From d1555c407a65db42126b295425379acb393ba83a Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Andreas=20F=C3=A4rber?= Date: Mon, 28 Jul 2014 15:05:03 +0200 Subject: [PATCH 01/16] ASoC: axi: Fix ADI AXI SPDIF specification MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit The specification requires compatible = "adi,axi-spdif-1.00.a" but driver and example and file name indicate "adi,axi-spdif-tx-1.00.a". Change the specification to match the implementation. Acked-by: Lars-Peter Clausen Reviewed-by: Michal Simek Fixes: d7b528eff927 ("dt: Add bindings documentation for the ADI AXI-SPDIF audio controller") Signed-off-by: Andreas Färber Signed-off-by: Mark Brown --- Documentation/devicetree/bindings/sound/adi,axi-spdif-tx.txt | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/Documentation/devicetree/bindings/sound/adi,axi-spdif-tx.txt b/Documentation/devicetree/bindings/sound/adi,axi-spdif-tx.txt index 46f344965313..4eb7997674a0 100644 --- a/Documentation/devicetree/bindings/sound/adi,axi-spdif-tx.txt +++ b/Documentation/devicetree/bindings/sound/adi,axi-spdif-tx.txt @@ -1,7 +1,7 @@ ADI AXI-SPDIF controller Required properties: - - compatible : Must be "adi,axi-spdif-1.00.a" + - compatible : Must be "adi,axi-spdif-tx-1.00.a" - reg : Must contain SPDIF core's registers location and length - clocks : Pairs of phandle and specifier referencing the controller's clocks. The controller expects two clocks, the clock used for the AXI interface and From f294afed03b154fbfaa9a32a0ebe7abdbf98070c Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Sun, 17 Aug 2014 11:28:35 +0200 Subject: [PATCH 02/16] ASoC: Use dev_set_name() instead of init_name init_name is basically a hack and should only be used for statically allocated device structs. For dynamically allocated devices dev_set_name() should be used. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index d4bfd4a9076f..889f4e3d35dc 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1325,7 +1325,7 @@ static int soc_post_component_init(struct snd_soc_pcm_runtime *rtd, device_initialize(rtd->dev); rtd->dev->parent = rtd->card->dev; rtd->dev->release = rtd_release; - rtd->dev->init_name = name; + dev_set_name(rtd->dev, "%s", name); dev_set_drvdata(rtd->dev, rtd); mutex_init(&rtd->pcm_mutex); INIT_LIST_HEAD(&rtd->dpcm[SNDRV_PCM_STREAM_PLAYBACK].be_clients); From c8e6e960733f4a5835265c15429fced4d2f1595e Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Sun, 17 Aug 2014 16:18:19 +0200 Subject: [PATCH 03/16] ASoC: rcar: Use && instead of & for boolean expressions Sparse spits out the following warning: sound/soc/sh/rcar/gen.c:250:21: warning: dubious: x & !y It does this because sometimes mixing boolean and bit-wise logic has not the intended result. In this case we are fine, but replacing the bit-wise '&' with the boolean '&&' silences the sparse warning. The generated code for both cases is the same. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/sh/rcar/gen.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/sh/rcar/gen.c b/sound/soc/sh/rcar/gen.c index 3fdf3be7b99a..f95e7ab135e8 100644 --- a/sound/soc/sh/rcar/gen.c +++ b/sound/soc/sh/rcar/gen.c @@ -247,7 +247,7 @@ rsnd_gen2_dma_addr(struct rsnd_priv *priv, }; /* it shouldn't happen */ - if (use_dvc & !use_src) + if (use_dvc && !use_src) dev_err(dev, "DVC is selected without SRC\n"); /* use SSIU or SSI ? */ From 4d61b39bc117b36682c1dd67ee386960ae826bef Mon Sep 17 00:00:00 2001 From: "Subhransu S. Prusty" Date: Mon, 18 Aug 2014 14:53:03 +0530 Subject: [PATCH 04/16] ASoC: core: fix .info for SND_SOC_BYTES_TLV Commit 7523a271 - "ASoC: core: add a helper for extended byte controls using TLV" introduced support for TLV byte controls but had a typo for the info function, so fix the same Signed-off-by: Subhransu S. Prusty Signed-off-by: Vinod Koul Signed-off-by: Mark Brown --- include/sound/soc.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index be6ecae247b0..c83a334dd00f 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -277,7 +277,7 @@ .access = SNDRV_CTL_ELEM_ACCESS_TLV_READWRITE | \ SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK, \ .tlv.c = (snd_soc_bytes_tlv_callback), \ - .info = snd_soc_info_bytes_ext, \ + .info = snd_soc_bytes_info_ext, \ .private_value = (unsigned long)&(struct soc_bytes_ext) \ {.max = xcount, .get = xhandler_get, .put = xhandler_put, } } #define SOC_SINGLE_XR_SX(xname, xregbase, xregcount, xnbits, \ From aa47746269b0f87b3c042db7453b9e461029aed7 Mon Sep 17 00:00:00 2001 From: Rasmus Villemoes Date: Fri, 22 Aug 2014 11:25:13 +0200 Subject: [PATCH 05/16] ASoC: da732x: Fix typo in include guard Signed-off-by: Rasmus Villemoes Signed-off-by: Mark Brown --- sound/soc/codecs/da732x.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/da732x.h b/sound/soc/codecs/da732x.h index 1dceafeec415..f586cbd30b77 100644 --- a/sound/soc/codecs/da732x.h +++ b/sound/soc/codecs/da732x.h @@ -11,7 +11,7 @@ */ #ifndef __DA732X_H_ -#define __DA732X_H +#define __DA732X_H_ #include From d50884afdf592ebfe449b0a7cd741dd658716b13 Mon Sep 17 00:00:00 2001 From: Rasmus Villemoes Date: Fri, 22 Aug 2014 11:27:07 +0200 Subject: [PATCH 06/16] ASoC: tegra: Fix typo in include guard Signed-off-by: Rasmus Villemoes Acked-by: Thierry Reding Signed-off-by: Mark Brown --- sound/soc/tegra/tegra_asoc_utils.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/tegra/tegra_asoc_utils.h b/sound/soc/tegra/tegra_asoc_utils.h index 9577121ce971..ca8037634100 100644 --- a/sound/soc/tegra/tegra_asoc_utils.h +++ b/sound/soc/tegra/tegra_asoc_utils.h @@ -21,7 +21,7 @@ */ #ifndef __TEGRA_ASOC_UTILS_H__ -#define __TEGRA_ASOC_UTILS_H_ +#define __TEGRA_ASOC_UTILS_H__ struct clk; struct device; From f4821e8e8e957fe4c601a49b9a97b7399d5f7ab1 Mon Sep 17 00:00:00 2001 From: Jarkko Nikula Date: Tue, 26 Aug 2014 17:03:13 +0300 Subject: [PATCH 07/16] ASoC: rt5640: Do not allow regmap to use bulk read-write operations Debugging showed Realtek RT5642 doesn't support autoincrementing writes so driver should set the use_single_rw flag for regmap. Signed-off-by: Jarkko Nikula Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/codecs/rt5640.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c index 6bc6efdec550..f1ec6e6bd08a 100644 --- a/sound/soc/codecs/rt5640.c +++ b/sound/soc/codecs/rt5640.c @@ -2059,6 +2059,7 @@ static struct snd_soc_codec_driver soc_codec_dev_rt5640 = { static const struct regmap_config rt5640_regmap = { .reg_bits = 8, .val_bits = 16, + .use_single_rw = true, .max_register = RT5640_VENDOR_ID2 + 1 + (ARRAY_SIZE(rt5640_ranges) * RT5640_PR_SPACING), From 22e51345a9f272e20cea3d679dca8a0e19a178e1 Mon Sep 17 00:00:00 2001 From: Bard Liao Date: Wed, 27 Aug 2014 19:50:33 +0800 Subject: [PATCH 08/16] ASoC: rt5677: correct mismatch widget name We name MICBIAS1 in dapm widget, but micbias1 in route table. Signed-off-by: Bard Liao Signed-off-by: Mark Brown --- sound/soc/codecs/rt5677.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c index 67f14556462f..5337c448b5e3 100644 --- a/sound/soc/codecs/rt5677.c +++ b/sound/soc/codecs/rt5677.c @@ -2135,10 +2135,10 @@ static const struct snd_soc_dapm_route rt5677_dapm_routes[] = { { "BST2", NULL, "IN2P" }, { "BST2", NULL, "IN2N" }, - { "IN1P", NULL, "micbias1" }, - { "IN1N", NULL, "micbias1" }, - { "IN2P", NULL, "micbias1" }, - { "IN2N", NULL, "micbias1" }, + { "IN1P", NULL, "MICBIAS1" }, + { "IN1N", NULL, "MICBIAS1" }, + { "IN2P", NULL, "MICBIAS1" }, + { "IN2N", NULL, "MICBIAS1" }, { "ADC 1", NULL, "BST1" }, { "ADC 1", NULL, "ADC 1 power" }, From c98853aec1f7a05545642b6ca8591fd13b2fc7b6 Mon Sep 17 00:00:00 2001 From: Paul Handrigan Date: Thu, 28 Aug 2014 10:54:09 -0500 Subject: [PATCH 09/16] ASoC: cs4265: Fix clock rates in clock map table MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Reported-by: Zoltán Szenczi Signed-off-by: Paul Handrigan Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/codecs/cs4265.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/cs4265.c b/sound/soc/codecs/cs4265.c index a20b30ca52c0..8811689e372b 100644 --- a/sound/soc/codecs/cs4265.c +++ b/sound/soc/codecs/cs4265.c @@ -282,10 +282,10 @@ static const struct cs4265_clk_para clk_map_table[] = { /*64k*/ {8192000, 64000, 1, 0}, - {1228800, 64000, 1, 1}, - {1693440, 64000, 1, 2}, - {2457600, 64000, 1, 3}, - {3276800, 64000, 1, 4}, + {12288000, 64000, 1, 1}, + {16934400, 64000, 1, 2}, + {24576000, 64000, 1, 3}, + {32768000, 64000, 1, 4}, /* 88.2k */ {11289600, 88200, 1, 0}, From fb18cd2a62f597557d5078d8fa03bb6930fe839f Mon Sep 17 00:00:00 2001 From: Paul Handrigan Date: Thu, 28 Aug 2014 10:54:10 -0500 Subject: [PATCH 10/16] ASoC: cs4265: Fix setting of functional mode and clock divider MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Reported-by: Zoltán Szenczi Signed-off-by: Paul Handrigan Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/codecs/cs4265.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/cs4265.c b/sound/soc/codecs/cs4265.c index 8811689e372b..98523209f739 100644 --- a/sound/soc/codecs/cs4265.c +++ b/sound/soc/codecs/cs4265.c @@ -435,10 +435,10 @@ static int cs4265_pcm_hw_params(struct snd_pcm_substream *substream, index = cs4265_get_clk_index(cs4265->sysclk, params_rate(params)); if (index >= 0) { snd_soc_update_bits(codec, CS4265_ADC_CTL, - CS4265_ADC_FM, clk_map_table[index].fm_mode); + CS4265_ADC_FM, clk_map_table[index].fm_mode << 6); snd_soc_update_bits(codec, CS4265_MCLK_FREQ, CS4265_MCLK_FREQ_MASK, - clk_map_table[index].mclkdiv); + clk_map_table[index].mclkdiv << 4); } else { dev_err(codec->dev, "can't get correct mclk\n"); From 1033eb5b5aeeb526c22068e0fb0cef9f3c14231e Mon Sep 17 00:00:00 2001 From: Takashi Sakamoto Date: Fri, 29 Aug 2014 13:40:44 +0900 Subject: [PATCH 11/16] ALSA: dice: fix wrong channel mappping at higher sampling rate The channel mapping is initialized by amdtp_stream_set_parameters(), however Dice driver set it before calling this function. Furthermore, the setting is wrong because the index is the value of array, and vice versa. This commit moves codes for channel mapping after the function and set it correctly. Reported-by: Daniel Robbins Fixes: 10550bea44a8 ("ALSA: dice/firewire-lib: Keep dualwire mode but obsolete CIP_HI_DUALWIRE") Signed-off-by: Takashi Sakamoto Cc: # 3.16 Signed-off-by: Takashi Iwai --- sound/firewire/dice.c | 14 +++++++++----- 1 file changed, 9 insertions(+), 5 deletions(-) diff --git a/sound/firewire/dice.c b/sound/firewire/dice.c index a9a30c0161f1..4cf8eb704045 100644 --- a/sound/firewire/dice.c +++ b/sound/firewire/dice.c @@ -579,11 +579,6 @@ static int dice_hw_params(struct snd_pcm_substream *substream, return err; } - for (i = 0; i < channels; i++) { - dice->stream.pcm_positions[i * 2] = i; - dice->stream.pcm_positions[i * 2 + 1] = i + channels; - } - rate /= 2; channels *= 2; } @@ -591,6 +586,15 @@ static int dice_hw_params(struct snd_pcm_substream *substream, mode = rate_index_to_mode(rate_index); amdtp_stream_set_parameters(&dice->stream, rate, channels, dice->rx_midi_ports[mode]); + if (rate_index > 4) { + channels /= 2; + + for (i = 0; i < channels; i++) { + dice->stream.pcm_positions[i] = i * 2; + dice->stream.pcm_positions[i + channels] = i * 2 + 1; + } + } + amdtp_stream_set_pcm_format(&dice->stream, params_format(hw_params)); From 65845f29bec6bc17f80eff25c3bc39bcf3be9bf9 Mon Sep 17 00:00:00 2001 From: Takashi Sakamoto Date: Fri, 29 Aug 2014 13:40:45 +0900 Subject: [PATCH 12/16] ALSA: firewire-lib/dice: add arrangements of PCM pointer and interrupts for Dice quirk In IEC 61883-6, one data block transfers one event. In ALSA, the event equals one PCM frame, hence one data block transfers one PCM frame. But Dice has a quirk at higher sampling rate (176.4/192.0 kHz) that one data block transfers two PCM frames. Commit 10550bea44a8 ("ALSA: dice/firewire-lib: Keep dualwire mode but obsolete CIP_HI_DUALWIRE") moved some codes related to this quirk into Dice driver. But the commit forgot to add arrangements for PCM period interrupts and DMA pointer updates. As a result, Dice driver cannot work correctly at higher sampling rate. This commit adds 'double_pcm_frames' parameter to amdtp structure for this quirk. When this parameter is set, PCM period interrupts and DMA pointer updates occur at double speed than in IEC 61883-6. Reported-by: Daniel Robbins Fixes: 10550bea44a8 ("ALSA: dice/firewire-lib: Keep dualwire mode but obsolete CIP_HI_DUALWIRE") Signed-off-by: Takashi Sakamoto Cc: # 3.16 Signed-off-by: Takashi Iwai --- sound/firewire/amdtp.c | 11 ++++++++++- sound/firewire/amdtp.h | 1 + sound/firewire/dice.c | 15 +++++++++++---- 3 files changed, 22 insertions(+), 5 deletions(-) diff --git a/sound/firewire/amdtp.c b/sound/firewire/amdtp.c index f96bf4c7c232..95fc2eaf11dc 100644 --- a/sound/firewire/amdtp.c +++ b/sound/firewire/amdtp.c @@ -507,7 +507,16 @@ static void amdtp_pull_midi(struct amdtp_stream *s, static void update_pcm_pointers(struct amdtp_stream *s, struct snd_pcm_substream *pcm, unsigned int frames) -{ unsigned int ptr; +{ + unsigned int ptr; + + /* + * In IEC 61883-6, one data block represents one event. In ALSA, one + * event equals to one PCM frame. But Dice has a quirk to transfer + * two PCM frames in one data block. + */ + if (s->double_pcm_frames) + frames *= 2; ptr = s->pcm_buffer_pointer + frames; if (ptr >= pcm->runtime->buffer_size) diff --git a/sound/firewire/amdtp.h b/sound/firewire/amdtp.h index d8ee7b0e9386..4823c08196ac 100644 --- a/sound/firewire/amdtp.h +++ b/sound/firewire/amdtp.h @@ -125,6 +125,7 @@ struct amdtp_stream { unsigned int pcm_buffer_pointer; unsigned int pcm_period_pointer; bool pointer_flush; + bool double_pcm_frames; struct snd_rawmidi_substream *midi[AMDTP_MAX_CHANNELS_FOR_MIDI * 8]; diff --git a/sound/firewire/dice.c b/sound/firewire/dice.c index 4cf8eb704045..e3a04d69c853 100644 --- a/sound/firewire/dice.c +++ b/sound/firewire/dice.c @@ -567,10 +567,14 @@ static int dice_hw_params(struct snd_pcm_substream *substream, return err; /* - * At rates above 96 kHz, pretend that the stream runs at half the - * actual sample rate with twice the number of channels; two samples - * of a channel are stored consecutively in the packet. Requires - * blocking mode and PCM buffer size should be aligned to SYT_INTERVAL. + * At 176.4/192.0 kHz, Dice has a quirk to transfer two PCM frames in + * one data block of AMDTP packet. Thus sampling transfer frequency is + * a half of PCM sampling frequency, i.e. PCM frames at 192.0 kHz are + * transferred on AMDTP packets at 96 kHz. Two successive samples of a + * channel are stored consecutively in the packet. This quirk is called + * as 'Dual Wire'. + * For this quirk, blocking mode is required and PCM buffer size should + * be aligned to SYT_INTERVAL. */ channels = params_channels(hw_params); if (rate_index > 4) { @@ -581,6 +585,9 @@ static int dice_hw_params(struct snd_pcm_substream *substream, rate /= 2; channels *= 2; + dice->stream.double_pcm_frames = true; + } else { + dice->stream.double_pcm_frames = false; } mode = rate_index_to_mode(rate_index); From fdaf42c0105a24de8aefa60f6f7360842c4e673e Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Fri, 29 Aug 2014 13:30:23 +0300 Subject: [PATCH 13/16] ASoC: omap-twl4030: Fix typo in 2nd dai link's platform_name The platform_name should be omap-mcasp3 for the 2nd link which is used for voice connection. Reported-by: Tony Lindgren Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/omap/omap-twl4030.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/omap/omap-twl4030.c b/sound/soc/omap/omap-twl4030.c index f8a6adc2d81c..4336d1831485 100644 --- a/sound/soc/omap/omap-twl4030.c +++ b/sound/soc/omap/omap-twl4030.c @@ -260,7 +260,7 @@ static struct snd_soc_dai_link omap_twl4030_dai_links[] = { .stream_name = "TWL4030 Voice", .cpu_dai_name = "omap-mcbsp.3", .codec_dai_name = "twl4030-voice", - .platform_name = "omap-mcbsp.2", + .platform_name = "omap-mcbsp.3", .codec_name = "twl4030-codec", .dai_fmt = SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_IB_NF | SND_SOC_DAIFMT_CBM_CFM, From ff50479ad61069f3ee14863225aebe36d598e93e Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 1 Sep 2014 14:26:49 +0200 Subject: [PATCH 14/16] ALSA: hda - Fix digital mic on Acer Aspire 3830TG Acer Aspire 3830TG with CX20588 codec has a digital built-in mic that has the same problem like many others, the inverted signal in stereo. Apply the same fixup to this machine, too. Cc: Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_conexant.c | 9 ++++++++- 1 file changed, 8 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 6f2fa838b635..6e5d0cb4e3d7 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -217,6 +217,7 @@ enum { CXT_FIXUP_HEADPHONE_MIC_PIN, CXT_FIXUP_HEADPHONE_MIC, CXT_FIXUP_GPIO1, + CXT_FIXUP_ASPIRE_DMIC, CXT_FIXUP_THINKPAD_ACPI, CXT_FIXUP_OLPC_XO, CXT_FIXUP_CAP_MIX_AMP, @@ -664,6 +665,12 @@ static const struct hda_fixup cxt_fixups[] = { { } }, }, + [CXT_FIXUP_ASPIRE_DMIC] = { + .type = HDA_FIXUP_FUNC, + .v.func = cxt_fixup_stereo_dmic, + .chained = true, + .chain_id = CXT_FIXUP_GPIO1, + }, [CXT_FIXUP_THINKPAD_ACPI] = { .type = HDA_FIXUP_FUNC, .v.func = hda_fixup_thinkpad_acpi, @@ -744,7 +751,7 @@ static const struct hda_model_fixup cxt5051_fixup_models[] = { static const struct snd_pci_quirk cxt5066_fixups[] = { SND_PCI_QUIRK(0x1025, 0x0543, "Acer Aspire One 522", CXT_FIXUP_STEREO_DMIC), - SND_PCI_QUIRK(0x1025, 0x054c, "Acer Aspire 3830TG", CXT_FIXUP_GPIO1), + SND_PCI_QUIRK(0x1025, 0x054c, "Acer Aspire 3830TG", CXT_FIXUP_ASPIRE_DMIC), SND_PCI_QUIRK(0x1043, 0x138d, "Asus", CXT_FIXUP_HEADPHONE_MIC_PIN), SND_PCI_QUIRK(0x152d, 0x0833, "OLPC XO-1.5", CXT_FIXUP_OLPC_XO), SND_PCI_QUIRK(0x17aa, 0x20f2, "Lenovo T400", CXT_PINCFG_LENOVO_TP410), From e3c4a28b611b03d69bfbdffda985ef0dd94c2794 Mon Sep 17 00:00:00 2001 From: Xiubo Li Date: Mon, 1 Sep 2014 14:46:52 +0800 Subject: [PATCH 15/16] ASoC: simple-card: Fix bug of wrong decrement DT node's refcount DAI links's cpu_of_node's and codec_of_node's refcounts shouldn't be decremented immediately at the end of the probe() fucntion. Because we will still use them before the audio card is removed. Signed-off-by: Xiubo Li Signed-off-by: Mark Brown --- sound/soc/generic/simple-card.c | 8 ++++++++ 1 file changed, 8 insertions(+) diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index 159e517fa09a..cef7776b712c 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -481,12 +481,19 @@ static int asoc_simple_card_probe(struct platform_device *pdev) snd_soc_card_set_drvdata(&priv->snd_card, priv); ret = devm_snd_soc_register_card(&pdev->dev, &priv->snd_card); + if (ret >= 0) + return ret; err: asoc_simple_card_unref(pdev); return ret; } +static int asoc_simple_card_remove(struct platform_device *pdev) +{ + return asoc_simple_card_unref(pdev); +} + static const struct of_device_id asoc_simple_of_match[] = { { .compatible = "simple-audio-card", }, {}, @@ -500,6 +507,7 @@ static struct platform_driver asoc_simple_card = { .of_match_table = asoc_simple_of_match, }, .probe = asoc_simple_card_probe, + .remove = asoc_simple_card_remove, }; module_platform_driver(asoc_simple_card); From acf08081adb5e8fe0519eb97bb49797ef52614d6 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 2 Sep 2014 07:21:56 +0200 Subject: [PATCH 16/16] ALSA: hda - Fix COEF setups for ALC1150 codec ALC1150 codec seems to need the COEF- and PLL-setups just like its compatible ALC882 codec. Some machines (e.g. SunMicro X10SAT) show the problem like too low output volumes unless the COEF setup is applied. Reported-and-tested-by: Dana Goyette Cc: Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index d446ac3137b3..1ba22fb527c2 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -328,6 +328,7 @@ static void alc_auto_init_amp(struct hda_codec *codec, int type) case 0x10ec0885: case 0x10ec0887: /*case 0x10ec0889:*/ /* this causes an SPDIF problem */ + case 0x10ec0900: alc889_coef_init(codec); break; case 0x10ec0888: @@ -2350,6 +2351,7 @@ static int patch_alc882(struct hda_codec *codec) switch (codec->vendor_id) { case 0x10ec0882: case 0x10ec0885: + case 0x10ec0900: break; default: /* ALC883 and variants */