From 2596e07a3ed5a5f4d8b89be316c2b704d6f5dc5f Mon Sep 17 00:00:00 2001 From: Arnd Bergmann Date: Thu, 17 Mar 2016 18:23:40 +0100 Subject: regmap: fix documentation to match code The regmap binding talks about one thing, which is register endianess, and it gets almost every aspect of it wrong. This replaces the current text of the file with a version that makes more sense and that matches what we implement now. Signed-off-by: Arnd Bergmann Fixes: a06c488da0b0 ("regmap: Add explict native endian flag to DT bindings") Fixes: 275876e208e2 ("regmap: Add the DT binding documentation for endianness") Signed-off-by: Mark Brown --- .../devicetree/bindings/regmap/regmap.txt | 59 +++++++--------------- 1 file changed, 19 insertions(+), 40 deletions(-) (limited to 'Documentation') diff --git a/Documentation/devicetree/bindings/regmap/regmap.txt b/Documentation/devicetree/bindings/regmap/regmap.txt index e98a9652ccc8..0127be360fe8 100644 --- a/Documentation/devicetree/bindings/regmap/regmap.txt +++ b/Documentation/devicetree/bindings/regmap/regmap.txt @@ -1,50 +1,29 @@ -Device-Tree binding for regmap - -The endianness mode of CPU & Device scenarios: -Index Device Endianness properties ---------------------------------------------------- -1 BE 'big-endian' -2 LE 'little-endian' -3 Native 'native-endian' - -For one device driver, which will run in different scenarios above -on different SoCs using the devicetree, we need one way to simplify -this. +Devicetree binding for regmap Optional properties: -- {big,little,native}-endian: these are boolean properties, if absent - then the implementation will choose a default based on the device - being controlled. These properties are for register values and all - the buffers only. Native endian means that the CPU and device have - the same endianness. -Examples: -Scenario 1 : CPU in LE mode & device in LE mode. -dev: dev@40031000 { - compatible = "name"; - reg = <0x40031000 0x1000>; - ... -}; + little-endian, + big-endian, + native-endian: See common-properties.txt for a definition -Scenario 2 : CPU in LE mode & device in BE mode. -dev: dev@40031000 { - compatible = "name"; - reg = <0x40031000 0x1000>; - ... - big-endian; -}; +Note: +Regmap defaults to little-endian register access on MMIO based +devices, this is by far the most common setting. On CPU +architectures that typically run big-endian operating systems +(e.g. PowerPC), registers can be defined as big-endian and must +be marked that way in the devicetree. -Scenario 3 : CPU in BE mode & device in BE mode. -dev: dev@40031000 { - compatible = "name"; - reg = <0x40031000 0x1000>; - ... -}; +On SoCs that can be operated in both big-endian and little-endian +modes, with a single hardware switch controlling both the endianess +of the CPU and a byteswap for MMIO registers (e.g. many Broadcom MIPS +chips), "native-endian" is used to allow using the same device tree +blob in both cases. -Scenario 4 : CPU in BE mode & device in LE mode. +Examples: +Scenario 1 : a register set in big-endian mode. dev: dev@40031000 { - compatible = "name"; + compatible = "syscon"; reg = <0x40031000 0x1000>; + big-endian; ... - little-endian; }; -- cgit v1.2.3 From a42121b7a1d80a0eaac9ed1442760bf4e65f5352 Mon Sep 17 00:00:00 2001 From: Petr Kulhavy Date: Thu, 31 Mar 2016 18:41:27 +0200 Subject: ASoC: tas571x: new chip added into TAS571x binding This adds the TAS5721 into the TAS571x binding. Signed-off-by: Petr Kulhavy Reviewed-by: Kevin Cernekee Acked-by: Rob Herring Signed-off-by: Mark Brown --- Documentation/devicetree/bindings/sound/tas571x.txt | 10 ++++++++-- 1 file changed, 8 insertions(+), 2 deletions(-) (limited to 'Documentation') diff --git a/Documentation/devicetree/bindings/sound/tas571x.txt b/Documentation/devicetree/bindings/sound/tas571x.txt index 0ac31d8d5ac4..b4959f10b74b 100644 --- a/Documentation/devicetree/bindings/sound/tas571x.txt +++ b/Documentation/devicetree/bindings/sound/tas571x.txt @@ -1,4 +1,4 @@ -Texas Instruments TAS5711/TAS5717/TAS5719 stereo power amplifiers +Texas Instruments TAS5711/TAS5717/TAS5719/TAS5721 stereo power amplifiers The codec is controlled through an I2C interface. It also has two other signals that can be wired up to GPIOs: reset (strongly recommended), and @@ -6,7 +6,11 @@ powerdown (optional). Required properties: -- compatible: "ti,tas5711", "ti,tas5717", or "ti,tas5719" +- compatible: should be one of the following: + - "ti,tas5711", + - "ti,tas5717", + - "ti,tas5719", + - "ti,tas5721" - reg: The I2C address of the device - #sound-dai-cells: must be equal to 0 @@ -25,6 +29,8 @@ Optional properties: - PVDD_B-supply: regulator phandle for the PVDD_B supply (5711) - PVDD_C-supply: regulator phandle for the PVDD_C supply (5711) - PVDD_D-supply: regulator phandle for the PVDD_D supply (5711) +- DRVDD-supply: regulator phandle for the DRVDD supply (5721) +- PVDD-supply: regulator phandle for the PVDD supply (5721) Example: -- cgit v1.2.3 From 5ba10dd4a1727714faaac9a48f6aa69d8ee33248 Mon Sep 17 00:00:00 2001 From: Moise Gergaud Date: Thu, 31 Mar 2016 18:00:55 +0200 Subject: ASoC: sti: correct typo errors Signed-off-by: Moise Gergaud Acked-by: Arnaud Pouliquen Signed-off-by: Mark Brown --- .../devicetree/bindings/sound/st,sti-asoc-card.txt | 4 ++-- sound/soc/sti/uniperif.h | 22 +++++++++++----------- 2 files changed, 13 insertions(+), 13 deletions(-) (limited to 'Documentation') diff --git a/Documentation/devicetree/bindings/sound/st,sti-asoc-card.txt b/Documentation/devicetree/bindings/sound/st,sti-asoc-card.txt index 028fa1c82f50..f546dd914812 100644 --- a/Documentation/devicetree/bindings/sound/st,sti-asoc-card.txt +++ b/Documentation/devicetree/bindings/sound/st,sti-asoc-card.txt @@ -65,7 +65,7 @@ Example: reg = <0x8D82000 0x158>; interrupts = ; dmas = <&fdma0 4 0 1>; - dai-name = "Uni Player #1 (DAC)"; + dai-name = "Uni Player #2 (DAC)"; dma-names = "tx"; uniperiph-id = <2>; version = <5>; @@ -82,7 +82,7 @@ Example: interrupts = ; dmas = <&fdma0 7 0 1>; dma-names = "tx"; - dai-name = "Uni Player #1 (PIO)"; + dai-name = "Uni Player #3 (SPDIF)"; uniperiph-id = <3>; version = <5>; mode = "SPDIF"; diff --git a/sound/soc/sti/uniperif.h b/sound/soc/sti/uniperif.h index f0fd5a9944e9..1f82faafb869 100644 --- a/sound/soc/sti/uniperif.h +++ b/sound/soc/sti/uniperif.h @@ -25,7 +25,7 @@ writel_relaxed((((value) & mask) << shift), ip->base + offset) /* - * AUD_UNIPERIF_SOFT_RST reg + * UNIPERIF_SOFT_RST reg */ #define UNIPERIF_SOFT_RST_OFFSET(ip) 0x0000 @@ -50,7 +50,7 @@ UNIPERIF_SOFT_RST_SOFT_RST_MASK(ip)) /* - * AUD_UNIPERIF_FIFO_DATA reg + * UNIPERIF_FIFO_DATA reg */ #define UNIPERIF_FIFO_DATA_OFFSET(ip) 0x0004 @@ -58,7 +58,7 @@ writel_relaxed(value, ip->base + UNIPERIF_FIFO_DATA_OFFSET(ip)) /* - * AUD_UNIPERIF_CHANNEL_STA_REGN reg + * UNIPERIF_CHANNEL_STA_REGN reg */ #define UNIPERIF_CHANNEL_STA_REGN(ip, n) (0x0060 + (4 * n)) @@ -105,7 +105,7 @@ writel_relaxed(value, ip->base + UNIPERIF_CHANNEL_STA_REG5_OFFSET(ip)) /* - * AUD_UNIPERIF_ITS reg + * UNIPERIF_ITS reg */ #define UNIPERIF_ITS_OFFSET(ip) 0x000C @@ -143,7 +143,7 @@ 0 : (BIT(UNIPERIF_ITS_UNDERFLOW_REC_FAILED_SHIFT(ip)))) /* - * AUD_UNIPERIF_ITS_BCLR reg + * UNIPERIF_ITS_BCLR reg */ /* FIFO_ERROR */ @@ -160,7 +160,7 @@ writel_relaxed(value, ip->base + UNIPERIF_ITS_BCLR_OFFSET(ip)) /* - * AUD_UNIPERIF_ITM reg + * UNIPERIF_ITM reg */ #define UNIPERIF_ITM_OFFSET(ip) 0x0018 @@ -188,7 +188,7 @@ 0 : (BIT(UNIPERIF_ITM_UNDERFLOW_REC_FAILED_SHIFT(ip)))) /* - * AUD_UNIPERIF_ITM_BCLR reg + * UNIPERIF_ITM_BCLR reg */ #define UNIPERIF_ITM_BCLR_OFFSET(ip) 0x001c @@ -213,7 +213,7 @@ UNIPERIF_ITM_BCLR_DMA_ERROR_MASK(ip)) /* - * AUD_UNIPERIF_ITM_BSET reg + * UNIPERIF_ITM_BSET reg */ #define UNIPERIF_ITM_BSET_OFFSET(ip) 0x0020 @@ -767,7 +767,7 @@ SET_UNIPERIF_REG(ip, \ UNIPERIF_CTRL_OFFSET(ip), \ UNIPERIF_CTRL_READER_OUT_SEL_SHIFT(ip), \ - CORAUD_UNIPERIF_CTRL_READER_OUT_SEL_MASK(ip), 1) + UNIPERIF_CTRL_READER_OUT_SEL_MASK(ip), 1) /* UNDERFLOW_REC_WINDOW */ #define UNIPERIF_CTRL_UNDERFLOW_REC_WINDOW_SHIFT(ip) 20 @@ -1046,7 +1046,7 @@ UNIPERIF_STATUS_1_UNDERFLOW_DURATION_MASK(ip), value) /* - * AUD_UNIPERIF_CHANNEL_STA_REGN reg + * UNIPERIF_CHANNEL_STA_REGN reg */ #define UNIPERIF_CHANNEL_STA_REGN(ip, n) (0x0060 + (4 * n)) @@ -1057,7 +1057,7 @@ UNIPERIF_CHANNEL_STA_REGN(ip, n)) /* - * AUD_UNIPERIF_USER_VALIDITY reg + * UNIPERIF_USER_VALIDITY reg */ #define UNIPERIF_USER_VALIDITY_OFFSET(ip) 0x0090 -- cgit v1.2.3 From ee4c879b53eb5d757661b9bc8fcdf1a3d2e5fdfb Mon Sep 17 00:00:00 2001 From: Moise Gergaud Date: Tue, 19 Apr 2016 11:24:45 +0200 Subject: ASoC: sti-asoc-card: update tdm mode - Add "TDM" in the st,mode property list - st,mode property is also mandatory for reader - add tdm playback dai-link example Signed-off-by: Moise Gergaud Acked-by: Arnaud Pouliquen Signed-off-by: Mark Brown --- .../devicetree/bindings/sound/st,sti-asoc-card.txt | 44 +++++++++++++++++++--- 1 file changed, 39 insertions(+), 5 deletions(-) (limited to 'Documentation') diff --git a/Documentation/devicetree/bindings/sound/st,sti-asoc-card.txt b/Documentation/devicetree/bindings/sound/st,sti-asoc-card.txt index f546dd914812..4d9a83d9a017 100644 --- a/Documentation/devicetree/bindings/sound/st,sti-asoc-card.txt +++ b/Documentation/devicetree/bindings/sound/st,sti-asoc-card.txt @@ -37,17 +37,18 @@ Required properties: - dai-name: DAI name that describes the IP. + - IP mode: IP working mode depending on associated codec. + "HDMI" connected to HDMI codec and support IEC HDMI formats (player only). + "SPDIF" connected to SPDIF codec and support SPDIF formats (player only). + "PCM" PCM standard mode for I2S or TDM bus. + "TDM" TDM mode for TDM bus. + Required properties ("st,sti-uni-player" compatibility only): - clocks: CPU_DAI IP clock source, listed in the same order than the CPU_DAI properties. - uniperiph-id: internal SOC IP instance ID. - - IP mode: IP working mode depending on associated codec. - "HDMI" connected to HDMI codec IP and IEC HDMI formats. - "SPDIF"connected to SPDIF codec and support SPDIF formats. - "PCM" PCM standard mode for I2S or TDM bus. - Optional properties: - pinctrl-0: defined for CPU_DAI@1 and CPU_DAI@4 to describe I2S PIOs for external codecs connection. @@ -56,6 +57,22 @@ Optional properties: Example: + sti_uni_player1: sti-uni-player@1 { + compatible = "st,sti-uni-player"; + status = "okay"; + #sound-dai-cells = <0>; + st,syscfg = <&syscfg_core>; + clocks = <&clk_s_d0_flexgen CLK_PCM_1>; + reg = <0x8D81000 0x158>; + interrupts = ; + dmas = <&fdma0 3 0 1>; + st,dai-name = "Uni Player #1 (I2S)"; + dma-names = "tx"; + st,uniperiph-id = <1>; + st,version = <5>; + st,mode = "TDM"; + }; + sti_uni_player2: sti-uni-player@2 { compatible = "st,sti-uni-player"; status = "okay"; @@ -99,6 +116,7 @@ Example: dma-names = "rx"; dai-name = "Uni Reader #1 (HDMI RX)"; version = <3>; + st,mode = "PCM"; }; 2) sti-sas-codec: internal audio codec IPs driver @@ -152,4 +170,20 @@ Example of audio card declaration: sound-dai = <&sti_sasg_codec 0>; }; }; + simple-audio-card,dai-link@2 { + /* TDM playback */ + format = "left_j"; + frame-inversion = <1>; + cpu { + sound-dai = <&sti_uni_player1>; + dai-tdm-slot-num = <16>; + dai-tdm-slot-width = <16>; + dai-tdm-slot-tx-mask = + <1 1 1 1 0 0 0 0 0 0 1 1 0 0 1 1>; + }; + + codec { + sound-dai = <&sti_sasg_codec 3>; + }; + }; }; -- cgit v1.2.3 From d349caeb05104ef01392abc6c7cfc8ab516c7be4 Mon Sep 17 00:00:00 2001 From: PC Liao Date: Thu, 21 Apr 2016 19:38:14 +0800 Subject: ASoC: mediatek: Add second I2S on mt8173-rt5650 machine driver This patch adds second I2S connection to rt5650 codec for capture path on mt8173-rt5650 machine driver. Signed-off-by: PC Liao Signed-off-by: Mark Brown --- .../devicetree/bindings/sound/mt8173-rt5650.txt | 10 +++++ sound/soc/mediatek/mt8173-rt5650.c | 50 ++++++++++++++++++++-- 2 files changed, 57 insertions(+), 3 deletions(-) (limited to 'Documentation') diff --git a/Documentation/devicetree/bindings/sound/mt8173-rt5650.txt b/Documentation/devicetree/bindings/sound/mt8173-rt5650.txt index fe5a5ef1714d..5bfa6b60530b 100644 --- a/Documentation/devicetree/bindings/sound/mt8173-rt5650.txt +++ b/Documentation/devicetree/bindings/sound/mt8173-rt5650.txt @@ -5,11 +5,21 @@ Required properties: - mediatek,audio-codec: the phandles of rt5650 codecs - mediatek,platform: the phandle of MT8173 ASoC platform +Optional subnodes: +- codec-capture : the subnode of rt5650 codec capture +Required codec-capture subnode properties: +- sound-dai: audio codec dai name on capture path + <&rt5650 0> : Default setting. Connect rt5650 I2S1 for capture. (dai_name = rt5645-aif1) + <&rt5650 1> : Connect rt5650 I2S2 for capture. (dai_name = rt5645-aif2) + Example: sound { compatible = "mediatek,mt8173-rt5650"; mediatek,audio-codec = <&rt5650>; mediatek,platform = <&afe>; + codec-capture { + sound-dai = <&rt5650 1>; + }; }; diff --git a/sound/soc/mediatek/mt8173-rt5650.c b/sound/soc/mediatek/mt8173-rt5650.c index bb09bb1b7f1c..a27a6673dbe3 100644 --- a/sound/soc/mediatek/mt8173-rt5650.c +++ b/sound/soc/mediatek/mt8173-rt5650.c @@ -85,12 +85,29 @@ static int mt8173_rt5650_init(struct snd_soc_pcm_runtime *runtime) { struct snd_soc_card *card = runtime->card; struct snd_soc_codec *codec = runtime->codec_dais[0]->codec; + const char *codec_capture_dai = runtime->codec_dais[1]->name; int ret; rt5645_sel_asrc_clk_src(codec, - RT5645_DA_STEREO_FILTER | - RT5645_AD_STEREO_FILTER, + RT5645_DA_STEREO_FILTER, RT5645_CLK_SEL_I2S1_ASRC); + + if (!strcmp(codec_capture_dai, "rt5645-aif1")) { + rt5645_sel_asrc_clk_src(codec, + RT5645_AD_STEREO_FILTER, + RT5645_CLK_SEL_I2S1_ASRC); + } else if (!strcmp(codec_capture_dai, "rt5645-aif2")) { + rt5645_sel_asrc_clk_src(codec, + RT5645_AD_STEREO_FILTER, + RT5645_CLK_SEL_I2S2_ASRC); + } else { + dev_warn(card->dev, + "Only one dai codec found in DTS, enabled rt5645 AD filter\n"); + rt5645_sel_asrc_clk_src(codec, + RT5645_AD_STEREO_FILTER, + RT5645_CLK_SEL_I2S1_ASRC); + } + /* enable jack detection */ ret = snd_soc_card_jack_new(card, "Headset Jack", SND_JACK_HEADPHONE | SND_JACK_MICROPHONE | @@ -110,6 +127,11 @@ static int mt8173_rt5650_init(struct snd_soc_pcm_runtime *runtime) static struct snd_soc_dai_link_component mt8173_rt5650_codecs[] = { { + /* Playback */ + .dai_name = "rt5645-aif1", + }, + { + /* Capture */ .dai_name = "rt5645-aif1", }, }; @@ -149,7 +171,7 @@ static struct snd_soc_dai_link mt8173_rt5650_dais[] = { .cpu_dai_name = "I2S", .no_pcm = 1, .codecs = mt8173_rt5650_codecs, - .num_codecs = 1, + .num_codecs = 2, .init = mt8173_rt5650_init, .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS, @@ -177,6 +199,8 @@ static int mt8173_rt5650_dev_probe(struct platform_device *pdev) { struct snd_soc_card *card = &mt8173_rt5650_card; struct device_node *platform_node; + struct device_node *np; + const char *codec_capture_dai; int i, ret; platform_node = of_parse_phandle(pdev->dev.of_node, @@ -199,6 +223,26 @@ static int mt8173_rt5650_dev_probe(struct platform_device *pdev) "Property 'audio-codec' missing or invalid\n"); return -EINVAL; } + mt8173_rt5650_codecs[1].of_node = mt8173_rt5650_codecs[0].of_node; + + if (of_find_node_by_name(platform_node, "codec-capture")) { + np = of_get_child_by_name(pdev->dev.of_node, "codec-capture"); + if (!np) { + dev_err(&pdev->dev, + "%s: Can't find codec-capture DT node\n", + __func__); + return -EINVAL; + } + ret = snd_soc_of_get_dai_name(np, &codec_capture_dai); + if (ret < 0) { + dev_err(&pdev->dev, + "%s codec_capture_dai name fail %d\n", + __func__, ret); + return ret; + } + mt8173_rt5650_codecs[1].dai_name = codec_capture_dai; + } + card->dev = &pdev->dev; platform_set_drvdata(pdev, card); -- cgit v1.2.3 From c0133e3b0265341e7d62e150df18709af33c3a30 Mon Sep 17 00:00:00 2001 From: Koro Chen Date: Wed, 20 Apr 2016 10:59:56 +0200 Subject: ASoC: mediatek: Add HDMI dai-links in the mt8173-rt5650-rt5676 machine driver This creates pcmC0D2p for the HDMI playback in the same card. Signed-off-by: Koro Chen Signed-off-by: Philipp Zabel Signed-off-by: Mark Brown --- .../bindings/sound/mt8173-rt5650-rt5676.txt | 5 ++-- sound/soc/mediatek/Kconfig | 1 + sound/soc/mediatek/mt8173-rt5650-rt5676.c | 27 ++++++++++++++++++++++ 3 files changed, 31 insertions(+), 2 deletions(-) (limited to 'Documentation') diff --git a/Documentation/devicetree/bindings/sound/mt8173-rt5650-rt5676.txt b/Documentation/devicetree/bindings/sound/mt8173-rt5650-rt5676.txt index f205ce9e31dd..ac28cdb4910e 100644 --- a/Documentation/devicetree/bindings/sound/mt8173-rt5650-rt5676.txt +++ b/Documentation/devicetree/bindings/sound/mt8173-rt5650-rt5676.txt @@ -1,15 +1,16 @@ -MT8173 with RT5650 RT5676 CODECS +MT8173 with RT5650 RT5676 CODECS and HDMI via I2S Required properties: - compatible : "mediatek,mt8173-rt5650-rt5676" - mediatek,audio-codec: the phandles of rt5650 and rt5676 codecs + and of the hdmi encoder node - mediatek,platform: the phandle of MT8173 ASoC platform Example: sound { compatible = "mediatek,mt8173-rt5650-rt5676"; - mediatek,audio-codec = <&rt5650 &rt5676>; + mediatek,audio-codec = <&rt5650 &rt5676 &hdmi0>; mediatek,platform = <&afe>; }; diff --git a/sound/soc/mediatek/Kconfig b/sound/soc/mediatek/Kconfig index f7e789e97fbc..3abf51c07851 100644 --- a/sound/soc/mediatek/Kconfig +++ b/sound/soc/mediatek/Kconfig @@ -43,6 +43,7 @@ config SND_SOC_MT8173_RT5650_RT5676 depends on SND_SOC_MEDIATEK && I2C select SND_SOC_RT5645 select SND_SOC_RT5677 + select SND_SOC_HDMI_CODEC help This adds ASoC driver for Mediatek MT8173 boards with the RT5650 and RT5676 codecs. diff --git a/sound/soc/mediatek/mt8173-rt5650-rt5676.c b/sound/soc/mediatek/mt8173-rt5650-rt5676.c index 5c4c58c69c51..bb593926c62d 100644 --- a/sound/soc/mediatek/mt8173-rt5650-rt5676.c +++ b/sound/soc/mediatek/mt8173-rt5650-rt5676.c @@ -134,7 +134,9 @@ static struct snd_soc_dai_link_component mt8173_rt5650_rt5676_codecs[] = { enum { DAI_LINK_PLAYBACK, DAI_LINK_CAPTURE, + DAI_LINK_HDMI, DAI_LINK_CODEC_I2S, + DAI_LINK_HDMI_I2S, DAI_LINK_INTERCODEC }; @@ -161,6 +163,16 @@ static struct snd_soc_dai_link mt8173_rt5650_rt5676_dais[] = { .dynamic = 1, .dpcm_capture = 1, }, + [DAI_LINK_HDMI] = { + .name = "HDMI", + .stream_name = "HDMI PCM", + .cpu_dai_name = "HDMI", + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, + .dynamic = 1, + .dpcm_playback = 1, + }, /* Back End DAI links */ [DAI_LINK_CODEC_I2S] = { @@ -177,6 +189,13 @@ static struct snd_soc_dai_link mt8173_rt5650_rt5676_dais[] = { .dpcm_playback = 1, .dpcm_capture = 1, }, + [DAI_LINK_HDMI_I2S] = { + .name = "HDMI BE", + .cpu_dai_name = "HDMIO", + .no_pcm = 1, + .codec_dai_name = "i2s-hifi", + .dpcm_playback = 1, + }, /* rt5676 <-> rt5650 intercodec link: Sets rt5676 I2S2 as master */ [DAI_LINK_INTERCODEC] = { .name = "rt5650_rt5676 intercodec", @@ -251,6 +270,14 @@ static int mt8173_rt5650_rt5676_dev_probe(struct platform_device *pdev) mt8173_rt5650_rt5676_dais[DAI_LINK_INTERCODEC].codec_of_node = mt8173_rt5650_rt5676_codecs[1].of_node; + mt8173_rt5650_rt5676_dais[DAI_LINK_HDMI_I2S].codec_of_node = + of_parse_phandle(pdev->dev.of_node, "mediatek,audio-codec", 2); + if (!mt8173_rt5650_rt5676_dais[DAI_LINK_HDMI_I2S].codec_of_node) { + dev_err(&pdev->dev, + "Property 'audio-codec' missing or invalid\n"); + return -EINVAL; + } + card->dev = &pdev->dev; platform_set_drvdata(pdev, card); -- cgit v1.2.3 From 4e2f17be8f63a6ad9ebd4cdfce627e3dd25d80bb Mon Sep 17 00:00:00 2001 From: Andreas Dannenberg Date: Tue, 26 Apr 2016 17:15:56 -0500 Subject: ASoC: add TA5720 digital amplifier DT bindings The Texas Instruments TAS5720L/M device is a high-efficiency mono Class-D audio power amplifier optimized for high transient power capability to use the dynamic power headroom of small loudspeakers. Its digital time division multiplexed (TDM) interface enables up to 16 devices to share the same bus. Signed-off-by: Andreas Dannenberg Signed-off-by: Mark Brown --- .../devicetree/bindings/sound/tas5720.txt | 25 ++++++++++++++++++++++ 1 file changed, 25 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/tas5720.txt (limited to 'Documentation') diff --git a/Documentation/devicetree/bindings/sound/tas5720.txt b/Documentation/devicetree/bindings/sound/tas5720.txt new file mode 100644 index 000000000000..806ea7381483 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/tas5720.txt @@ -0,0 +1,25 @@ +Texas Instruments TAS5720 Mono Audio amplifier + +The TAS5720 serial control bus communicates through the I2C protocol only. The +serial bus is also used for periodic codec fault checking/reporting during +audio playback. For more product information please see the links below: + +http://www.ti.com/product/TAS5720L +http://www.ti.com/product/TAS5720M + +Required properties: + +- compatible : "ti,tas5720" +- reg : I2C slave address +- dvdd-supply : phandle to a 3.3-V supply for the digital circuitry +- pvdd-supply : phandle to a supply used for the Class-D amp and the analog + +Example: + +tas5720: tas5720@6c { + status = "okay"; + compatible = "ti,tas5720"; + reg = <0x6c>; + dvdd-supply = <&vdd_3v3_reg>; + pvdd-supply = <&_supply_reg>; +}; -- cgit v1.2.3 From ca2cd6bc6663314ba7fbf66ba7b14e099671420e Mon Sep 17 00:00:00 2001 From: anish kumar Date: Wed, 27 Apr 2016 15:39:05 -0700 Subject: ASoC: Add max98371 codec driver Signed-off-by: anish kumar Signed-off-by: Mark Brown --- .../devicetree/bindings/sound/max98371.txt | 17 + sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 1 + sound/soc/codecs/max98371.c | 442 +++++++++++++++++++++ sound/soc/codecs/max98371.h | 67 ++++ 5 files changed, 531 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/max98371.txt create mode 100644 sound/soc/codecs/max98371.c create mode 100644 sound/soc/codecs/max98371.h (limited to 'Documentation') diff --git a/Documentation/devicetree/bindings/sound/max98371.txt b/Documentation/devicetree/bindings/sound/max98371.txt new file mode 100644 index 000000000000..6c285235e64b --- /dev/null +++ b/Documentation/devicetree/bindings/sound/max98371.txt @@ -0,0 +1,17 @@ +max98371 codec + +This device supports I2C mode only. + +Required properties: + +- compatible : "maxim,max98371" +- reg : The chip select number on the I2C bus + +Example: + +&i2c { + max98371: max98371@0x31 { + compatible = "maxim,max98371"; + reg = <0x31>; + }; +}; diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 649e92a252ae..d764f1329042 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -79,6 +79,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_MAX98090 if I2C select SND_SOC_MAX98095 if I2C select SND_SOC_MAX98357A if GPIOLIB + select SND_SOC_MAX98371 if I2C select SND_SOC_MAX9867 if I2C select SND_SOC_MAX98925 if I2C select SND_SOC_MAX98926 if I2C @@ -522,6 +523,9 @@ config SND_SOC_MAX98095 config SND_SOC_MAX98357A tristate +config SND_SOC_MAX98371 + tristate + config SND_SOC_MAX9867 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 185a712a7fe7..92ef8719c311 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -74,6 +74,7 @@ snd-soc-max98088-objs := max98088.o snd-soc-max98090-objs := max98090.o snd-soc-max98095-objs := max98095.o snd-soc-max98357a-objs := max98357a.o +snd-soc-max98371-objs := max98371.o snd-soc-max9867-objs := max9867.o snd-soc-max98925-objs := max98925.o snd-soc-max98926-objs := max98926.o diff --git a/sound/soc/codecs/max98371.c b/sound/soc/codecs/max98371.c new file mode 100644 index 000000000000..21fae0932137 --- /dev/null +++ b/sound/soc/codecs/max98371.c @@ -0,0 +1,442 @@ +/* + * max98371.c -- ALSA SoC Stereo MAX98371 driver + * + * Copyright 2015-16 Maxim Integrated Products + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include "max98371.h" + +static const char *const monomix_text[] = { + "Left", "Right", "LeftRightDiv2", +}; + +static const char *const hpf_cutoff_txt[] = { + "Disable", "DC Block", "50Hz", + "100Hz", "200Hz", "400Hz", "800Hz", +}; + +static SOC_ENUM_SINGLE_DECL(max98371_monomix, MAX98371_MONOMIX_CFG, 0, + monomix_text); + +static SOC_ENUM_SINGLE_DECL(max98371_hpf_cutoff, MAX98371_HPF, 0, + hpf_cutoff_txt); + +static const DECLARE_TLV_DB_RANGE(max98371_dht_min_gain, + 0, 1, TLV_DB_SCALE_ITEM(537, 66, 0), + 2, 3, TLV_DB_SCALE_ITEM(677, 82, 0), + 4, 5, TLV_DB_SCALE_ITEM(852, 104, 0), + 6, 7, TLV_DB_SCALE_ITEM(1072, 131, 0), + 8, 9, TLV_DB_SCALE_ITEM(1350, 165, 0), + 10, 11, TLV_DB_SCALE_ITEM(1699, 101, 0), +); + +static const DECLARE_TLV_DB_RANGE(max98371_dht_max_gain, + 0, 1, TLV_DB_SCALE_ITEM(537, 66, 0), + 2, 3, TLV_DB_SCALE_ITEM(677, 82, 0), + 4, 5, TLV_DB_SCALE_ITEM(852, 104, 0), + 6, 7, TLV_DB_SCALE_ITEM(1072, 131, 0), + 8, 9, TLV_DB_SCALE_ITEM(1350, 165, 0), + 10, 11, TLV_DB_SCALE_ITEM(1699, 208, 0), +); + +static const DECLARE_TLV_DB_RANGE(max98371_dht_rot_gain, + 0, 1, TLV_DB_SCALE_ITEM(-50, -50, 0), + 2, 6, TLV_DB_SCALE_ITEM(-100, -100, 0), + 7, 8, TLV_DB_SCALE_ITEM(-800, -200, 0), + 9, 11, TLV_DB_SCALE_ITEM(-1200, -300, 0), + 12, 13, TLV_DB_SCALE_ITEM(-2000, -200, 0), + 14, 15, TLV_DB_SCALE_ITEM(-2500, -500, 0), +); + +static const struct reg_default max98371_reg[] = { + { 0x01, 0x00 }, + { 0x02, 0x00 }, + { 0x03, 0x00 }, + { 0x04, 0x00 }, + { 0x05, 0x00 }, + { 0x06, 0x00 }, + { 0x07, 0x00 }, + { 0x08, 0x00 }, + { 0x09, 0x00 }, + { 0x0A, 0x00 }, + { 0x10, 0x06 }, + { 0x11, 0x08 }, + { 0x14, 0x80 }, + { 0x15, 0x00 }, + { 0x16, 0x00 }, + { 0x18, 0x00 }, + { 0x19, 0x00 }, + { 0x1C, 0x00 }, + { 0x1D, 0x00 }, + { 0x1E, 0x00 }, + { 0x1F, 0x00 }, + { 0x20, 0x00 }, + { 0x21, 0x00 }, + { 0x22, 0x00 }, + { 0x23, 0x00 }, + { 0x24, 0x00 }, + { 0x25, 0x00 }, + { 0x26, 0x00 }, + { 0x27, 0x00 }, + { 0x28, 0x00 }, + { 0x29, 0x00 }, + { 0x2A, 0x00 }, + { 0x2B, 0x00 }, + { 0x2C, 0x00 }, + { 0x2D, 0x00 }, + { 0x2E, 0x0B }, + { 0x31, 0x00 }, + { 0x32, 0x18 }, + { 0x33, 0x00 }, + { 0x34, 0x00 }, + { 0x36, 0x00 }, + { 0x37, 0x00 }, + { 0x38, 0x00 }, + { 0x39, 0x00 }, + { 0x3A, 0x00 }, + { 0x3B, 0x00 }, + { 0x3B, 0x00 }, + { 0x3C, 0x00 }, + { 0x3D, 0x00 }, + { 0x3E, 0x00 }, + { 0x3F, 0x00 }, + { 0x40, 0x00 }, + { 0x41, 0x00 }, + { 0x42, 0x00 }, + { 0x43, 0x00 }, + { 0x4A, 0x00 }, + { 0x4B, 0x00 }, + { 0x4C, 0x00 }, + { 0x4D, 0x00 }, + { 0x4E, 0x00 }, + { 0x50, 0x00 }, + { 0x51, 0x00 }, + { 0x55, 0x00 }, + { 0x58, 0x00 }, + { 0x59, 0x00 }, + { 0x5C, 0x00 }, + { 0xFF, 0x43 }, +}; + +static bool max98371_volatile_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case MAX98371_IRQ_CLEAR1: + case MAX98371_IRQ_CLEAR2: + case MAX98371_IRQ_CLEAR3: + case MAX98371_VERSION: + return true; + default: + return false; + } +} + +static bool max98371_readable_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case MAX98371_SOFT_RESET: + return false; + default: + return true; + } +}; + +static const DECLARE_TLV_DB_RANGE(max98371_gain_tlv, + 0, 7, TLV_DB_SCALE_ITEM(0, 50, 0), + 8, 10, TLV_DB_SCALE_ITEM(400, 100, 0) +); + +static const DECLARE_TLV_DB_RANGE(max98371_noload_gain_tlv, + 0, 11, TLV_DB_SCALE_ITEM(950, 100, 0), +); + +static const DECLARE_TLV_DB_SCALE(digital_tlv, -6300, 50, 1); + +static const struct snd_kcontrol_new max98371_snd_controls[] = { + SOC_SINGLE_TLV("Speaker Volume", MAX98371_GAIN, + MAX98371_GAIN_SHIFT, (1<codec; + struct max98371_priv *max98371 = snd_soc_codec_get_drvdata(codec); + unsigned int val = 0; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + break; + default: + dev_err(codec->dev, "DAI clock mode unsupported"); + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + val |= 0; + break; + case SND_SOC_DAIFMT_RIGHT_J: + val |= MAX98371_DAI_RIGHT; + break; + case SND_SOC_DAIFMT_LEFT_J: + val |= MAX98371_DAI_LEFT; + break; + default: + dev_err(codec->dev, "DAI wrong mode unsupported"); + return -EINVAL; + } + regmap_update_bits(max98371->regmap, MAX98371_FMT, + MAX98371_FMT_MODE_MASK, val); + return 0; +} + +static int max98371_dai_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct max98371_priv *max98371 = snd_soc_codec_get_drvdata(codec); + int blr_clk_ratio, ch_size, channels = params_channels(params); + int rate = params_rate(params); + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S8: + regmap_update_bits(max98371->regmap, MAX98371_FMT, + MAX98371_FMT_MASK, MAX98371_DAI_CHANSZ_16); + ch_size = 8; + break; + case SNDRV_PCM_FORMAT_S16_LE: + regmap_update_bits(max98371->regmap, MAX98371_FMT, + MAX98371_FMT_MASK, MAX98371_DAI_CHANSZ_16); + ch_size = 16; + break; + case SNDRV_PCM_FORMAT_S24_LE: + regmap_update_bits(max98371->regmap, MAX98371_FMT, + MAX98371_FMT_MASK, MAX98371_DAI_CHANSZ_32); + ch_size = 24; + break; + case SNDRV_PCM_FORMAT_S32_LE: + regmap_update_bits(max98371->regmap, MAX98371_FMT, + MAX98371_FMT_MASK, MAX98371_DAI_CHANSZ_32); + ch_size = 32; + break; + default: + return -EINVAL; + } + + /* BCLK/LRCLK ratio calculation */ + blr_clk_ratio = channels * ch_size; + switch (blr_clk_ratio) { + case 32: + regmap_update_bits(max98371->regmap, + MAX98371_DAI_CLK, + MAX98371_DAI_BSEL_MASK, MAX98371_DAI_BSEL_32); + break; + case 48: + regmap_update_bits(max98371->regmap, + MAX98371_DAI_CLK, + MAX98371_DAI_BSEL_MASK, MAX98371_DAI_BSEL_48); + break; + case 64: + regmap_update_bits(max98371->regmap, + MAX98371_DAI_CLK, + MAX98371_DAI_BSEL_MASK, MAX98371_DAI_BSEL_64); + break; + default: + return -EINVAL; + } + + switch (rate) { + case 32000: + regmap_update_bits(max98371->regmap, + MAX98371_SPK_SR, + MAX98371_SPK_SR_MASK, MAX98371_SPK_SR_32); + break; + case 44100: + regmap_update_bits(max98371->regmap, + MAX98371_SPK_SR, + MAX98371_SPK_SR_MASK, MAX98371_SPK_SR_44); + break; + case 48000: + regmap_update_bits(max98371->regmap, + MAX98371_SPK_SR, + MAX98371_SPK_SR_MASK, MAX98371_SPK_SR_48); + break; + case 88200: + regmap_update_bits(max98371->regmap, + MAX98371_SPK_SR, + MAX98371_SPK_SR_MASK, MAX98371_SPK_SR_88); + break; + case 96000: + regmap_update_bits(max98371->regmap, + MAX98371_SPK_SR, + MAX98371_SPK_SR_MASK, MAX98371_SPK_SR_96); + break; + default: + return -EINVAL; + } + + /* enabling both the RX channels*/ + regmap_update_bits(max98371->regmap, MAX98371_MONOMIX_SRC, + MAX98371_MONOMIX_SRC_MASK, MONOMIX_RX_0_1); + regmap_update_bits(max98371->regmap, MAX98371_DAI_CHANNEL, + MAX98371_CHANNEL_MASK, MAX98371_CHANNEL_MASK); + return 0; +} + +static const struct snd_soc_dapm_widget max98371_dapm_widgets[] = { + SND_SOC_DAPM_DAC("DAC", NULL, MAX98371_SPK_ENABLE, 0, 0), + SND_SOC_DAPM_SUPPLY("Global Enable", MAX98371_GLOBAL_ENABLE, + 0, 0, NULL, 0), + SND_SOC_DAPM_OUTPUT("SPK_OUT"), +}; + +static const struct snd_soc_dapm_route max98371_audio_map[] = { + {"DAC", NULL, "HiFi Playback"}, + {"SPK_OUT", NULL, "DAC"}, + {"SPK_OUT", NULL, "Global Enable"}, +}; + +#define MAX98371_RATES SNDRV_PCM_RATE_8000_48000 +#define MAX98371_FORMATS (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_BE | \ + SNDRV_PCM_FMTBIT_S24_BE | SNDRV_PCM_FMTBIT_S32_BE) + +static const struct snd_soc_dai_ops max98371_dai_ops = { + .set_fmt = max98371_dai_set_fmt, + .hw_params = max98371_dai_hw_params, +}; + +static struct snd_soc_dai_driver max98371_dai[] = { + { + .name = "max98371-aif1", + .playback = { + .stream_name = "HiFi Playback", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = MAX98371_FORMATS, + }, + .ops = &max98371_dai_ops, + } +}; + +static const struct snd_soc_codec_driver max98371_codec = { + .controls = max98371_snd_controls, + .num_controls = ARRAY_SIZE(max98371_snd_controls), + .dapm_routes = max98371_audio_map, + .num_dapm_routes = ARRAY_SIZE(max98371_audio_map), + .dapm_widgets = max98371_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(max98371_dapm_widgets), +}; + +static const struct regmap_config max98371_regmap = { + .reg_bits = 8, + .val_bits = 8, + .max_register = MAX98371_VERSION, + .reg_defaults = max98371_reg, + .num_reg_defaults = ARRAY_SIZE(max98371_reg), + .volatile_reg = max98371_volatile_register, + .readable_reg = max98371_readable_register, + .cache_type = REGCACHE_RBTREE, +}; + +static int max98371_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct max98371_priv *max98371; + int ret, reg; + + max98371 = devm_kzalloc(&i2c->dev, + sizeof(*max98371), GFP_KERNEL); + if (!max98371) + return -ENOMEM; + + i2c_set_clientdata(i2c, max98371); + max98371->regmap = devm_regmap_init_i2c(i2c, &max98371_regmap); + if (IS_ERR(max98371->regmap)) { + ret = PTR_ERR(max98371->regmap); + dev_err(&i2c->dev, + "Failed to allocate regmap: %d\n", ret); + return ret; + } + + ret = regmap_read(max98371->regmap, MAX98371_VERSION, ®); + if (ret < 0) { + dev_info(&i2c->dev, "device error %d\n", ret); + return ret; + } + dev_info(&i2c->dev, "device version %x\n", reg); + + ret = snd_soc_register_codec(&i2c->dev, &max98371_codec, + max98371_dai, ARRAY_SIZE(max98371_dai)); + if (ret < 0) { + dev_err(&i2c->dev, "Failed to register codec: %d\n", ret); + return ret; + } + return ret; +} + +static int max98371_i2c_remove(struct i2c_client *client) +{ + snd_soc_unregister_codec(&client->dev); + return 0; +} + +static const struct i2c_device_id max98371_i2c_id[] = { + { "max98371", 0 }, +}; + +MODULE_DEVICE_TABLE(i2c, max98371_i2c_id); + +static const struct of_device_id max98371_of_match[] = { + { .compatible = "maxim,max98371", }, + { } +}; +MODULE_DEVICE_TABLE(of, max98371_of_match); + +static struct i2c_driver max98371_i2c_driver = { + .driver = { + .name = "max98371", + .owner = THIS_MODULE, + .pm = NULL, + .of_match_table = of_match_ptr(max98371_of_match), + }, + .probe = max98371_i2c_probe, + .remove = max98371_i2c_remove, + .id_table = max98371_i2c_id, +}; + +module_i2c_driver(max98371_i2c_driver); + +MODULE_AUTHOR("anish kumar "); +MODULE_DESCRIPTION("ALSA SoC MAX98371 driver"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/max98371.h b/sound/soc/codecs/max98371.h new file mode 100644 index 000000000000..9f6330964d98 --- /dev/null +++ b/sound/soc/codecs/max98371.h @@ -0,0 +1,67 @@ +/* + * max98371.h -- MAX98371 ALSA SoC Audio driver + * + * Copyright 2011-2012 Maxim Integrated Products + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef _MAX98371_H +#define _MAX98371_H + +#define MAX98371_IRQ_CLEAR1 0x01 +#define MAX98371_IRQ_CLEAR2 0x02 +#define MAX98371_IRQ_CLEAR3 0x03 +#define MAX98371_DAI_CLK 0x10 +#define MAX98371_DAI_BSEL_MASK 0xF +#define MAX98371_DAI_BSEL_32 2 +#define MAX98371_DAI_BSEL_48 3 +#define MAX98371_DAI_BSEL_64 4 +#define MAX98371_SPK_SR 0x11 +#define MAX98371_SPK_SR_MASK 0xF +#define MAX98371_SPK_SR_32 6 +#define MAX98371_SPK_SR_44 7 +#define MAX98371_SPK_SR_48 8 +#define MAX98371_SPK_SR_88 10 +#define MAX98371_SPK_SR_96 11 +#define MAX98371_DAI_CHANNEL 0x15 +#define MAX98371_CHANNEL_MASK 0x3 +#define MAX98371_MONOMIX_SRC 0x18 +#define MAX98371_MONOMIX_CFG 0x19 +#define MAX98371_HPF 0x1C +#define MAX98371_MONOMIX_SRC_MASK 0xFF +#define MONOMIX_RX_0_1 ((0x1)<<(4)) +#define M98371_DAI_CHANNEL_I2S 0x3 +#define MAX98371_DIGITAL_GAIN 0x2D +#define MAX98371_DIGITAL_GAIN_WIDTH 0x7 +#define MAX98371_GAIN 0x2E +#define MAX98371_GAIN_SHIFT 0x4 +#define MAX98371_GAIN_WIDTH 0x4 +#define MAX98371_DHT_MAX_WIDTH 4 +#define MAX98371_FMT 0x14 +#define MAX98371_CHANSZ_WIDTH 6 +#define MAX98371_FMT_MASK ((0x3)<<(MAX98371_CHANSZ_WIDTH)) +#define MAX98371_FMT_MODE_MASK ((0x7)<<(3)) +#define MAX98371_DAI_LEFT ((0x1)<<(3)) +#define MAX98371_DAI_RIGHT ((0x2)<<(3)) +#define MAX98371_DAI_CHANSZ_16 ((1)<<(MAX98371_CHANSZ_WIDTH)) +#define MAX98371_DAI_CHANSZ_24 ((2)<<(MAX98371_CHANSZ_WIDTH)) +#define MAX98371_DAI_CHANSZ_32 ((3)<<(MAX98371_CHANSZ_WIDTH)) +#define MAX98371_DHT 0x32 +#define MAX98371_DHT_STEP 0x3 +#define MAX98371_DHT_GAIN 0x31 +#define MAX98371_DHT_GAIN_WIDTH 0x4 +#define MAX98371_DHT_ROT_WIDTH 0x4 +#define MAX98371_SPK_ENABLE 0x4A +#define MAX98371_GLOBAL_ENABLE 0x50 +#define MAX98371_SOFT_RESET 0x51 +#define MAX98371_VERSION 0xFF + + +struct max98371_priv { + struct regmap *regmap; + struct snd_soc_codec *codec; +}; +#endif -- cgit v1.2.3 From c81aa7979432aa10f23656ef6fa113764eab5e5c Mon Sep 17 00:00:00 2001 From: Shmulik Ladkani Date: Fri, 6 May 2016 20:27:43 +0300 Subject: Documentation/networking: more accurate LCO explanation In few places the term "ones-complement sum" was used but the actual meaning is "the complement of the ones-complement sum". Also, avoid enclosing long statements with underscore, to ease readability. Signed-off-by: Shmulik Ladkani Acked-by: Edward Cree Signed-off-by: David S. Miller --- Documentation/networking/checksum-offloads.txt | 14 +++++++------- 1 file changed, 7 insertions(+), 7 deletions(-) (limited to 'Documentation') diff --git a/Documentation/networking/checksum-offloads.txt b/Documentation/networking/checksum-offloads.txt index de2a327766a7..56e36861245f 100644 --- a/Documentation/networking/checksum-offloads.txt +++ b/Documentation/networking/checksum-offloads.txt @@ -69,18 +69,18 @@ LCO: Local Checksum Offload LCO is a technique for efficiently computing the outer checksum of an encapsulated datagram when the inner checksum is due to be offloaded. The ones-complement sum of a correctly checksummed TCP or UDP packet is - equal to the sum of the pseudo header, because everything else gets - 'cancelled out' by the checksum field. This is because the sum was + equal to the complement of the sum of the pseudo header, because everything + else gets 'cancelled out' by the checksum field. This is because the sum was complemented before being written to the checksum field. More generally, this holds in any case where the 'IP-style' ones complement checksum is used, and thus any checksum that TX Checksum Offload supports. That is, if we have set up TX Checksum Offload with a start/offset pair, we - know that _after the device has filled in that checksum_, the ones + know that after the device has filled in that checksum, the ones complement sum from csum_start to the end of the packet will be equal to - _whatever value we put in the checksum field beforehand_. This allows us - to compute the outer checksum without looking at the payload: we simply - stop summing when we get to csum_start, then add the 16-bit word at - (csum_start + csum_offset). + the complement of whatever value we put in the checksum field beforehand. + This allows us to compute the outer checksum without looking at the payload: + we simply stop summing when we get to csum_start, then add the complement of + the 16-bit word at (csum_start + csum_offset). Then, when the true inner checksum is filled in (either by hardware or by skb_checksum_help()), the outer checksum will become correct by virtue of the arithmetic. -- cgit v1.2.3 From 0161028b7c8aebef64194d3d73e43bc3b53b5c66 Mon Sep 17 00:00:00 2001 From: Andy Lutomirski Date: Mon, 9 May 2016 15:48:51 -0700 Subject: perf/core: Change the default paranoia level to 2 Allowing unprivileged kernel profiling lets any user dump follow kernel control flow and dump kernel registers. This most likely allows trivial kASLR bypassing, and it may allow other mischief as well. (Off the top of my head, the PERF_SAMPLE_REGS_INTR output during /dev/urandom reads could be quite interesting.) Signed-off-by: Andy Lutomirski Acked-by: Kees Cook Signed-off-by: Linus Torvalds --- Documentation/sysctl/kernel.txt | 2 +- kernel/events/core.c | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) (limited to 'Documentation') diff --git a/Documentation/sysctl/kernel.txt b/Documentation/sysctl/kernel.txt index 57653a44b128..fcddfd5ded99 100644 --- a/Documentation/sysctl/kernel.txt +++ b/Documentation/sysctl/kernel.txt @@ -645,7 +645,7 @@ allowed to execute. perf_event_paranoid: Controls use of the performance events system by unprivileged -users (without CAP_SYS_ADMIN). The default value is 1. +users (without CAP_SYS_ADMIN). The default value is 2. -1: Allow use of (almost) all events by all users >=0: Disallow raw tracepoint access by users without CAP_IOC_LOCK diff --git a/kernel/events/core.c b/kernel/events/core.c index 4e2ebf6f2f1f..c0ded2416615 100644 --- a/kernel/events/core.c +++ b/kernel/events/core.c @@ -351,7 +351,7 @@ static struct srcu_struct pmus_srcu; * 1 - disallow cpu events for unpriv * 2 - disallow kernel profiling for unpriv */ -int sysctl_perf_event_paranoid __read_mostly = 1; +int sysctl_perf_event_paranoid __read_mostly = 2; /* Minimum for 512 kiB + 1 user control page */ int sysctl_perf_event_mlock __read_mostly = 512 + (PAGE_SIZE / 1024); /* 'free' kiB per user */ -- cgit v1.2.3