplg_tlv; struct snd_ctl_tlv *tlv; + int size; - if (tlv_size == 0) + if (tplg_tlv->count == 0) return 0; - tplg_tlv = (struct snd_soc_tplg_ctl_tlv *) tplg->pos; - tplg->pos += tlv_size; - - tlv = kzalloc(sizeof(*tlv) + tlv_size, GFP_KERNEL); + size = ((tplg_tlv->count + (sizeof(unsigned int) - 1)) & + ~(sizeof(unsigned int) - 1)); + tlv = kzalloc(sizeof(*tlv) + size, GFP_KERNEL); if (tlv == NULL) return -ENOMEM; dev_dbg(tplg->dev, " created TLV type %d size %d bytes\n", - tplg_tlv->numid, tplg_tlv->size); + tplg_tlv->numid, size); tlv->numid = tplg_tlv->numid; - tlv->length = tplg_tlv->size; - memcpy(tlv->tlv, tplg_tlv + 1, tplg_tlv->size); + tlv->length = size; + memcpy(&tlv->tlv[0], tplg_tlv->data, size); kc->tlv.p = (void *)tlv; return 0; @@ -773,7 +772,7 @@ static int soc_tplg_dmixer_create(struct soc_tplg *tplg, unsigned int count, } /* create any TLV data */ - soc_tplg_create_tlv(tplg, &kc, mc->hdr.tlv_size); + soc_tplg_create_tlv(tplg, &kc, &mc->tlv); /* register control here */ err = soc_tplg_add_kcontrol(tplg, &kc, -- cgit v1.2.3 From e50b1e06b79e9d51efbff9627b4dd407184ef43f Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Mon, 6 Jul 2015 17:01:24 +0200 Subject: ASoC: dapm: Lock during userspace access The DAPM lock must be held when accessing the DAPM graph status through sysfs or debugfs, otherwise concurrent changes to the graph can result in undefined behaviour. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/soc-dapm.c | 9 +++++++++ 1 file changed, 9 insertions(+) (limited to 'sound') diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index a47a8ce7a5ea..1779430013ea 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1952,6 +1952,7 @@ static ssize_t dapm_widget_power_read_file(struct file *file, size_t count, loff_t *ppos) { struct snd_soc_dapm_widget *w = file->private_data; + struct snd_soc_card *card = w->dapm->card; char *buf; int in, out; ssize_t ret; @@ -1961,6 +1962,8 @@ static ssize_t dapm_widget_power_read_file(struct file *file, if (!buf) return -ENOMEM; + mutex_lock(&card->dapm_mutex); + /* Supply widgets are not handled by is_connected_{input,output}_ep() */ if (w->is_supply) { in = 0; @@ -2007,6 +2010,8 @@ static ssize_t dapm_widget_power_read_file(struct file *file, p->sink->name); } + mutex_unlock(&card->dapm_mutex); + ret = simple_read_from_buffer(user_buf, count, ppos, buf, ret); kfree(buf); @@ -2281,11 +2286,15 @@ static ssize_t dapm_widget_show(struct device *dev, struct snd_soc_pcm_runtime *rtd = dev_get_drvdata(dev); int i, count = 0; + mutex_lock(&rtd->card->dapm_mutex); + for (i = 0; i < rtd->num_codecs; i++) { struct snd_soc_codec *codec = rtd->codec_dais[i]->codec; count += dapm_widget_show_codec(codec, buf + count); } + mutex_unlock(&rtd->card->dapm_mutex); + return count; } -- cgit v1.2.3 From 8bc76c8bf6e5d96985eb05afe1b94699d580eb68 Mon Sep 17 00:00:00 2001 From: "Fang, Yang A" Date: Mon, 6 Jul 2015 14:12:38 -0700 Subject: ASoC: Intel: fix incorrect widget name We should use "HiFi Playback" and "HiFi Capture".it will fix below err cht-bsw-max98090: ASoC: no sink widget found for AIF1 Playback cht-bsw-max98090: ASoC: Failed to add route ssp2 Tx -> direct -> AIF1 Playback cht-bsw-max98090: ASoC: no source widget found for AIF1 Capture cht-bsw-max98090: ASoC: Failed to add route AIF1 Capture -> direct -> ssp2 Rx Signed-off-by: Fang, Yang A Signed-off-by: Mark Brown --- sound/soc/intel/boards/cht_bsw_max98090_ti.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/intel/boards/cht_bsw_max98090_ti.c b/sound/soc/intel/boards/cht_bsw_max98090_ti.c index d604ee80eda4..70f832114a5a 100644 --- a/sound/soc/intel/boards/cht_bsw_max98090_ti.c +++ b/sound/soc/intel/boards/cht_bsw_max98090_ti.c @@ -69,12 +69,12 @@ static const struct snd_soc_dapm_route cht_audio_map[] = { {"Headphone", NULL, "HPR"}, {"Ext Spk", NULL, "SPKL"}, {"Ext Spk", NULL, "SPKR"}, - {"AIF1 Playback", NULL, "ssp2 Tx"}, + {"HiFi Playback", NULL, "ssp2 Tx"}, {"ssp2 Tx", NULL, "codec_out0"}, {"ssp2 Tx", NULL, "codec_out1"}, {"codec_in0", NULL, "ssp2 Rx" }, {"codec_in1", NULL, "ssp2 Rx" }, - {"ssp2 Rx", NULL, "AIF1 Capture"}, + {"ssp2 Rx", NULL, "HiFi Capture"}, }; static const struct snd_kcontrol_new cht_mc_controls[] = { -- cgit v1.2.3 From ebac95a9208e6b5f134df8518df1bfd1b3fee354 Mon Sep 17 00:00:00 2001 From: Juergen Borleis Date: Fri, 3 Jul 2015 12:39:36 +0200 Subject: ASoC: fsl-ssi: Fix bitclock calculation for master mode According to the datasheet 'pm', 'psr' and 'div2' should never be all 0. Since commit 541b03ad6cfe ("ASoC: fsl_ssi: Fix the incorrect limitation of the bit clock rate") this can happen, because for some bitclock rates 'pm' = 0 seems to be a valid choice but does not work due to hardware restrictions. This results into a bad hardware behaviour (slow audio for example). Feature tested on a i.MX25. Signed-off-by: Juergen Borleis Acked-by: Nicolin Chen Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_ssi.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index c7647e066cfd..c0b940e2019f 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -633,7 +633,7 @@ static int fsl_ssi_set_bclk(struct snd_pcm_substream *substream, sub *= 100000; do_div(sub, freq); - if (sub < savesub) { + if (sub < savesub && !(i == 0 && psr == 0 && div2 == 0)) { baudrate = tmprate; savesub = sub; pm = i; -- cgit v1.2.3 From 56e7366e43ca676dd28f0e91240a579ad41e9b71 Mon Sep 17 00:00:00 2001 From: Vinod Koul Date: Fri, 19 Jun 2015 23:55:26 +0530 Subject: ASoC: Intel: use CONFIG_SND_SOC for intel boards The Intel boards directory was under CONFIG_SND_SOC_INTEL_SST so the machines which don't need these were not allowed to be selected/compiled without enabling this symbol The machine should be allowed to selected by ASoC and then they should select rest of symbols required Reported-by: Michele Curti Signed-off-by: Vinod Koul Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/intel/Makefile | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/intel/Makefile b/sound/soc/intel/Makefile index 3853ec2ddbc7..6de5d5cd3280 100644 --- a/sound/soc/intel/Makefile +++ b/sound/soc/intel/Makefile @@ -7,4 +7,4 @@ obj-$(CONFIG_SND_SOC_INTEL_BAYTRAIL) += baytrail/ obj-$(CONFIG_SND_SST_MFLD_PLATFORM) += atom/ # Machine support -obj-$(CONFIG_SND_SOC_INTEL_SST) += boards/ +obj-$(CONFIG_SND_SOC) += boards/ -- cgit v1.2.3 From e18077b6e5dfe26e9fbbdc1fd1085a1701c24bea Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Wed, 8 Jul 2015 21:59:59 +0200 Subject: ASoC: dapm: Fix kcontrol widget name memory management The name field of the widget template is only used inside snd_soc_dapm_new_control_unlocked() which allocates a copy for the actual widget. This means we need to free the name allocated for the template in dapm_kcontrol_data_alloc() and not the name of the actual widget in dapm_kcontrol_free(). Otherwise we get a double free on the widget name and a memory leak on the template name. Fixes: 773da9b358bf ("ASoC: dapm: Append "Autodisable" to autodisable widget names") Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/soc-dapm.c | 10 ++++------ 1 file changed, 4 insertions(+), 6 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 1779430013ea..9f270c0308b7 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -358,9 +358,10 @@ static int dapm_kcontrol_data_alloc(struct snd_soc_dapm_widget *widget, data->widget = snd_soc_dapm_new_control_unlocked(widget->dapm, &template); + kfree(name); if (!data->widget) { ret = -ENOMEM; - goto err_name; + goto err_data; } } break; @@ -391,9 +392,10 @@ static int dapm_kcontrol_data_alloc(struct snd_soc_dapm_widget *widget, data->widget = snd_soc_dapm_new_control_unlocked( widget->dapm, &template); + kfree(name); if (!data->widget) { ret = -ENOMEM; - goto err_name; + goto err_data; } snd_soc_dapm_add_path(widget->dapm, data->widget, @@ -408,8 +410,6 @@ static int dapm_kcontrol_data_alloc(struct snd_soc_dapm_widget *widget, return 0; -err_name: - kfree(name); err_data: kfree(data); return ret; @@ -418,8 +418,6 @@ err_data: static void dapm_kcontrol_free(struct snd_kcontrol *kctl) { struct dapm_kcontrol_data *data = snd_kcontrol_chip(kctl); - if (data->widget) - kfree(data->widget->name); kfree(data->wlist); kfree(data); } -- cgit v1.2.3 From 2210438b6aae9668cf89c272fef935b83aedf81d Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Wed, 8 Jul 2015 22:14:48 +0200 Subject: ASoC: Free card DAPM context on snd_soc_instantiate_card() error path Make sure the to free the card DAPM context if snd_soc_instantiate_card() fails, otherwise the memory allocated for the DAPM widgets is leaked. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 3a4a5c0e3f97..0e1e69c7abd5 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1716,6 +1716,7 @@ card_probe_error: if (card->remove) card->remove(card); + snd_soc_dapm_free(&card->dapm); soc_cleanup_card_debugfs(card); snd_card_free(card->snd_card); -- cgit v1.2.3 From 94319ba10ecabc8f28129566d1f5793e3e7a0a79 Mon Sep 17 00:00:00 2001 From: Koro Chen Date: Thu, 9 Jul 2015 10:51:27 +0800 Subject: ASoC: mediatek: Use platform_of_node for machine drivers This replaces the platform_name in snd_soc_dai_link by device tree node. Signed-off-by: Koro Chen Signed-off-by: Mark Brown --- .../devicetree/bindings/sound/mt8173-max98090.txt | 2 ++ .../bindings/sound/mt8173-rt5650-rt5676.txt | 2 ++ sound/soc/mediatek/mt8173-max98090.c | 17 +++++++++++++---- sound/soc/mediatek/mt8173-rt5650-rt5676.c | 19 +++++++++++++++---- 4 files changed, 32 insertions(+), 8 deletions(-) (limited to 'sound') diff --git a/Documentation/devicetree/bindings/sound/mt8173-max98090.txt b/Documentation/devicetree/bindings/sound/mt8173-max98090.txt index 829bd26d17f8..519e97c8f1b8 100644 --- a/Documentation/devicetree/bindings/sound/mt8173-max98090.txt +++ b/Documentation/devicetree/bindings/sound/mt8173-max98090.txt @@ -3,11 +3,13 @@ MT8173 with MAX98090 CODEC Required properties: - compatible : "mediatek,mt8173-max98090" - mediatek,audio-codec: the phandle of the MAX98090 audio codec +- mediatek,platform: the phandle of MT8173 ASoC platform Example: sound { compatible = "mediatek,mt8173-max98090"; mediatek,audio-codec = <&max98090>; + mediatek,platform = <&afe>; }; diff --git a/Documentation/devicetree/bindings/sound/mt8173-rt5650-rt5676.txt b/Documentation/devicetree/bindings/sound/mt8173-rt5650-rt5676.txt index 61e98c976bd4..f205ce9e31dd 100644 --- a/Documentation/devicetree/bindings/sound/mt8173-rt5650-rt5676.txt +++ b/Documentation/devicetree/bindings/sound/mt8173-rt5650-rt5676.txt @@ -3,11 +3,13 @@ MT8173 with RT5650 RT5676 CODECS Required properties: - compatible : "mediatek,mt8173-rt5650-rt5676" - mediatek,audio-codec: the phandles of rt5650 and rt5676 codecs +- mediatek,platform: the phandle of MT8173 ASoC platform Example: sound { compatible = "mediatek,mt8173-rt5650-rt5676"; mediatek,audio-codec = <&rt5650 &rt5676>; + mediatek,platform = <&afe>; }; diff --git a/sound/soc/mediatek/mt8173-max98090.c b/sound/soc/mediatek/mt8173-max98090.c index 4d44b5803e55..2d2536af141f 100644 --- a/sound/soc/mediatek/mt8173-max98090.c +++ b/sound/soc/mediatek/mt8173-max98090.c @@ -103,7 +103,6 @@ static struct snd_soc_dai_link mt8173_max98090_dais[] = { .name = "MAX98090 Playback", .stream_name = "MAX98090 Playback", .cpu_dai_name = "DL1", - .platform_name = "11220000.mt8173-afe-pcm", .codec_name = "snd-soc-dummy", .codec_dai_name = "snd-soc-dummy-dai", .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, @@ -114,7 +113,6 @@ static struct snd_soc_dai_link mt8173_max98090_dais[] = { .name = "MAX98090 Capture", .stream_name = "MAX98090 Capture", .cpu_dai_name = "VUL", - .platform_name = "11220000.mt8173-afe-pcm", .codec_name = "snd-soc-dummy", .codec_dai_name = "snd-soc-dummy-dai", .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, @@ -125,7 +123,6 @@ static struct snd_soc_dai_link mt8173_max98090_dais[] = { { .name = "Codec", .cpu_dai_name = "I2S", - .platform_name = "11220000.mt8173-afe-pcm", .no_pcm = 1, .codec_dai_name = "HiFi", .init = mt8173_max98090_init, @@ -152,9 +149,21 @@ static struct snd_soc_card mt8173_max98090_card = { static int mt8173_max98090_dev_probe(struct platform_device *pdev) { struct snd_soc_card *card = &mt8173_max98090_card; - struct device_node *codec_node; + struct device_node *codec_node, *platform_node; int ret, i; + platform_node = of_parse_phandle(pdev->dev.of_node, + "mediatek,platform", 0); + if (!platform_node) { + dev_err(&pdev->dev, "Property 'platform' missing or invalid\n"); + return -EINVAL; + } + for (i = 0; i < card->num_links; i++) { + if (mt8173_max98090_dais[i].platform_name) + continue; + mt8173_max98090_dais[i].platform_of_node = platform_node; + } + codec_node = of_parse_phandle(pdev->dev.of_node, "mediatek,audio-codec", 0); if (!codec_node) { diff --git a/sound/soc/mediatek/mt8173-rt5650-rt5676.c b/sound/soc/mediatek/mt8173-rt5650-rt5676.c index 094055323059..6f52eca05e26 100644 --- a/sound/soc/mediatek/mt8173-rt5650-rt5676.c +++ b/sound/soc/mediatek/mt8173-rt5650-rt5676.c @@ -138,7 +138,6 @@ static struct snd_soc_dai_link mt8173_rt5650_rt5676_dais[] = { .name = "rt5650_rt5676 Playback", .stream_name = "rt5650_rt5676 Playback", .cpu_dai_name = "DL1", - .platform_name = "11220000.mt8173-afe-pcm", .codec_name = "snd-soc-dummy", .codec_dai_name = "snd-soc-dummy-dai", .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, @@ -149,7 +148,6 @@ static struct snd_soc_dai_link mt8173_rt5650_rt5676_dais[] = { .name = "rt5650_rt5676 Capture", .stream_name = "rt5650_rt5676 Capture", .cpu_dai_name = "VUL", - .platform_name = "11220000.mt8173-afe-pcm", .codec_name = "snd-soc-dummy", .codec_dai_name = "snd-soc-dummy-dai", .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, @@ -161,7 +159,6 @@ static struct snd_soc_dai_link mt8173_rt5650_rt5676_dais[] = { { .name = "Codec", .cpu_dai_name = "I2S", - .platform_name = "11220000.mt8173-afe-pcm", .no_pcm = 1, .codecs = mt8173_rt5650_rt5676_codecs, .num_codecs = 2, @@ -209,7 +206,21 @@ static struct snd_soc_card mt8173_rt5650_rt5676_card = { static int mt8173_rt5650_rt5676_dev_probe(struct platform_device *pdev) { struct snd_soc_card *card = &mt8173_rt5650_rt5676_card; - int ret; + struct device_node *platform_node; + int i, ret; + + platform_node = of_parse_phandle(pdev->dev.of_node, + "mediatek,platform", 0); + if (!platform_node) { + dev_err(&pdev->dev, "Property 'platform' missing or invalid\n"); + return -EINVAL; + } + + for (i = 0; i < card->num_links; i++) { + if (mt8173_rt5650_rt5676_dais[i].platform_name) + continue; + mt8173_rt5650_rt5676_dais[i].platform_of_node = platform_node; + } mt8173_rt5650_rt5676_codecs[0].of_node = of_parse_phandle(pdev->dev.of_node, "mediatek,audio-codec", 0); -- cgit v1.2.3 From a18da49ff3ec849c4584ae3abe2b83ff50705ace Mon Sep 17 00:00:00 2001 From: Krzysztof Kozlowski Date: Thu, 9 Jul 2015 22:19:28 +0900 Subject: ASoC: zx: i2s: Fix devm_ioremap_resource return value check Value returned by devm_ioremap_resource() was checked for non-NULL but devm_ioremap_resource() returns IOMEM_ERR_PTR, not NULL. In case of error this could lead to dereference of ERR_PTR. Signed-off-by: Krzysztof Kozlowski Reviewed-by: Jun Nie Signed-off-by: Mark Brown --- sound/soc/zte/zx296702-i2s.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/zte/zx296702-i2s.c b/sound/soc/zte/zx296702-i2s.c index 98d96e1b17e0..1930c42e1f55 100644 --- a/sound/soc/zte/zx296702-i2s.c +++ b/sound/soc/zte/zx296702-i2s.c @@ -393,9 +393,9 @@ static int zx_i2s_probe(struct platform_device *pdev) res = platform_get_resource(pdev, IORESOURCE_MEM, 0); zx_i2s->mapbase = res->start; zx_i2s->reg_base = devm_ioremap_resource(&pdev->dev, res); - if (!zx_i2s->reg_base) { + if (IS_ERR(zx_i2s->reg_base)) { dev_err(&pdev->dev, "ioremap failed!\n"); - return -EIO; + return PTR_ERR(zx_i2s->reg_base); } writel_relaxed(0, zx_i2s->reg_base + ZX_I2S_FIFO_CTRL); -- cgit v1.2.3 From 2dfbe9afcda3243dd309826cbbb8ae9e9f602006 Mon Sep 17 00:00:00 2001 From: Krzysztof Kozlowski Date: Thu, 9 Jul 2015 22:19:29 +0900 Subject: ASoC: zx: spdif: Fix devm_ioremap_resource return value check Value returned by devm_ioremap_resource() was checked for non-NULL but devm_ioremap_resource() returns IOMEM_ERR_PTR, not NULL. In case of error this could lead to dereference of ERR_PTR. Signed-off-by: Krzysztof Kozlowski Reviewed-by: Jun Nie Signed-off-by: Mark Brown --- sound/soc/zte/zx296702-spdif.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/zte/zx296702-spdif.c b/sound/soc/zte/zx296702-spdif.c index 11a0e46a1156..26265ce4caca 100644 --- a/sound/soc/zte/zx296702-spdif.c +++ b/sound/soc/zte/zx296702-spdif.c @@ -322,9 +322,9 @@ static int zx_spdif_probe(struct platform_device *pdev) res = platform_get_resource(pdev, IORESOURCE_MEM, 0); zx_spdif->mapbase = res->start; zx_spdif->reg_base = devm_ioremap_resource(&pdev->dev, res); - if (!zx_spdif->reg_base) { + if (IS_ERR(zx_spdif->reg_base)) { dev_err(&pdev->dev, "ioremap failed!\n"); - return -EIO; + return PTR_ERR(zx_spdif->reg_base); } zx_spdif_dev_init(zx_spdif->reg_base); -- cgit v1.2.3 From 2c57d478018551f2a0983293413f2f198e49ec23 Mon Sep 17 00:00:00 2001 From: Jeeja KP Date: Tue, 14 Jul 2015 13:10:47 +0530 Subject: ASoC: topology: Fix to add dapm mixer info Mixer control for widgets can't be created if the info is NULL. So assign the correct info for this. Signed-off-by: Jeeja KP Signed-off-by: Subhransu S. Prusty Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/soc-topology.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index 7a19df313fe8..59ac211f8fe7 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -144,7 +144,7 @@ static const struct snd_soc_tplg_kcontrol_ops io_ops[] = { {SND_SOC_TPLG_CTL_STROBE, snd_soc_get_strobe, snd_soc_put_strobe, NULL}, {SND_SOC_TPLG_DAPM_CTL_VOLSW, snd_soc_dapm_get_volsw, - snd_soc_dapm_put_volsw, NULL}, + snd_soc_dapm_put_volsw, snd_soc_info_volsw}, {SND_SOC_TPLG_DAPM_CTL_ENUM_DOUBLE, snd_soc_dapm_get_enum_double, snd_soc_dapm_put_enum_double, snd_soc_info_enum_double}, {SND_SOC_TPLG_DAPM_CTL_ENUM_VIRT, snd_soc_dapm_get_enum_double, -- cgit v1.2.3 From 412efa73dcd3bd03c1838c91e094533a95529039 Mon Sep 17 00:00:00 2001 From: Shilpa Sreeramalu Date: Wed, 15 Jul 2015 07:58:09 -0700 Subject: ASoC: Intel: Get correct usage_count value to load firmware The usage_count variable was read before it was set to the correct value, due to which the firmware load was failing. Because of this IPC messages sent to the firmware were timing out causing a delay of about 1 second while playing audio from the internal speakers. With this patch the usage_count is read after the function call pm_runtime_get_sync which will increment the usage_count variable and the firmware load is successful and all the IPC messages are processed correctly. Signed-off-by: Shilpa Sreeramalu Signed-off-by: Fang, Yang A Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/intel/atom/sst/sst_drv_interface.c | 14 +++++++------- 1 file changed, 7 insertions(+), 7 deletions(-) (limited to 'sound') diff --git a/sound/soc/intel/atom/sst/sst_drv_interface.c b/sound/soc/intel/atom/sst/sst_drv_interface.c index 620da1d1b9e3..0e0e4d9c021f 100644 --- a/sound/soc/intel/atom/sst/sst_drv_interface.c +++ b/sound/soc/intel/atom/sst/sst_drv_interface.c @@ -42,6 +42,11 @@ #define MIN_FRAGMENT_SIZE (50 * 1024) #define MAX_FRAGMENT_SIZE (1024 * 1024) #define SST_GET_BYTES_PER_SAMPLE(pcm_wd_sz) (((pcm_wd_sz + 15) >> 4) << 1) +#ifdef CONFIG_PM +#define GET_USAGE_COUNT(dev) (atomic_read(&dev->power.usage_count)) +#else +#define GET_USAGE_COUNT(dev) 1 +#endif int free_stream_context(struct intel_sst_drv *ctx, unsigned int str_id) { @@ -141,15 +146,9 @@ static int sst_power_control(struct device *dev, bool state) int ret = 0; int usage_count = 0; -#ifdef CONFIG_PM - usage_count = atomic_read(&dev->power.usage_count); -#else - usage_count = 1; -#endif - if (state == true) { ret = pm_runtime_get_sync(dev); - + usage_count = GET_USAGE_COUNT(dev); dev_dbg(ctx->dev, "Enable: pm usage count: %d\n", usage_count); if (ret < 0) { dev_err(ctx->dev, "Runtime get failed with err: %d\n", ret); @@ -164,6 +163,7 @@ static int sst_power_control(struct device *dev, bool state) } } } else { + usage_count = GET_USAGE_COUNT(dev); dev_dbg(ctx->dev, "Disable: pm usage count: %d\n", usage_count); return sst_pm_runtime_put(ctx); } -- cgit v1.2.3 From f2a5ded38592e5936a099ea6535ad5d3addcbc9d Mon Sep 17 00:00:00 2001 From: Nicolas Boichat Date: Fri, 17 Jul 2015 10:36:57 +0800 Subject: ASoC: rt5645: Check if codec is initialized in workqueue handler This fixes kernel panic on boot, if rt5645->codec is NULL when rt5645_jack_detect_work is first called. rt5645_jack_detect_work needs rt5645->codec to be initialized to setup dapm pins. Also, reporting jack events is useless, as the jacks cannot be set before the codec is ready. Since we manually call the interrupt handler in rt5645_set_jack_detect, the initial jack state will be reported correctly, and dapm pins will be setup at that time. Signed-off-by: Nicolas Boichat Signed-off-by: Mark Brown --- sound/soc/codecs/rt5645.c | 3 +++ 1 file changed, 3 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 9ce311e088fc..e9cc3aae5366 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -2943,6 +2943,9 @@ static int rt5645_irq_detection(struct rt5645_priv *rt5645) { int val, btn_type, gpio_state = 0, report = 0; + if (!rt5645->codec) + return -EINVAL; + switch (rt5645->pdata.jd_mode) { case 0: /* Not using rt5645 JD */ if (rt5645->gpiod_hp_det) { -- cgit v1.2.3 From 67756e3191c90e7c0b94b8b2fb63de255b6cd337 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 17 Jul 2015 15:22:33 +0200 Subject: ALSA: pcm: Fix lockdep warning with nonatomic PCM ops With the nonatomic PCM ops, the system may spew lockdep warnings like: ============================================= [ INFO: possible recursive locking detected ] 4.2.0-rc1-jeejaval3 #12 Not tainted --------------------------------------------- aplay/4029 is trying to acquire lock: (snd_pcm_link_rwsem){.+.+.+}, at: [] snd_pcm_stream_lock+0x43/0x60 but task is already holding lock: (snd_pcm_link_rwsem){.+.+.+}, at: [] snd_pcm_action_nonatomic+0x29/0x80 other info that might help us debug this: Possible unsafe locking scenario: CPU0 ---- lock(snd_pcm_link_rwsem); lock(snd_pcm_link_rwsem); Although this is false-positive as the rwsem is taken always as read-only for these code paths, it's certainly annoying to see this at any occasion. A simple fix is to use down_read_nested() in snd_pcm_stream_lock() that can be called inside another lock. Reported-by: Vinod Koul Reported-by: Jeeja Kp Tested-by: Jeeja Kp Cc: Signed-off-by: Takashi Iwai --- sound/core/pcm_native.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index d126c03361ae..75888dd38a7f 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -85,7 +85,7 @@ static DECLARE_RWSEM(snd_pcm_link_rwsem); void snd_pcm_stream_lock(struct snd_pcm_substream *substream) { if (substream->pcm->nonatomic) { - down_read(&snd_pcm_link_rwsem); + down_read_nested(&snd_pcm_link_rwsem, SINGLE_DEPTH_NESTING); mutex_lock(&substream->self_group.mutex); } else { read_lock(&snd_pcm_link_rwlock); -- cgit v1.2.3 From 25e5eaf19962bd48788371e4f516bdd89ce248bc Mon Sep 17 00:00:00 2001 From: Christophe JAILLET Date: Fri, 17 Jul 2015 20:33:21 +0200 Subject: ALSA: sparc: Add missing kfree in error path If 'of_ioremap' fails, then 'amd' should be freed, otherwise, there is a memory leak. Signed-off-by: Christophe JAILLET Signed-off-by: Takashi Iwai --- sound/sparc/amd7930.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/sparc/amd7930.c b/sound/sparc/amd7930.c index 1b1a89e80d13..784ceb85b2d9 100644 --- a/sound/sparc/amd7930.c +++ b/sound/sparc/amd7930.c @@ -956,6 +956,7 @@ static int snd_amd7930_create(struct snd_card *card, if (!amd->regs) { snd_printk(KERN_ERR "amd7930-%d: Unable to map chip registers.\n", dev); + kfree(amd); return -EIO; } -- cgit v1.2.3 From 0420694dddeb9e269a1ab2129a0119a5cea294a4 Mon Sep 17 00:00:00 2001 From: Mateusz Sylwestrzak Date: Sun, 19 Jul 2015 17:38:56 +0200 Subject: ALSA: hda - Add headset mic support for Acer Aspire V5-573G Acer Aspire V5 with the ALC282 codec is given the wrong value for the 0x19 PIN by the laptop's BIOS. Overriding it with the correct value adds support for the headset microphone which would not otherwise be visible in the system. The fix is based on commit 7819717b1134 with a similar quirk for Acer Aspire with the ALC269 codec. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=96201 Cc: Signed-off-by: Mateusz Sylwestrzak Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index d35cf506a7db..caba66b2311c 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5061,7 +5061,7 @@ static const struct hda_fixup alc269_fixups[] = { { 0x14, 0x90170110 }, { 0x17, 0x40000008 }, { 0x18, 0x411111f0 }, - { 0x19, 0x411111f0 }, + { 0x19, 0x01a1913c }, { 0x1a, 0x411111f0 }, { 0x1b, 0x411111f0 }, { 0x1d, 0x40f89b2d }, -- cgit v1.2.3 From 033ea349a7cd1aa15357cd6575de35188fc85b9a Mon Sep 17 00:00:00 2001 From: David Henningsson Date: Thu, 16 Jul 2015 10:39:24 +0200 Subject: ALSA: hda - Fix Skylake codec timeout When the controller is powered up but the HDMI codec is powered down on Skylake, the power well is turned off. When the codec is then powered up again, we need to poke the codec a little extra to make sure it wakes up. Otherwise we'll get sad "no response from codec" messages and broken audio. This also changes azx_runtime_resume to actually call snd_hdac_set_codec_wakeup for Skylake (before STATETS read). (Otherwise it would only have been called for Haswell and Broadwell, which both do not need it, so this probably was not the author's intention.) Signed-off-by: David Henningsson Reviewed-by: Libin Yang Signed-off-by: Takashi Iwai --- sound/hda/hdac_i915.c | 5 ++++- sound/pci/hda/hda_intel.c | 18 ++++++++++-------- 2 files changed, 14 insertions(+), 9 deletions(-) (limited to 'sound') diff --git a/sound/hda/hdac_i915.c b/sound/hda/hdac_i915.c index 442500e06b7c..5676b849379d 100644 --- a/sound/hda/hdac_i915.c +++ b/sound/hda/hdac_i915.c @@ -56,8 +56,11 @@ int snd_hdac_display_power(struct hdac_bus *bus, bool enable) enable ? "enable" : "disable"); if (enable) { - if (!bus->i915_power_refcount++) + if (!bus->i915_power_refcount++) { acomp->ops->get_power(acomp->dev); + snd_hdac_set_codec_wakeup(bus, true); + snd_hdac_set_codec_wakeup(bus, false); + } } else { WARN_ON(!bus->i915_power_refcount); if (!--bus->i915_power_refcount) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index ca151b45eeef..996223706676 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -979,14 +979,16 @@ static int azx_runtime_resume(struct device *dev) if (!azx_has_pm_runtime(chip)) return 0; - if (chip->driver_caps & AZX_DCAPS_I915_POWERWELL - && hda->need_i915_power) { - bus = azx_bus(chip); - snd_hdac_display_power(bus, true); - haswell_set_bclk(hda); - /* toggle codec wakeup bit for STATESTS read */ - snd_hdac_set_codec_wakeup(bus, true); - snd_hdac_set_codec_wakeup(bus, false); + if (chip->driver_caps & AZX_DCAPS_I915_POWERWELL) { + bus = azx_bus(chip); + if (hda->need_i915_power) { + snd_hdac_display_power(bus, true); + haswell_set_bclk(hda); + } else { + /* toggle codec wakeup bit for STATESTS read */ + snd_hdac_set_codec_wakeup(bus, true); + snd_hdac_set_codec_wakeup(bus, false); + } } /* Read STATESTS before controller reset */ -- cgit v1.2.3 From 5022813ddb28b7679e8285812d52aaeb7e1e7657 Mon Sep 17 00:00:00 2001 From: Maruthi Srinivas Bayyavarapu Date: Mon, 20 Jul 2015 19:56:18 +0530 Subject: ALSA: hda: add new AMD PCI IDs with proper driver caps Fixes audio problems on newer asics Signed-off-by: Maruthi Bayyavarapu Cc: stable@vger.kernel.org Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 8 ++++++++ 1 file changed, 8 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 996223706676..f581b12211e8 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2184,6 +2184,8 @@ static const struct pci_device_id azx_ids[] = { /* ATI HDMI */ { PCI_DEVICE(0x1002, 0x1308), .driver_data = AZX_DRIVER_ATIHDMI_NS | AZX_DCAPS_PRESET_ATI_HDMI_NS }, + { PCI_DEVICE(0x1002, 0x157a), + .driver_data = AZX_DRIVER_ATIHDMI_NS | AZX_DCAPS_PRESET_ATI_HDMI_NS }, { PCI_DEVICE(0x1002, 0x793b), .driver_data = AZX_DRIVER_ATIHDMI | AZX_DCAPS_PRESET_ATI_HDMI }, { PCI_DEVICE(0x1002, 0x7919), @@ -2238,8 +2240,14 @@ static const struct pci_device_id azx_ids[] = { .driver_data = AZX_DRIVER_ATIHDMI_NS | AZX_DCAPS_PRESET_ATI_HDMI_NS }, { PCI_DEVICE(0x1002, 0xaab0), .driver_data = AZX_DRIVER_ATIHDMI_NS | AZX_DCAPS_PRESET_ATI_HDMI_NS }, + { PCI_DEVICE(0x1002, 0xaac0), + .driver_data = AZX_DRIVER_ATIHDMI_NS | AZX_DCAPS_PRESET_ATI_HDMI_NS }, { PCI_DEVICE(0x1002, 0xaac8), .driver_data = AZX_DRIVER_ATIHDMI_NS | AZX_DCAPS_PRESET_ATI_HDMI_NS }, + { PCI_DEVICE(0x1002, 0xaad8), + .driver_data = AZX_DRIVER_ATIHDMI_NS | AZX_DCAPS_PRESET_ATI_HDMI_NS }, + { PCI_DEVICE(0x1002, 0xaae8), + .driver_data = AZX_DRIVER_ATIHDMI_NS | AZX_DCAPS_PRESET_ATI_HDMI_NS }, /* VIA VT8251/VT8237A */ { PCI_DEVICE(0x1106, 0x3288), .driver_data = AZX_DRIVER_VIA | AZX_DCAPS_POSFIX_VIA }, -- cgit v1.2.3 From 6c3d91193d829bf58a35a10650415b05a736ca6c Mon Sep 17 00:00:00 2001 From: Aaron Plattner Date: Mon, 20 Jul 2015 17:14:14 -0700 Subject: ALSA: hda - Add new GPU codec ID 0x10de007d to snd-hda Vendor ID 0x10de007d is used by a yet-to-be-named GPU chip. This chip also has the 2-ch audio swapping bug, so patch_nvhdmi is appropriate here. Signed-off-by: Aaron Plattner Cc: Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_hdmi.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 95158914cc6c..a97db5fc8a15 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -3512,6 +3512,7 @@ static const struct hda_codec_preset snd_hda_preset_hdmi[] = { { .id = 0x10de0070, .name = "GPU 70 HDMI/DP", .patch = patch_nvhdmi }, { .id = 0x10de0071, .name = "GPU 71 HDMI/DP", .patch = patch_nvhdmi }, { .id = 0x10de0072, .name = "GPU 72 HDMI/DP", .patch = patch_nvhdmi }, +{ .id = 0x10de007d, .name = "GPU 7d HDMI/DP", .patch = patch_nvhdmi }, { .id = 0x10de8001, .name = "MCP73 HDMI", .patch = patch_nvhdmi_2ch }, { .id = 0x11069f80, .name = "VX900 HDMI/DP", .patch = patch_via_hdmi }, { .id = 0x11069f81, .name = "VX900 HDMI/DP", .patch = patch_via_hdmi }, @@ -3576,6 +3577,7 @@ MODULE_ALIAS("snd-hda-codec-id:10de0067"); MODULE_ALIAS("snd-hda-codec-id:10de0070"); MODULE_ALIAS("snd-hda-codec-id:10de0071"); MODULE_ALIAS("snd-hda-codec-id:10de0072"); +MODULE_ALIAS("snd-hda-codec-id:10de007d"); MODULE_ALIAS("snd-hda-codec-id:10de8001"); MODULE_ALIAS("snd-hda-codec-id:11069f80"); MODULE_ALIAS("snd-hda-codec-id:11069f81"); -- cgit v1.2.3 From a798c24a69b64f09e2d323ac8155a36373e5d5fd Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Tue, 21 Jul 2015 11:51:35 +0200 Subject: ASoC: dapm: Don't add prefix to widget stream name Commit fdb6eb0a1287 ("ASoC: dapm: Modify widget stream name according to prefix") fixed the case where a DAPM route between a DAI widget and a DAC/ADC/AIF widget with a matching stream name was not created when the DAPM context was using a prefix. Unfortunately the patch introduced a few issues on its own like leaking the dynamically allocated stream name memory and also not checking whether the allocation succeeded in the first place. It is also incomplete in that it still does not handle the case where stream name of the widget is a substring of the stream name of the DAI, which is explicitly allowed and works fine if no DAPM prefix is used. Revert the commit and take a slightly different approach to solving the issue. Instead of comparing the widget's stream name to the name of the DAI widget compare it to the stream name of the DAI widget. The stream name of the DAI widget is identical to the name of the DAI widget except that it wont have the DAPM prefix added. So this approach behaves identical regardless to whether the DAPM context uses a prefix or not. We don't have to worry about potentially matching with a widget with the same stream name, but from a different DAPM context with a different prefix, since the code already makes sure that both the DAI widget and the matched widget are from the same DAPM context. Fixes: fdb6eb0a1287 ("ASoC: dapm: Modify widget stream name according to prefix") Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/soc-dapm.c | 12 +++--------- 1 file changed, 3 insertions(+), 9 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 9f270c0308b7..e0de8072c514 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -3341,16 +3341,10 @@ snd_soc_dapm_new_control_unlocked(struct snd_soc_dapm_context *dapm, } prefix = soc_dapm_prefix(dapm); - if (prefix) { + if (prefix) w->name = kasprintf(GFP_KERNEL, "%s %s", prefix, widget->name); - if (widget->sname) - w->sname = kasprintf(GFP_KERNEL, "%s %s", prefix, - widget->sname); - } else { + else w->name = kasprintf(GFP_KERNEL, "%s", widget->name); - if (widget->sname) - w->sname = kasprintf(GFP_KERNEL, "%s", widget->sname); - } if (w->name == NULL) { kfree(w); return NULL; @@ -3799,7 +3793,7 @@ int snd_soc_dapm_link_dai_widgets(struct snd_soc_card *card) break; } - if (!w->sname || !strstr(w->sname, dai_w->name)) + if (!w->sname || !strstr(w->sname, dai_w->sname)) continue; if (dai_w->id == snd_soc_dapm_dai_in) { -- cgit v1.2.3 From 21e9d017b88ea0baa367ef0b6516d794fa23e85e Mon Sep 17 00:00:00 2001 From: Hui Wang Date: Wed, 22 Jul 2015 10:33:34 +0800 Subject: ALSA: hda - remove one pin from ALC292_STANDARD_PINS One more Dell laptop with alc293 codec needs ALC293_FIXUP_DELL1_MIC_NO_PRESENCE, but the pin 0x1e does not match the corresponding one in the ALC292_STANDARD_PINS. To use this macro for this machine, we need to remove pin 0x1e from it. BugLink: https://bugs.launchpad.net/bugs/1476888 Cc: Signed-off-by: Hui Wang Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 26 +++++++++++++++++++------- 1 file changed, 19 insertions(+), 7 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index caba66b2311c..ee6f13af647d 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5430,8 +5430,7 @@ static const struct hda_model_fixup alc269_fixup_models[] = { {0x15, 0x0221401f}, \ {0x1a, 0x411111f0}, \ {0x1b, 0x411111f0}, \ - {0x1d, 0x40700001}, \ - {0x1e, 0x411111f0} + {0x1d, 0x40700001} #define ALC298_STANDARD_PINS \ {0x18, 0x411111f0}, \ @@ -5690,35 +5689,48 @@ static const struct snd_hda_pin_quirk alc269_pin_fixup_tbl[] = { {0x13, 0x411111f0}, {0x16, 0x01014020}, {0x18, 0x411111f0}, - {0x19, 0x01a19030}), + {0x19, 0x01a19030}, + {0x1e, 0x411111f0}), SND_HDA_PIN_QUIRK(0x10ec0292, 0x1028, "Dell", ALC269_FIXUP_DELL2_MIC_NO_PRESENCE, ALC292_STANDARD_PINS, {0x12, 0x90a60140}, {0x13, 0x411111f0}, {0x16, 0x01014020}, {0x18, 0x02a19031}, - {0x19, 0x01a1903e}), + {0x19, 0x01a1903e}, + {0x1e, 0x411111f0}), SND_HDA_PIN_QUIRK(0x10ec0292, 0x1028, "Dell", ALC269_FIXUP_DELL3_MIC_NO_PRESENCE, ALC292_STANDARD_PINS, {0x12, 0x90a60140}, {0x13, 0x411111f0}, {0x16, 0x411111f0}, {0x18, 0x411111f0}, - {0x19, 0x411111f0}), + {0x19, 0x411111f0}, + {0x1e, 0x411111f0}), SND_HDA_PIN_QUIRK(0x10ec0293, 0x1028, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE, ALC292_STANDARD_PINS, {0x12, 0x40000000}, {0x13, 0x90a60140}, {0x16, 0x21014020}, {0x18, 0x411111f0}, - {0x19, 0x21a19030}), + {0x19, 0x21a19030}, + {0x1e, 0x411111f0}), SND_HDA_PIN_QUIRK(0x10ec0293, 0x1028, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE, ALC292_STANDARD_PINS, {0x12, 0x40000000}, {0x13, 0x90a60140}, {0x16, 0x411111f0}, {0x18, 0x411111f0}, - {0x19, 0x411111f0}), + {0x19, 0x411111f0}, + {0x1e, 0x411111f0}), + SND_HDA_PIN_QUIRK(0x10ec0293, 0x1028, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE, + ALC292_STANDARD_PINS, + {0x12, 0x40000000}, + {0x13, 0x90a60140}, + {0x16, 0x21014020}, + {0x18, 0x411111f0}, + {0x19, 0x21a19030}, + {0x1e, 0x411111ff}), SND_HDA_PIN_QUIRK(0x10ec0298, 0x1028, "Dell", ALC298_FIXUP_DELL1_MIC_NO_PRESENCE, ALC298_STANDARD_PINS, {0x12, 0x90a60130}, -- cgit v1.2.3 From cba59972a1191a0c1647a52fe745eed7a4b34b38 Mon Sep 17 00:00:00 2001 From: David Henningsson Date: Wed, 22 Jul 2015 10:00:25 +0200 Subject: ALSA: hda - Add headset mic pin quirk for a Dell device Without this patch, the headset mic will not work on this machine. BugLink: https://bugs.launchpad.net/bugs/1476987 Signed-off-by: David Henningsson Cc: Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 11 +++++++++++ 1 file changed, 11 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index ee6f13af647d..742fc626f9e1 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5461,6 +5461,17 @@ static const struct snd_hda_pin_quirk alc269_pin_fixup_tbl[] = { {0x17, 0x40000000}, {0x1d, 0x40700001}, {0x21, 0x02211030}), + SND_HDA_PIN_QUIRK(0x10ec0255, 0x1028, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE, + {0x12, 0x40000000}, + {0x14, 0x90170130}, + {0x17, 0x411111f0}, + {0x18, 0x411111f0}, + {0x19, 0x411111f0}, + {0x1a, 0x411111f0}, + {0x1b, 0x01014020}, + {0x1d, 0x4054c029}, + {0x1e, 0x411111f0}, + {0x21, 0x0221103f}), SND_HDA_PIN_QUIRK(0x10ec0255, 0x1028, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE, {0x12, 0x90a60160}, {0x14, 0x90170120}, -- cgit v1.2.3 From b101acfabc9377469af3abfb7cb63112da367284 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Wed, 22 Jul 2015 11:27:45 +0800 Subject: ASoC: sgtl5000: Fix up define for SGTL5000_SMALL_POP Currently, below code actually does not update any bit because SGTL5000_SMALL_POP is 0. snd_soc_update_bits(codec, SGTL5000_CHIP_REF_CTRL, SGTL5000_SMALL_POP, 1); The SGTL5000_SMALL_POP should be BIT(0) rather than 0, fix it. Signed-off-by: Axel Lin Acked-By: Alexander Stein Signed-off-by: Mark Brown --- sound/soc/codecs/sgtl5000.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/sgtl5000.h b/sound/soc/codecs/sgtl5000.h index bd7a344bf8c5..1c317de26176 100644 --- a/sound/soc/codecs/sgtl5000.h +++ b/sound/soc/codecs/sgtl5000.h @@ -275,7 +275,7 @@ #define SGTL5000_BIAS_CTRL_MASK 0x000e #define SGTL5000_BIAS_CTRL_SHIFT 1 #define SGTL5000_BIAS_CTRL_WIDTH 3 -#define SGTL5000_SMALL_POP 0 +#define SGTL5000_SMALL_POP 1 /* * SGTL5000_CHIP_MIC_CTRL -- cgit v1.2.3 From a6c2a32ac83567f15e9af3dcbc73148ce68b2ced Mon Sep 17 00:00:00 2001 From: Ben Zhang Date: Tue, 21 Jul 2015 14:46:26 -0700 Subject: ASoC: ssm4567: Keep TDM_BCLKS in ssm4567_set_dai_fmt The regmap_write in ssm4567_set_dai_fmt accidentally clears the TDM_BCLKS field which was set earlier by ssm4567_set_tdm_slot. This patch fixes it by using regmap_update_bits with proper mask. Signed-off-by: Ben Zhang Acked-by: Lars-Peter Clausen Acked-by: Anatol Pomozov Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/codecs/ssm4567.c | 8 +++++++- 1 file changed, 7 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/ssm4567.c b/sound/soc/codecs/ssm4567.c index 938d2cb6d78b..84a4f5ad8064 100644 --- a/sound/soc/codecs/ssm4567.c +++ b/sound/soc/codecs/ssm4567.c @@ -315,7 +315,13 @@ static int ssm4567_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) if (invert_fclk) ctrl1 |= SSM4567_SAI_CTRL_1_FSYNC; - return regmap_write(ssm4567->regmap, SSM4567_REG_SAI_CTRL_1, ctrl1); + return regmap_update_bits(ssm4567->regmap, SSM4567_REG_SAI_CTRL_1, + SSM4567_SAI_CTRL_1_BCLK | + SSM4567_SAI_CTRL_1_FSYNC | + SSM4567_SAI_CTRL_1_LJ | + SSM4567_SAI_CTRL_1_TDM | + SSM4567_SAI_CTRL_1_PDM, + ctrl1); } static int ssm4567_set_power(struct ssm4567 *ssm4567, bool enable) -- cgit v1.2.3 From fa8173a3ef0570affde7da352de202190b3786c2 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Thu, 23 Jul 2015 23:22:26 +0800 Subject: ASoC: pcm1681: Fix setting de-emphasis sampling rate selection The de-emphasis sampling rate selection is controlled by BIT[3:4] of PCM1681_DEEMPH_CONTROL register. Do proper left shift to set it. Signed-off-by: Axel Lin Acked-by: Marek Belisko Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/codecs/pcm1681.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/pcm1681.c b/sound/soc/codecs/pcm1681.c index 477e13d30971..e7ba557979cb 100644 --- a/sound/soc/codecs/pcm1681.c +++ b/sound/soc/codecs/pcm1681.c @@ -102,7 +102,7 @@ static int pcm1681_set_deemph(struct snd_soc_codec *codec) if (val != -1) { regmap_update_bits(priv->regmap, PCM1681_DEEMPH_CONTROL, - PCM1681_DEEMPH_RATE_MASK, val); + PCM1681_DEEMPH_RATE_MASK, val << 3); enable = 1; } else enable = 0; -- cgit v1.2.3 From bffc4496886683fac86d31e5d0cf9a22f8044e3d Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Sun, 19 Jul 2015 12:09:16 +0800 Subject: ASoC: cs4265: Fix setting dai format for Left/Right Justified MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit The settings in current code does not match the datasheet, fix it. DAC Control - Address 03h DAC Digital Interface Format (Bits 5:4) DAC_DIF1 DAC_DIF0 Description 0 0 Left Justified, up to 24-bit data (default) 0 1 I²S, up to 24-bit data 1 0 Right-Justified, 16-bit Data 1 1 Right-Justified, 24-bit Data Transmitter Control 2 - Address 12h Transmitter Digital Interface Format (Bits 7:6) Tx_DIF1 Tx_DIF0 Description Format Figure 0 0 Left Justified, up to 24-bit data (default) 0 1 I²S, up to 24-bit data 1 0 Right-Justified, 16-bit Data 1 1 Right-Justified, 24-bit Data Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/codecs/cs4265.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/cs4265.c b/sound/soc/codecs/cs4265.c index d7ec4756e45b..8e36198474d9 100644 --- a/sound/soc/codecs/cs4265.c +++ b/sound/soc/codecs/cs4265.c @@ -457,14 +457,14 @@ static int cs4265_pcm_hw_params(struct snd_pcm_substream *substream, case SND_SOC_DAIFMT_RIGHT_J: if (params_width(params) == 16) { snd_soc_update_bits(codec, CS4265_DAC_CTL, - CS4265_DAC_CTL_DIF, (1 << 5)); + CS4265_DAC_CTL_DIF, (2 << 4)); snd_soc_update_bits(codec, CS4265_SPDIF_CTL2, - CS4265_SPDIF_CTL2_DIF, (1 << 7)); + CS4265_SPDIF_CTL2_DIF, (2 << 6)); } else { snd_soc_update_bits(codec, CS4265_DAC_CTL, - CS4265_DAC_CTL_DIF, (3 << 5)); + CS4265_DAC_CTL_DIF, (3 << 4)); snd_soc_update_bits(codec, CS4265_SPDIF_CTL2, - CS4265_SPDIF_CTL2_DIF, (1 << 7)); + CS4265_SPDIF_CTL2_DIF, (3 << 6)); } break; case SND_SOC_DAIFMT_LEFT_J: @@ -473,7 +473,7 @@ static int cs4265_pcm_hw_params(struct snd_pcm_substream *substream, snd_soc_update_bits(codec, CS4265_ADC_CTL, CS4265_ADC_DIF, 0); snd_soc_update_bits(codec, CS4265_SPDIF_CTL2, - CS4265_SPDIF_CTL2_DIF, (1 << 6)); + CS4265_SPDIF_CTL2_DIF, 0); break; default: -- cgit v1.2.3 From 44008f0896ae205b02b0882dbf807f0de149efc4 Mon Sep 17 00:00:00 2001 From: Dan Carpenter Date: Sat, 25 Jul 2015 03:03:38 +0300 Subject: ALSA: hda - fix cs4210_spdif_automute() Smatch complains that we have nested checks for "spdif_present". It turns out the current behavior isn't correct, we should remove the first check and keep the second. Fixes: 1077a024812d ('ALSA: hda - Use generic parser for Cirrus codec driver') Signed-off-by: Dan Carpenter Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_cirrus.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c index 25ccf781fbe7..584a0343ab0c 100644 --- a/sound/pci/hda/patch_cirrus.c +++ b/sound/pci/hda/patch_cirrus.c @@ -999,9 +999,7 @@ static void cs4210_spdif_automute(struct hda_codec *codec, spec->spdif_present = spdif_present; /* SPDIF TX on/off */ - if (spdif_present) - snd_hda_set_pin_ctl(codec, spdif_pin, - spdif_present ? PIN_OUT : 0); + snd_hda_set_pin_ctl(codec, spdif_pin, spdif_present ? PIN_OUT : 0); cs_automute(codec); } -- cgit v1.2.3 From e9c28e16a0b7071c88a206ad8ce0c73f6605bba7 Mon Sep 17 00:00:00 2001 From: Woodrow Shen Date: Sat, 25 Jul 2015 10:36:16 +0800 Subject: ALSA: hda - Fix the headset mic that will not work on Dell desktop machine When the headset was plugged in the Dell desktop, the mic of headset can't be detected and workable. According to the alsa-info, we found the differece between alsa and init_pin_configs on the machine, so we need to add the pin configs to make headset work. Codec: Realtek ALC3234 Vendor Id: 0x10ec0255 Subsystem Id: 0x102806bb BugLink: https://bugs.launchpad.net/bugs/1477900 Signed-off-by: Woodrow Shen Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 22 ++++++++++++++++++++++ 1 file changed, 22 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 742fc626f9e1..d94c0e33f58d 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5472,6 +5472,28 @@ static const struct snd_hda_pin_quirk alc269_pin_fixup_tbl[] = { {0x1d, 0x4054c029}, {0x1e, 0x411111f0}, {0x21, 0x0221103f}), + SND_HDA_PIN_QUIRK(0x10ec0255, 0x1028, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE, + {0x12, 0x40000000}, + {0x14, 0x90170150}, + {0x17, 0x411111f0}, + {0x18, 0x411111f0}, + {0x19, 0x411111f0}, + {0x1a, 0x411111f0}, + {0x1b, 0x02011020}, + {0x1d, 0x4054c029}, + {0x1e, 0x411111f0}, + {0x21, 0x0221105f}), + SND_HDA_PIN_QUIRK(0x10ec0255, 0x1028, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE, + {0x12, 0x40000000}, + {0x14, 0x90170110}, + {0x17, 0x411111f0}, + {0x18, 0x411111f0}, + {0x19, 0x411111f0}, + {0x1a, 0x411111f0}, + {0x1b, 0x01014020}, + {0x1d, 0x4054c029}, + {0x1e, 0x411111f0}, + {0x21, 0x0221101f}), SND_HDA_PIN_QUIRK(0x10ec0255, 0x1028, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE, {0x12, 0x90a60160}, {0x14, 0x90170120}, -- cgit v1.2.3 From 9c6893e0be38b6ca9a56a854226e51dee0a16a5a Mon Sep 17 00:00:00 2001 From: Takashi Sakamoto Date: Sun, 26 Jul 2015 16:10:09 +0900 Subject: ALSA: fireworks: add support for AudioFire2 quirk Fireworks uses TSB43CB43(IceLynx-Micro) as its IEC 61883-1/6 interface. This chip includes ARM7 core, and loads and runs program. The firmware is stored in on-board memory and loaded every powering-on. Echo Audio ships several versions of firmwares for each model. These firmwares have each quirk and the quirk changes a sequence of packets. AudioFire2 has a quirk to transfer a first packet with non-zero in its dbc field. This causes ALSA Fireworks driver to detect discontinuity. As long as I investigated, firmware 5.7, 5.7.6 and 5.8 have this quirk. This commit adds a support for the quirk to handle AudioFire2 packets. For safe, CIP_SKIP_INIT_DBC_CHECK is applied to all versions of AudioFire2's firmwares. 02 00050002 90ffffff <- 42 0005000a 90013000 42 00050012 90014400 42 0005001a 90015800 02 0005001a 90ffffff 42 00050022 90019000 42 0005002a 9001a400 42 00050032 9001b800 02 00050032 90ffffff 42 0005003a 9001d000 42 00050042 9001e400 42 0005004a 9001f800 02 0005004a 90ffffff Signed-off-by: Takashi Sakamoto Signed-off-by: Takashi Iwai --- sound/firewire/fireworks/fireworks.c | 2 ++ sound/firewire/fireworks/fireworks.h | 1 + sound/firewire/fireworks/fireworks_stream.c | 3 +++ 3 files changed, 6 insertions(+) (limited to 'sound') diff --git a/sound/firewire/fireworks/fireworks.c b/sound/firewire/fireworks/fireworks.c index 2682e7e3e5c9..c670db4eee70 100644 --- a/sound/firewire/fireworks/fireworks.c +++ b/sound/firewire/fireworks/fireworks.c @@ -248,6 +248,8 @@ efw_probe(struct fw_unit *unit, err = get_hardware_info(efw); if (err < 0) goto error; + if (entry->model_id == MODEL_ECHO_AUDIOFIRE_2) + efw->is_af2 = true; if (entry->model_id == MODEL_ECHO_AUDIOFIRE_9) efw->is_af9 = true; diff --git a/sound/firewire/fireworks/fireworks.h b/sound/firewire/fireworks/fireworks.h index 4f0201a95222..c33252b7bc84 100644 --- a/sound/firewire/fireworks/fireworks.h +++ b/sound/firewire/fireworks/fireworks.h @@ -70,6 +70,7 @@ struct snd_efw { bool resp_addr_changable; /* for quirks */ + bool is_af2; bool is_af9; u32 firmware_version; diff --git a/sound/firewire/fireworks/fireworks_stream.c b/sound/firewire/fireworks/fireworks_stream.c index c55db1bddc80..a0762dd6231e 100644 --- a/sound/firewire/fireworks/fireworks_stream.c +++ b/sound/firewire/fireworks/fireworks_stream.c @@ -172,6 +172,9 @@ int snd_efw_stream_init_duplex(struct snd_efw *efw) efw->tx_stream.flags |= CIP_DBC_IS_END_EVENT; /* Fireworks reset dbc at bus reset. */ efw->tx_stream.flags |= CIP_SKIP_DBC_ZERO_CHECK; + /* AudioFire2 starts packets with non-zero dbc. */ + if (efw->is_af2) + efw->tx_stream.flags |= CIP_SKIP_INIT_DBC_CHECK; /* AudioFire9 always reports wrong dbs. */ if (efw->is_af9) efw->tx_stream.flags |= CIP_WRONG_DBS; -- cgit v1.2.3 From b9d9c9efc292dd0ffe172780f915ed74eba3556c Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 27 Jul 2015 10:13:37 +0200 Subject: ALSA: hda - Apply fixup for another Toshiba Satellite S50D Toshiba Satellite S50D has another model with a different PCI SSID (1179:fa93) while the previous fixup was for 1179:fa91. Adjust the fixup entry with SND_PCI_QUIRK_MASK() to match with both devices. Reported-by: Tim Sample Cc: Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_sigmatel.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index dcc7fe91244c..9d947aef2c8b 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -2920,7 +2920,8 @@ static const struct snd_pci_quirk stac92hd83xxx_fixup_tbl[] = { SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x148a, "HP Mini", STAC_92HD83XXX_HP_LED), SND_PCI_QUIRK_VENDOR(PCI_VENDOR_ID_HP, "HP", STAC_92HD83XXX_HP), - SND_PCI_QUIRK(PCI_VENDOR_ID_TOSHIBA, 0xfa91, + /* match both for 0xfa91 and 0xfa93 */ + SND_PCI_QUIRK_MASK(PCI_VENDOR_ID_TOSHIBA, 0xfffd, 0xfa91, "Toshiba Satellite S50D", STAC_92HD83XXX_GPIO10_EAPD), {} /* terminator */ }; -- cgit v1.2.3 From 5ce000b297a1c1bb126a14b02acb40318b88a903 Mon Sep 17 00:00:00 2001 From: Woodrow Shen Date: Mon, 27 Jul 2015 18:34:31 +0800 Subject: ALSA: hda - Add pin quirk for the headset mic jack detection on Dell laptop The new Dell laptop with codec 256 can't detect headset mic when headset was inserted on the machine. From alsa-info, we check init_pin_configs and need to define the new register value for pin 0x1d & 0x1e because the original macro ALC256_STANDARD_PINS can't match pin definition. Also, the macro ALC256_STANDARD_PINS is simplified by removing them. This makes headset mic works on laptop. Codec: Realtek ALC256 Vendor Id: 0x10ec0256 Subsystem Id: 0x102806f2 BugLink: https://bugs.launchpad.net/bugs/1478497 Signed-off-by: Woodrow Shen Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 15 +++++++++++---- 1 file changed, 11 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index d94c0e33f58d..4ae877c3b6a1 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5398,8 +5398,6 @@ static const struct hda_model_fixup alc269_fixup_models[] = { {0x19, 0x411111f0}, \ {0x1a, 0x411111f0}, \ {0x1b, 0x411111f0}, \ - {0x1d, 0x40700001}, \ - {0x1e, 0x411111f0}, \ {0x21, 0x02211020} #define ALC282_STANDARD_PINS \ @@ -5556,10 +5554,19 @@ static const struct snd_hda_pin_quirk alc269_pin_fixup_tbl[] = { {0x21, 0x02211030}), SND_HDA_PIN_QUIRK(0x10ec0256, 0x1028, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE, ALC256_STANDARD_PINS, - {0x13, 0x40000000}), + {0x13, 0x40000000}, + {0x1d, 0x40700001}, + {0x1e, 0x411111f0}), SND_HDA_PIN_QUIRK(0x10ec0256, 0x1028, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE, ALC256_STANDARD_PINS, - {0x13, 0x411111f0}), + {0x13, 0x411111f0}, + {0x1d, 0x40700001}, + {0x1e, 0x411111f0}), + SND_HDA_PIN_QUIRK(0x10ec0256, 0x1028, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE, + ALC256_STANDARD_PINS, + {0x13, 0x411111f0}, + {0x1d, 0x4077992d}, + {0x1e, 0x411111ff}), SND_HDA_PIN_QUIRK(0x10ec0280, 0x103c, "HP", ALC280_FIXUP_HP_GPIO4, {0x12, 0x90a60130}, {0x13, 0x40000000}, -- cgit v1.2.3 From 3a05d12f46cb95a6a685114819363a56e6170996 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 29 Jul 2015 09:04:52 +0200 Subject: ALSA: hda - Apply a fixup to Dell Vostro 5480 Dell Vostro 5480 (1028:069a) needs the very same quirk used for Vostro 5470 model to make bass speakers properly working. Reported-and-tested-by: Paulo Roberto de Oliveira Castro Cc: Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 4ae877c3b6a1..18ae17ebb356 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5185,6 +5185,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x064a, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x064b, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0665, "Dell XPS 13", ALC288_FIXUP_DELL_XPS_13), + SND_PCI_QUIRK(0x1028, 0x069a, "Dell Vostro 5480", ALC290_FIXUP_SUBWOOFER_HSJACK), SND_PCI_QUIRK(0x1028, 0x06c7, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x06d9, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x06da, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE), -- cgit v1.2.3 From 2d1cb7f658fb9c3ba8f9dab8aca297d4dfdec835 Mon Sep 17 00:00:00 2001 From: Yao-Wen Mao Date: Wed, 29 Jul 2015 15:13:54 +0800 Subject: ALSA: usb-audio: add dB range mapping for some devices Add the correct dB ranges of Bose Companion 5 and Drangonfly DAC 1.2. Signed-off-by: Yao-Wen Mao Cc: Signed-off-by: Takashi Iwai --- sound/usb/mixer_maps.c | 24 ++++++++++++++++++++++++ 1 file changed, 24 insertions(+) (limited to 'sound') diff --git a/sound/usb/mixer_maps.c b/sound/usb/mixer_maps.c index e5000da9e9d7..6a803eff87f7 100644 --- a/sound/usb/mixer_maps.c +++ b/sound/usb/mixer_maps.c @@ -341,6 +341,20 @@ static const struct usbmix_name_map scms_usb3318_map[] = { { 0 } }; +/* Bose companion 5, the dB conversion factor is 16 instead of 256 */ +static struct usbmix_dB_map bose_companion5_dB = {-5006, -6}; +static struct usbmix_name_map bose_companion5_map[] = { + { 3, NULL, .dB = &bose_companion5_dB }, + { 0 } /* terminator */ +}; + +/* Dragonfly DAC 1.2, the dB conversion factor is 1 instead of 256 */ +static struct usbmix_dB_map dragonfly_1_2_dB = {0, 5000}; +static struct usbmix_name_map dragonfly_1_2_map[] = { + { 7, NULL, .dB = &dragonfly_1_2_dB }, + { 0 } /* terminator */ +}; + /* * Control map entries */ @@ -451,6 +465,16 @@ static struct usbmix_ctl_map usbmix_ctl_maps[] = { .id = USB_ID(0x25c4, 0x0003), .map = scms_usb3318_map, }, + { + /* Bose Companion 5 */ + .id = USB_ID(0x05a7, 0x1020), + .map = bose_companion5_map, + }, + { + /* Dragonfly DAC 1.2 */ + .id = USB_ID(0x21b4, 0x0081), + .map = dragonfly_1_2_map, + }, { 0 } /* terminator */ }; -- cgit v1.2.3 From 4b563ea317c2262987f0675abf15066614f536a1 Mon Sep 17 00:00:00 2001 From: Jie Yang Date: Tue, 28 Jul 2015 16:01:05 +0800 Subject: ASoC: Intel: haswell: fix initialize 'NULL device *' issue Add initialization for sst_hsw.dev at init stage, which fix the 'NULL device *' warning issues. Signed-off-by: Jie Yang Signed-off-by: Mark Brown --- sound/soc/intel/haswell/sst-haswell-ipc.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound') diff --git a/sound/soc/intel/haswell/sst-haswell-ipc.c b/sound/soc/intel/haswell/sst-haswell-ipc.c index f95f271aab0c..f6efa9d4acad 100644 --- a/sound/soc/intel/haswell/sst-haswell-ipc.c +++ b/sound/soc/intel/haswell/sst-haswell-ipc.c @@ -2119,6 +2119,8 @@ int sst_hsw_dsp_init(struct device *dev, struct sst_pdata *pdata) if (hsw == NULL) return -ENOMEM; + hsw->dev = dev; + ipc = &hsw->ipc; ipc->dev = dev; ipc->ops.tx_msg = hsw_tx_msg; -- cgit v1.2.3 From 45f503df1ba445359b94e1758c5e4f2c3460c8e4 Mon Sep 17 00:00:00 2001 From: Jie Yang Date: Tue, 28 Jul 2015 16:01:06 +0800 Subject: ASoC: Intel: sst_byt: fix initialize 'NULL device *' issue Add initialization for sst_byt.dev at init stage, which fix the 'NULL device *' warning issues. Signed-off-by: Jie Yang Signed-off-by: Mark Brown --- sound/soc/intel/baytrail/sst-baytrail-ipc.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound') diff --git a/sound/soc/intel/baytrail/sst-baytrail-ipc.c b/sound/soc/intel/baytrail/sst-baytrail-ipc.c index 4c01bb43928d..5bbaa667bec1 100644 --- a/sound/soc/intel/baytrail/sst-baytrail-ipc.c +++ b/sound/soc/intel/baytrail/sst-baytrail-ipc.c @@ -701,6 +701,8 @@ int sst_byt_dsp_init(struct device *dev, struct sst_pdata *pdata) if (byt == NULL) return -ENOMEM; + byt->dev = dev; + ipc = &byt->ipc; ipc->dev = dev; ipc->ops.tx_msg = byt_tx_msg; -- cgit v1.2.3 From 342e84490574cbb2a9c5b1d0886a112ad2bcf4d7 Mon Sep 17 00:00:00 2001 From: "U. Artie Eoff" Date: Tue, 28 Jul 2015 13:29:56 -0700 Subject: ALSA: hda - Fix race between PM ops and HDA init/probe PM ops could be triggered before HDA is done initializing and cause PM to set HDA controller to D3Hot. This can result in "CORB reset timeout#2, CORBRP = 65535" and "no codecs initialized". Additionally, PM ops can be triggered before azx_probe_continue finishes (async probe). This can result in a NULL deref kernel crash. To fix this, avoid PM ops if !chip->running. Signed-off-by: U. Artie Eoff Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 735bdcb04ce8..c38c68f57938 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -867,7 +867,7 @@ static int azx_suspend(struct device *dev) chip = card->private_data; hda = container_of(chip, struct hda_intel, chip); - if (chip->disabled || hda->init_failed) + if (chip->disabled || hda->init_failed || !chip->running) return 0; bus = azx_bus(chip); @@ -902,7 +902,7 @@ static int azx_resume(struct device *dev) chip = card->private_data; hda = container_of(chip, struct hda_intel, chip); - if (chip->disabled || hda->init_failed) + if (chip->disabled || hda->init_failed || !chip->running) return 0; if (chip->driver_caps & AZX_DCAPS_I915_POWERWELL @@ -1027,7 +1027,7 @@ static int azx_runtime_idle(struct device *dev) return 0; if (!power_save_controller || !azx_has_pm_runtime(chip) || - azx_bus(chip)->codec_powered) + azx_bus(chip)->codec_powered || !chip->running) return -EBUSY; return 0; -- cgit v1.2.3 From 649ccd08534ee26deb2e5b08509800d0e95167f5 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 30 Jul 2015 22:30:29 +0200 Subject: ALSA: hda - Fix MacBook Pro 5,2 quirk MacBook Pro 5,2 with ALC889 codec had already a fixup entry, but this seems not working correctly, a fix for pin NID 0x15 is needed in addition. It's equivalent with the fixup for MacBook Air 1,1, so use this instead. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=102131 Reported-and-tested-by: Jeffery Miller Cc: Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 18ae17ebb356..c456c04e0928 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -2222,7 +2222,7 @@ static const struct snd_pci_quirk alc882_fixup_tbl[] = { SND_PCI_QUIRK(0x106b, 0x4300, "iMac 9,1", ALC889_FIXUP_IMAC91_VREF), SND_PCI_QUIRK(0x106b, 0x4600, "MacbookPro 5,2", ALC889_FIXUP_IMAC91_VREF), SND_PCI_QUIRK(0x106b, 0x4900, "iMac 9,1 Aluminum", ALC889_FIXUP_IMAC91_VREF), - SND_PCI_QUIRK(0x106b, 0x4a00, "Macbook 5,2", ALC889_FIXUP_IMAC91_VREF), + SND_PCI_QUIRK(0x106b, 0x4a00, "Macbook 5,2", ALC889_FIXUP_MBA11_VREF), SND_PCI_QUIRK(0x1071, 0x8258, "Evesham Voyaeger", ALC882_FIXUP_EAPD), SND_PCI_QUIRK(0x1462, 0x7350, "MSI-7350", ALC889_FIXUP_CD), -- cgit v1.2.3 From 8ec7cfce3762299ae289c384e281b2f4010ae231 Mon Sep 17 00:00:00 2001 From: Tomer Barletz Date: Sun, 2 Aug 2015 02:08:57 -0700 Subject: ALSA: oxygen: Fix logical-not-parentheses warning This fixes the following warning, that is seen with gcc 5.1: warning: logical not is only applied to the left hand side of comparison [-Wlogical-not-parentheses]. Signed-off-by: Tomer Barletz Signed-off-by: Takashi Iwai --- sound/pci/oxygen/oxygen_mixer.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/oxygen/oxygen_mixer.c b/sound/pci/oxygen/oxygen_mixer.c index 6492bca8c70f..4ca12665ff73 100644 --- a/sound/pci/oxygen/oxygen_mixer.c +++ b/sound/pci/oxygen/oxygen_mixer.c @@ -88,7 +88,7 @@ static int dac_mute_put(struct snd_kcontrol *ctl, int changed; mutex_lock(&chip->mutex); - changed = !value->value.integer.value[0] != chip->dac_mute; + changed = (!value->value.integer.value[0]) != chip->dac_mute; if (changed) { chip->dac_mute = !value->value.integer.value[0]; chip->model.update_dac_mute(chip); -- cgit v1.2.3 From a094935e4ebdf5c22c45b8aeeb2d88e9e8c53dbf Mon Sep 17 00:00:00 2001 From: Bard Liao Date: Mon, 3 Aug 2015 12:17:39 +0800 Subject: ASoC: rt5645: Fix lost pin setting for DMIC1 I2S2_DAC pin can be used for I2S or GPIO. We should set it as GPIO if we use GPIO5 as DMIC1 data pin. Signed-off-by: Bard Liao Signed-off-by: Mark Brown --- sound/soc/codecs/rt5645.c | 2 ++ sound/soc/codecs/rt5645.h | 4 ++++ 2 files changed, 6 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index e9cc3aae5366..961bd7e5877e 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -3341,6 +3341,8 @@ static int rt5645_i2c_probe(struct i2c_client *i2c, break; case RT5645_DMIC_DATA_GPIO5: + regmap_update_bits(rt5645->regmap, RT5645_GPIO_CTRL1, + RT5645_I2S2_DAC_PIN_MASK, RT5645_I2S2_DAC_PIN_GPIO); regmap_update_bits(rt5645->regmap, RT5645_DMIC_CTRL1, RT5645_DMIC_1_DP_MASK, RT5645_DMIC_1_DP_GPIO5); regmap_update_bits(rt5645->regmap, RT5645_GPIO_CTRL1, diff --git a/sound/soc/codecs/rt5645.h b/sound/soc/codecs/rt5645.h index 0353a6a273ab..278bb9f464c4 100644 --- a/sound/soc/codecs/rt5645.h +++ b/sound/soc/codecs/rt5645.h @@ -1693,6 +1693,10 @@ #define RT5645_GP6_PIN_SFT 6 #define RT5645_GP6_PIN_GPIO6 (0x0 << 6) #define RT5645_GP6_PIN_DMIC2_SDA (0x1 << 6) +#define RT5645_I2S2_DAC_PIN_MASK (0x1 << 4) +#define RT5645_I2S2_DAC_PIN_SFT 4 +#define RT5645_I2S2_DAC_PIN_I2S (0x0 << 4) +#define RT5645_I2S2_DAC_PIN_GPIO (0x1 << 4) #define RT5645_GP8_PIN_MASK (0x1 << 3) #define RT5645_GP8_PIN_SFT 3 #define RT5645_GP8_PIN_GPIO8 (0x0 << 3) -- cgit v1.2.3 From 9b06dc939489152b583131f49929ed1c6ae83740 Mon Sep 17 00:00:00 2001 From: Jeeja KP Date: Tue, 4 Aug 2015 09:28:38 +0530 Subject: ALSA: HDA: Fix stream assignment for host in decoupled mode This fixes issue in assigning host stream in case of decoupled mode. The check to verify if the stream is already in use was wrong so fix that Signed-off-by: Jeeja KP Signed-off-by: Vinod Koul Signed-off-by: Takashi Iwai --- sound/hda/ext/hdac_ext_stream.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/hda/ext/hdac_ext_stream.c b/sound/hda/ext/hdac_ext_stream.c index f8ffbdbb450d..3de47dd1a76d 100644 --- a/sound/hda/ext/hdac_ext_stream.c +++ b/sound/hda/ext/hdac_ext_stream.c @@ -299,7 +299,7 @@ hdac_ext_host_stream_assign(struct hdac_ext_bus *ebus, if (stream->direction != substream->stream) continue; - if (stream->opened) { + if (!stream->opened) { if (!hstream->decoupled) snd_hdac_ext_stream_decouple(ebus, hstream, true); res = hstream; -- cgit v1.2.3 From 5d942ce63c8fd98794a8ba9af559925c8432a052 Mon Sep 17 00:00:00 2001 From: Jeeja KP Date: Tue, 4 Aug 2015 09:28:39 +0530 Subject: ALSA: HDA: Dont check return for snd_hdac_chip_readl The snd_hdac_chip_readl return can never be less than zeros, so no point in checking for the return value This fixes following static checker warnings in snd_hdac_ext_bus_parse_capabilities sound/hda/ext/hdac_ext_controller.c:47 snd_hdac_ext_bus_parse_capabilities() warn: unsigned 'offset' is never less than zero. sound/hda/ext/hdac_ext_controller.c:54 snd_hdac_ext_bus_parse_capabilities() warn: unsigned 'cur_cap' is never less than zero. Signed-off-by: Jeeja KP Reported-by: Dan Carpenter Signed-off-by: Vinod Koul Signed-off-by: Takashi Iwai --- sound/hda/ext/hdac_ext_controller.c | 6 ------ 1 file changed, 6 deletions(-) (limited to 'sound') diff --git a/sound/hda/ext/hdac_ext_controller.c b/sound/hda/ext/hdac_ext_controller.c index b2da19b60f4e..358f16195483 100644 --- a/sound/hda/ext/hdac_ext_controller.c +++ b/sound/hda/ext/hdac_ext_controller.c @@ -44,16 +44,10 @@ int snd_hdac_ext_bus_parse_capabilities(struct hdac_ext_bus *ebus) offset = snd_hdac_chip_readl(bus, LLCH); - if (offset < 0) - return -EIO; - /* Lets walk the linked capabilities list */ do { cur_cap = _snd_hdac_chip_read(l, bus, offset); - if (cur_cap < 0) - return -EIO; - dev_dbg(bus->dev, "Capability version: 0x%x\n", ((cur_cap & AZX_CAP_HDR_VER_MASK) >> AZX_CAP_HDR_VER_OFF)); -- cgit v1.2.3 From 5406898354ebfb11f49b955fb5e49a62786a542f Mon Sep 17 00:00:00 2001 From: Mengdong Lin Date: Tue, 4 Aug 2015 15:47:35 +0100 Subject: ASoC: topology: fix typo in soc_tplg_kcontrol_bind_io() Signed-off-by: Mengdong Lin Signed-off-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/soc-topology.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index d0960683c409..6a547c6dd3a1 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -534,7 +534,7 @@ static int soc_tplg_kcontrol_bind_io(struct snd_soc_tplg_ctl_hdr *hdr, k->put = bops[i].put; if (k->get == NULL && bops[i].id == hdr->ops.get) k->get = bops[i].get; - if (k->info == NULL && ops[i].id == hdr->ops.info) + if (k->info == NULL && bops[i].id == hdr->ops.info) k->info = bops[i].info; } -- cgit v1.2.3 From c85523d1d97cc86aadc388221aa83ae9bc1e7cca Mon Sep 17 00:00:00 2001 From: Takashi Sakamoto Date: Wed, 5 Aug 2015 09:21:04 +0900 Subject: Revert "ALSA: fireworks: add support for AudioFire2 quirk" This reverts commit 9c6893e0be38b6ca9a56a854226e51dee0a16a5a. The fix is superseded by the next commit as a better implementation for supporting AudioFire2/AudioFire4/AudioFirePre8 quirks. Signed-off-by: Takashi Sakamoto Signed-off-by: Takashi Iwai --- sound/firewire/fireworks/fireworks.c | 2 -- sound/firewire/fireworks/fireworks.h | 1 - sound/firewire/fireworks/fireworks_stream.c | 3 --- 3 files changed, 6 deletions(-) (limited to 'sound') diff --git a/sound/firewire/fireworks/fireworks.c b/sound/firewire/fireworks/fireworks.c index c670db4eee70..2682e7e3e5c9 100644 --- a/sound/firewire/fireworks/fireworks.c +++ b/sound/firewire/fireworks/fireworks.c @@ -248,8 +248,6 @@ efw_probe(struct fw_unit *unit, err = get_hardware_info(efw); if (err < 0) goto error; - if (entry->model_id == MODEL_ECHO_AUDIOFIRE_2) - efw->is_af2 = true; if (entry->model_id == MODEL_ECHO_AUDIOFIRE_9) efw->is_af9 = true; diff --git a/sound/firewire/fireworks/fireworks.h b/sound/firewire/fireworks/fireworks.h index c33252b7bc84..4f0201a95222 100644 --- a/sound/firewire/fireworks/fireworks.h +++ b/sound/firewire/fireworks/fireworks.h @@ -70,7 +70,6 @@ struct snd_efw { bool resp_addr_changable; /* for quirks */ - bool is_af2; bool is_af9; u32 firmware_version; diff --git a/sound/firewire/fireworks/fireworks_stream.c b/sound/firewire/fireworks/fireworks_stream.c index a0762dd6231e..c55db1bddc80 100644 --- a/sound/firewire/fireworks/fireworks_stream.c +++ b/sound/firewire/fireworks/fireworks_stream.c @@ -172,9 +172,6 @@ int snd_efw_stream_init_duplex(struct snd_efw *efw) efw->tx_stream.flags |= CIP_DBC_IS_END_EVENT; /* Fireworks reset dbc at bus reset. */ efw->tx_stream.flags |= CIP_SKIP_DBC_ZERO_CHECK; - /* AudioFire2 starts packets with non-zero dbc. */ - if (efw->is_af2) - efw->tx_stream.flags |= CIP_SKIP_INIT_DBC_CHECK; /* AudioFire9 always reports wrong dbs. */ if (efw->is_af9) efw->tx_stream.flags |= CIP_WRONG_DBS; -- cgit v1.2.3 From 18f5ed365d3f188a91149d528c853000330a4a58 Mon Sep 17 00:00:00 2001 From: Takashi Sakamoto Date: Wed, 5 Aug 2015 09:21:05 +0900 Subject: ALSA: fireworks/firewire-lib: add support for recent firmware quirk Fireworks uses TSB43CB43(IceLynx-Micro) as its IEC 61883-1/6 interface. This chip includes ARM7 core, and loads and runs program. The firmware is stored in on-board memory and loaded every powering-on from it. Echo Audio ships several versions of firmwares for each model. These firmwares have each quirk and the quirk changes a sequence of packets. As long as I investigated, AudioFire2/AudioFire4/AudioFirePre8 have a quirk to transfer a first packet with 0x02 in its dbc field. This causes ALSA Fireworks driver to detect discontinuity. In this case, firmware version 5.7.0, 5.7.3 and 5.8.0 are used. Payload CIP CIP quadlets header1 header2 02 00050002 90ffffff <- 42 0005000a 90013000 42 00050012 90014400 42 0005001a 90015800 02 0005001a 90ffffff 42 00050022 90019000 42 0005002a 9001a400 42 00050032 9001b800 02 00050032 90ffffff 42 0005003a 9001d000 42 00050042 9001e400 42 0005004a 9001f800 02 0005004a 90ffffff (AudioFire2 with firmware version 5.7.) $ dmesg snd-fireworks fw1.0: Detect discontinuity of CIP: 00 02 These models, AudioFire8 (since Jul 2009 ) and Gibson Robot Interface Pack series uses the same ARM binary as their firmware. Thus, this quirk may be observed among them. This commit adds a new member for AMDTP structure. This member represents the value of dbc field in a first AMDTP packet. Drivers can set it with a preferred value according to model's quirk. Tested-by: Johannes Oertei Signed-off-by: Takashi Sakamoto Cc: Signed-off-by: Takashi Iwai --- sound/firewire/amdtp.c | 5 +++-- sound/firewire/amdtp.h | 2 ++ sound/firewire/fireworks/fireworks.c | 8 ++++++++ sound/firewire/fireworks/fireworks.h | 1 + sound/firewire/fireworks/fireworks_stream.c | 9 +++++++++ 5 files changed, 23 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/firewire/amdtp.c b/sound/firewire/amdtp.c index 7bb988fa6b6d..2a153d260836 100644 --- a/sound/firewire/amdtp.c +++ b/sound/firewire/amdtp.c @@ -740,8 +740,9 @@ static int handle_in_packet(struct amdtp_stream *s, s->data_block_counter != UINT_MAX) data_block_counter = s->data_block_counter; - if (((s->flags & CIP_SKIP_DBC_ZERO_CHECK) && data_block_counter == 0) || - (s->data_block_counter == UINT_MAX)) { + if (((s->flags & CIP_SKIP_DBC_ZERO_CHECK) && + data_block_counter == s->tx_first_dbc) || + s->data_block_counter == UINT_MAX) { lost = false; } else if (!(s->flags & CIP_DBC_IS_END_EVENT)) { lost = data_block_counter != s->data_block_counter; diff --git a/sound/firewire/amdtp.h b/sound/firewire/amdtp.h index 26b909329e54..b2cf9e75693b 100644 --- a/sound/firewire/amdtp.h +++ b/sound/firewire/amdtp.h @@ -157,6 +157,8 @@ struct amdtp_stream { /* quirk: fixed interval of dbc between previos/current packets. */ unsigned int tx_dbc_interval; + /* quirk: indicate the value of dbc field in a first packet. */ + unsigned int tx_first_dbc; bool callbacked; wait_queue_head_t callback_wait; diff --git a/sound/firewire/fireworks/fireworks.c b/sound/firewire/fireworks/fireworks.c index 2682e7e3e5c9..c94a432f7cc6 100644 --- a/sound/firewire/fireworks/fireworks.c +++ b/sound/firewire/fireworks/fireworks.c @@ -248,8 +248,16 @@ efw_probe(struct fw_unit *unit, err = get_hardware_info(efw); if (err < 0) goto error; + /* AudioFire8 (since 2009) and AudioFirePre8 */ if (entry->model_id == MODEL_ECHO_AUDIOFIRE_9) efw->is_af9 = true; + /* These models uses the same firmware. */ + if (entry->model_id == MODEL_ECHO_AUDIOFIRE_2 || + entry->model_id == MODEL_ECHO_AUDIOFIRE_4 || + entry->model_id == MODEL_ECHO_AUDIOFIRE_9 || + entry->model_id == MODEL_GIBSON_RIP || + entry->model_id == MODEL_GIBSON_GOLDTOP) + efw->is_fireworks3 = true; snd_efw_proc_init(efw); diff --git a/sound/firewire/fireworks/fireworks.h b/sound/firewire/fireworks/fireworks.h index 4f0201a95222..084d414b228c 100644 --- a/sound/firewire/fireworks/fireworks.h +++ b/sound/firewire/fireworks/fireworks.h @@ -71,6 +71,7 @@ struct snd_efw { /* for quirks */ bool is_af9; + bool is_fireworks3; u32 firmware_version; unsigned int midi_in_ports; diff --git a/sound/firewire/fireworks/fireworks_stream.c b/sound/firewire/fireworks/fireworks_stream.c index c55db1bddc80..7e353f1f7bff 100644 --- a/sound/firewire/fireworks/fireworks_stream.c +++ b/sound/firewire/fireworks/fireworks_stream.c @@ -172,6 +172,15 @@ int snd_efw_stream_init_duplex(struct snd_efw *efw) efw->tx_stream.flags |= CIP_DBC_IS_END_EVENT; /* Fireworks reset dbc at bus reset. */ efw->tx_stream.flags |= CIP_SKIP_DBC_ZERO_CHECK; + /* + * But Recent firmwares starts packets with non-zero dbc. + * Driver version 5.7.6 installs firmware version 5.7.3. + */ + if (efw->is_fireworks3 && + (efw->firmware_version == 0x5070000 || + efw->firmware_version == 0x5070300 || + efw->firmware_version == 0x5080000)) + efw->tx_stream.tx_first_dbc = 0x02; /* AudioFire9 always reports wrong dbs. */ if (efw->is_af9) efw->tx_stream.flags |= CIP_WRONG_DBS; -- cgit v1.2.3 From 73851b36fe73819f8c201971e913324d4846a7ea Mon Sep 17 00:00:00 2001 From: Hui Wang Date: Wed, 5 Aug 2015 18:03:34 +0800 Subject: ALSA: hda - one Dell machine needs the headphone white noise fixup The fixup ALC292_FIXUP_DISABLE_AAMIX can fix the white noise of the headphone on this Dell machine. Cc: Signed-off-by: Hui Wang Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index c456c04e0928..0b9847affbec 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5189,6 +5189,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x06c7, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x06d9, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x06da, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x06de, "Dell", ALC292_FIXUP_DISABLE_AAMIX), SND_PCI_QUIRK(0x1028, 0x164a, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x164b, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x103c, 0x1586, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC2), -- cgit v1.2.3 From 6dc6db790a67d28e46abefc44ca1a3bd438b2920 Mon Sep 17 00:00:00 2001 From: "Subhransu S. Prusty" Date: Mon, 29 Jun 2015 17:36:44 +0100 Subject: ASoC: topology: Add subsequence in topology Some widgets may need sorting within, So add this support in topology. Signed-off-by: Subhransu S. Prusty Signed-off-by: Mark Brown --- include/uapi/sound/asoc.h | 1 + sound/soc/soc-topology.c | 1 + 2 files changed, 2 insertions(+) (limited to 'sound') diff --git a/include/uapi/sound/asoc.h b/include/uapi/sound/asoc.h index 12215205ab8d..7ae13fbc0a3e 100644 --- a/include/uapi/sound/asoc.h +++ b/include/uapi/sound/asoc.h @@ -347,6 +347,7 @@ struct snd_soc_tplg_dapm_widget { __le32 reg; /* negative reg = no direct dapm */ __le32 shift; /* bits to shift */ __le32 mask; /* non-shifted mask */ + __le32 subseq; /* sort within widget type */ __u32 invert; /* invert the power bit */ __u32 ignore_suspend; /* kept enabled over suspend */ __u16 event_flags; diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index 6a547c6dd3a1..9f2b048f1071 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -1351,6 +1351,7 @@ static int soc_tplg_dapm_widget_create(struct soc_tplg *tplg, template.reg = w->reg; template.shift = w->shift; template.mask = w->mask; + template.subseq = w->subseq; template.on_val = w->invert ? 0 : 1; template.off_val = w->invert ? 1 : 0; template.ignore_suspend = w->ignore_suspend; -- cgit v1.2.3 From 28a87eebcad40101b1b273cbd4f2a02c104f9367 Mon Sep 17 00:00:00 2001 From: Mengdong Lin Date: Wed, 5 Aug 2015 14:41:13 +0100 Subject: ASoC: topology: Update TLV support so we can support more TLV types Currently the TLV topology structure is targeted at only supporting the DB scale data. This patch extends support for the other TLV types so they can be easily added at a later stage. TLV structure is moved to common topology control header since it's a common field for controls and can be processed in a general way. Users must set a proper access flag for a control since it's used to decide if the TLV field is valid and if a TLV callback is needed. Removed the following fields from topology TLV struct: - size/count: type can decide the size. - numid: not needed to initialize TLV for kcontrol. - data: replaced by the type specific struct. Added TLV structure to generic control header and removed TLV structure from mixer control. Signed-off-by: Mengdong Lin Signed-off-by: Liam Girdwood Signed-off-by: Mark Brown --- include/uapi/sound/asoc.h | 19 +++++++++++----- sound/soc/soc-topology.c | 58 +++++++++++++++++++++++++++++++++-------------- 2 files changed, 54 insertions(+), 23 deletions(-) (limited to 'sound') diff --git a/include/uapi/sound/asoc.h b/include/uapi/sound/asoc.h index 2819fc1f8458..aa3a79b42438 100644 --- a/include/uapi/sound/asoc.h +++ b/include/uapi/sound/asoc.h @@ -137,11 +137,19 @@ struct snd_soc_tplg_private { /* * Kcontrol TLV data. */ +struct snd_soc_tplg_tlv_dbscale { + __le32 min; + __le32 step; + __le32 mute; +} __attribute__((packed)); + struct snd_soc_tplg_ctl_tlv { - __le32 size; /* in bytes aligned to 4 */ - __le32 numid; /* control element numeric identification */ - __le32 count; /* number of elem in data array */ - __le32 data[SND_SOC_TPLG_TLV_SIZE]; + __le32 size; /* in bytes of this structure */ + __le32 type; /* SNDRV_CTL_TLVT_*, type of TLV */ + union { + __le32 data[SND_SOC_TPLG_TLV_SIZE]; + struct snd_soc_tplg_tlv_dbscale scale; + }; } __attribute__((packed)); /* @@ -172,7 +180,7 @@ struct snd_soc_tplg_ctl_hdr { char name[SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; __le32 access; struct snd_soc_tplg_kcontrol_ops_id ops; - __le32 tlv_size; /* non zero means control has TLV data */ + struct snd_soc_tplg_ctl_tlv tlv; } __attribute__((packed)); /* @@ -260,7 +268,6 @@ struct snd_soc_tplg_mixer_control { __le32 invert; __le32 num_channels; struct snd_soc_tplg_channel channel[SND_SOC_TPLG_MAX_CHAN]; - struct snd_soc_tplg_ctl_tlv tlv; struct snd_soc_tplg_private priv; } __attribute__((packed)); diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index 2c70f30d2d78..31068b8f3db0 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -33,6 +33,7 @@ #include #include #include +#include /* * We make several passes over the data (since it wont necessarily be ordered) @@ -579,28 +580,51 @@ static int soc_tplg_init_kcontrol(struct soc_tplg *tplg, return 0; } + +static int soc_tplg_create_tlv_db_scale(struct soc_tplg *tplg, + struct snd_kcontrol_new *kc, struct snd_soc_tplg_tlv_dbscale *scale) +{ + unsigned int item_len = 2 * sizeof(unsigned int); + unsigned int *p; + + p = kzalloc(item_len + 2 * sizeof(unsigned int), GFP_KERNEL); + if (!p) + return -ENOMEM; + + p[0] = SNDRV_CTL_TLVT_DB_SCALE; + p[1] = item_len; + p[2] = scale->min; + p[3] = (scale->step & TLV_DB_SCALE_MASK) + | (scale->mute ? TLV_DB_SCALE_MUTE : 0); + + kc->tlv.p = (void *)p; + return 0; +} + static int soc_tplg_create_tlv(struct soc_tplg *tplg, - struct snd_kcontrol_new *kc, struct snd_soc_tplg_ctl_tlv *tplg_tlv) + struct snd_kcontrol_new *kc, struct snd_soc_tplg_ctl_hdr *tc) { - struct snd_ctl_tlv *tlv; - int size; + struct snd_soc_tplg_ctl_tlv *tplg_tlv; - if (tplg_tlv->count == 0) + if (!(tc->access & SNDRV_CTL_ELEM_ACCESS_TLV_READWRITE)) return 0; - size = ((tplg_tlv->count + (sizeof(unsigned int) - 1)) & - ~(sizeof(unsigned int) - 1)); - tlv = kzalloc(sizeof(*tlv) + size, GFP_KERNEL); - if (tlv == NULL) - return -ENOMEM; - - dev_dbg(tplg->dev, " created TLV type %d size %d bytes\n", - tplg_tlv->numid, size); + if (tc->access & SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK) { + kc->tlv.c = snd_soc_bytes_tlv_callback; + } else { + tplg_tlv = &tc->tlv; + switch (tplg_tlv->type) { + case SNDRV_CTL_TLVT_DB_SCALE: + return soc_tplg_create_tlv_db_scale(tplg, kc, + &tplg_tlv->scale); - tlv->numid = tplg_tlv->numid; - tlv->length = size; - memcpy(&tlv->tlv[0], tplg_tlv->data, size); - kc->tlv.p = (void *)tlv; + /* TODO: add support for other TLV types */ + default: + dev_dbg(tplg->dev, "Unsupported TLV type %d\n", + tplg_tlv->type); + return -EINVAL; + } + } return 0; } @@ -772,7 +796,7 @@ static int soc_tplg_dmixer_create(struct soc_tplg *tplg, unsigned int count, } /* create any TLV data */ - soc_tplg_create_tlv(tplg, &kc, &mc->tlv); + soc_tplg_create_tlv(tplg, &kc, &mc->hdr); /* register control here */ err = soc_tplg_add_kcontrol(tplg, &kc, -- cgit v1.2.3