From 4623a614e87e4f2df08c83b5b9f68af394951dc9 Mon Sep 17 00:00:00 2001 From: Koro Chen Date: Tue, 23 Jun 2015 19:01:20 +0800 Subject: [PATCH 01/29] 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(+) 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; From ffacb48e5a4665d3d7286babb38a5af855a36bc0 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Fri, 26 Jun 2015 10:39:43 +0100 Subject: [PATCH 02/29] 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(-) 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; From 7084ffbff494669b06ceb457150c38887e26d2a3 Mon Sep 17 00:00:00 2001 From: Liam Girdwood Date: Mon, 29 Jun 2015 17:36:43 +0100 Subject: [PATCH 03/29] 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(-) 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, From 9c3a473220dda073603a24e93d3015322bda950a Mon Sep 17 00:00:00 2001 From: Vinod Koul Date: Mon, 29 Jun 2015 17:36:45 +0100 Subject: [PATCH 04/29] ASoC: topology: fix typos in topology header Signed-off-by: Vinod Koul Signed-off-by: Mark Brown --- include/uapi/sound/asoc.h | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/include/uapi/sound/asoc.h b/include/uapi/sound/asoc.h index 12215205ab8d..785c5ca0994b 100644 --- a/include/uapi/sound/asoc.h +++ b/include/uapi/sound/asoc.h @@ -110,7 +110,7 @@ /* * Block Header. - * This header preceeds all object and object arrays below. + * This header precedes all object and object arrays below. */ struct snd_soc_tplg_hdr { __le32 magic; /* magic number */ @@ -222,7 +222,7 @@ struct snd_soc_tplg_stream_config { /* * Manifest. List totals for each payload type. Not used in parsing, but will * be passed to the component driver before any other objects in order for any - * global componnent resource allocations. + * global component resource allocations. * * File block representation for manifest :- * +-----------------------------------+----+ From e50b1e06b79e9d51efbff9627b4dd407184ef43f Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Mon, 6 Jul 2015 17:01:24 +0200 Subject: [PATCH 05/29] 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(+) 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; } From 8bc76c8bf6e5d96985eb05afe1b94699d580eb68 Mon Sep 17 00:00:00 2001 From: "Fang, Yang A" Date: Mon, 6 Jul 2015 14:12:38 -0700 Subject: [PATCH 06/29] 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(-) 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[] = { From ebac95a9208e6b5f134df8518df1bfd1b3fee354 Mon Sep 17 00:00:00 2001 From: Juergen Borleis Date: Fri, 3 Jul 2015 12:39:36 +0200 Subject: [PATCH 07/29] 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(-) 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; From 56e7366e43ca676dd28f0e91240a579ad41e9b71 Mon Sep 17 00:00:00 2001 From: Vinod Koul Date: Fri, 19 Jun 2015 23:55:26 +0530 Subject: [PATCH 08/29] 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(-) 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/ From e18077b6e5dfe26e9fbbdc1fd1085a1701c24bea Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Wed, 8 Jul 2015 21:59:59 +0200 Subject: [PATCH 09/29] 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(-) 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); } From 2210438b6aae9668cf89c272fef935b83aedf81d Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Wed, 8 Jul 2015 22:14:48 +0200 Subject: [PATCH 10/29] 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(+) 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); From 94319ba10ecabc8f28129566d1f5793e3e7a0a79 Mon Sep 17 00:00:00 2001 From: Koro Chen Date: Thu, 9 Jul 2015 10:51:27 +0800 Subject: [PATCH 11/29] 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 --- .../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(-) 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); From a18da49ff3ec849c4584ae3abe2b83ff50705ace Mon Sep 17 00:00:00 2001 From: Krzysztof Kozlowski Date: Thu, 9 Jul 2015 22:19:28 +0900 Subject: [PATCH 12/29] 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(-) 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); From 2dfbe9afcda3243dd309826cbbb8ae9e9f602006 Mon Sep 17 00:00:00 2001 From: Krzysztof Kozlowski Date: Thu, 9 Jul 2015 22:19:29 +0900 Subject: [PATCH 13/29] 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(-) 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); From 2c57d478018551f2a0983293413f2f198e49ec23 Mon Sep 17 00:00:00 2001 From: Jeeja KP Date: Tue, 14 Jul 2015 13:10:47 +0530 Subject: [PATCH 14/29] 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(-) 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, From 412efa73dcd3bd03c1838c91e094533a95529039 Mon Sep 17 00:00:00 2001 From: Shilpa Sreeramalu Date: Wed, 15 Jul 2015 07:58:09 -0700 Subject: [PATCH 15/29] 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(-) 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); } From f2a5ded38592e5936a099ea6535ad5d3addcbc9d Mon Sep 17 00:00:00 2001 From: Nicolas Boichat Date: Fri, 17 Jul 2015 10:36:57 +0800 Subject: [PATCH 16/29] 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(+) 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) { From a798c24a69b64f09e2d323ac8155a36373e5d5fd Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Tue, 21 Jul 2015 11:51:35 +0200 Subject: [PATCH 17/29] 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(-) 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) { From b101acfabc9377469af3abfb7cb63112da367284 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Wed, 22 Jul 2015 11:27:45 +0800 Subject: [PATCH 18/29] 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(-) 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 From a6c2a32ac83567f15e9af3dcbc73148ce68b2ced Mon Sep 17 00:00:00 2001 From: Ben Zhang Date: Tue, 21 Jul 2015 14:46:26 -0700 Subject: [PATCH 19/29] 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(-) 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) From fa8173a3ef0570affde7da352de202190b3786c2 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Thu, 23 Jul 2015 23:22:26 +0800 Subject: [PATCH 20/29] 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(-) 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; From 44008f0896ae205b02b0882dbf807f0de149efc4 Mon Sep 17 00:00:00 2001 From: Dan Carpenter Date: Sat, 25 Jul 2015 03:03:38 +0300 Subject: [PATCH 21/29] 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(-) 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); } From e9c28e16a0b7071c88a206ad8ce0c73f6605bba7 Mon Sep 17 00:00:00 2001 From: Woodrow Shen Date: Sat, 25 Jul 2015 10:36:16 +0800 Subject: [PATCH 22/29] 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(+) 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}, From 9c6893e0be38b6ca9a56a854226e51dee0a16a5a Mon Sep 17 00:00:00 2001 From: Takashi Sakamoto Date: Sun, 26 Jul 2015 16:10:09 +0900 Subject: [PATCH 23/29] 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(+) 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; From b9d9c9efc292dd0ffe172780f915ed74eba3556c Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 27 Jul 2015 10:13:37 +0200 Subject: [PATCH 24/29] 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(-) 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 */ }; From 5ce000b297a1c1bb126a14b02acb40318b88a903 Mon Sep 17 00:00:00 2001 From: Woodrow Shen Date: Mon, 27 Jul 2015 18:34:31 +0800 Subject: [PATCH 25/29] 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(-) 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}, From 3a05d12f46cb95a6a685114819363a56e6170996 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 29 Jul 2015 09:04:52 +0200 Subject: [PATCH 26/29] 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(+) 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), From 2d1cb7f658fb9c3ba8f9dab8aca297d4dfdec835 Mon Sep 17 00:00:00 2001 From: Yao-Wen Mao Date: Wed, 29 Jul 2015 15:13:54 +0800 Subject: [PATCH 27/29] 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(+) 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 */ }; From 342e84490574cbb2a9c5b1d0886a112ad2bcf4d7 Mon Sep 17 00:00:00 2001 From: "U. Artie Eoff" Date: Tue, 28 Jul 2015 13:29:56 -0700 Subject: [PATCH 28/29] 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(-) 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; From 649ccd08534ee26deb2e5b08509800d0e95167f5 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 30 Jul 2015 22:30:29 +0200 Subject: [PATCH 29/29] 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(-) 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),