September 2019 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] SoC: simple-card-utils: set 0Hz to sysclk when shutdown
by Katsuhiro Suzuki 09 Sep '19

09 Sep '19

This patch set 0Hz to sysclk when shutdown the card. Some codecs set rate constraints that derives from sysclk. This mechanism works correctly if machine drivers give fixed frequency. But simple-audio and audio-graph card set variable clock rate if 'mclk-fs' property exists. In this case, rate constraints will go bad scenario. For example a codec accepts three limited rates (mclk / 256, mclk / 384, mclk / 512). Bad scenario as follows (mclk-fs = 256): - Initialize sysclk by correct value (Ex. 12.288MHz) - Codec set constraints of PCM rate by sysclk 48kHz (1/256), 32kHz (1/384), 24kHz (1/512) - Play 48kHz sound, it's acceptable - Sysclk is not changed - Play 32kHz sound, it's acceptable - Set sysclk to 8.192MHz (= fs * mclk-fs = 32k * 256) - Codec set constraints of PCM rate by sysclk 32kHz (1/256), 21.33kHz (1/384), 16kHz (1/512) - Play 48kHz again, but it's NOT acceptable because constraints do not allow 48kHz So codecs treat 0Hz sysclk as signal of applying no constraints to avoid this problem. Signed-off-by: Katsuhiro Suzuki <katsuhiro(a)katsuster.net> --- sound/soc/generic/simple-card-utils.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/sound/soc/generic/simple-card-utils.c b/sound/soc/generic/simple-card-utils.c index 556b1a789629..9b794775df53 100644 --- a/sound/soc/generic/simple-card-utils.c +++ b/sound/soc/generic/simple-card-utils.c @@ -213,10 +213,17 @@ EXPORT_SYMBOL_GPL(asoc_simple_startup); void asoc_simple_shutdown(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; struct asoc_simple_priv *priv = snd_soc_card_get_drvdata(rtd->card); struct simple_dai_props *dai_props = simple_priv_to_props(priv, rtd->num); + if (dai_props->mclk_fs) { + snd_soc_dai_set_sysclk(codec_dai, 0, 0, SND_SOC_CLOCK_IN); + snd_soc_dai_set_sysclk(cpu_dai, 0, 0, SND_SOC_CLOCK_OUT); + } + asoc_simple_clk_disable(dai_props->cpu_dai); asoc_simple_clk_disable(dai_props->codec_dai); -- 2.23.0.rc1

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[alsa-devel] [PATCH] ASoC: rockchip: ignore 0Hz sysclk
by Katsuhiro Suzuki 09 Sep '19

09 Sep '19

This patch ignores sysclk setting if it is 0Hz. Some codecs treat 0Hz sysclk as signal of applying no constraints. This driver does not have such feature but current implementation outputs 'Failed to set mclk' error message if machine driver sets 0Hz sysclk to this driver. Signed-off-by: Katsuhiro Suzuki <katsuhiro(a)katsuster.net> --- sound/soc/rockchip/rockchip_i2s.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/rockchip/rockchip_i2s.c b/sound/soc/rockchip/rockchip_i2s.c index 88ebaf6e1880..af2d5a6124c8 100644 --- a/sound/soc/rockchip/rockchip_i2s.c +++ b/sound/soc/rockchip/rockchip_i2s.c @@ -419,6 +419,9 @@ static int rockchip_i2s_set_sysclk(struct snd_soc_dai *cpu_dai, int clk_id, struct rk_i2s_dev *i2s = to_info(cpu_dai); int ret; + if (freq == 0) + return 0; + ret = clk_set_rate(i2s->mclk, freq); if (ret) dev_err(i2s->dev, "Fail to set mclk %d\n", ret); -- 2.23.0.rc1

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[alsa-devel] [PATCH 1/2] ASoC: es8316: fix redundant codes of clock
by Katsuhiro Suzuki 09 Sep '19

09 Sep '19

This patch removes redundant null checks for optional MCLK clock. And fix DT binding document for changing clock property to optional from required. Signed-off-by: Katsuhiro Suzuki <katsuhiro(a)katsuster.net> --- .../bindings/sound/everest,es8316.txt | 3 ++ sound/soc/codecs/es8316.c | 31 ++++++++----------- 2 files changed, 16 insertions(+), 18 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/everest,es8316.txt b/Documentation/devicetree/bindings/sound/everest,es8316.txt index aefcff9c48a2..1bf03c5f2af4 100644 --- a/Documentation/devicetree/bindings/sound/everest,es8316.txt +++ b/Documentation/devicetree/bindings/sound/everest,es8316.txt @@ -6,6 +6,9 @@ Required properties: - compatible : should be "everest,es8316" - reg : the I2C address of the device for I2C + +Optional properties: + - clocks : a list of phandle, should contain entries for clock-names - clock-names : should include as follows: "mclk" : master clock (MCLK) of the device diff --git a/sound/soc/codecs/es8316.c b/sound/soc/codecs/es8316.c index d07d50f51b28..1424ecd6952c 100644 --- a/sound/soc/codecs/es8316.c +++ b/sound/soc/codecs/es8316.c @@ -373,11 +373,9 @@ static int es8316_set_dai_sysclk(struct snd_soc_dai *codec_dai, if (freq == 0) return 0; - if (es8316->mclk) { - ret = clk_set_rate(es8316->mclk, freq); - if (ret) - return ret; - } + ret = clk_set_rate(es8316->mclk, freq); + if (ret) + return ret; /* Limit supported sample rates to ones that can be autodetected * by the codec running in slave mode. @@ -712,20 +710,18 @@ static int es8316_probe(struct snd_soc_component *component) es8316->component = component; - es8316->mclk = devm_clk_get(component->dev, "mclk"); - if (PTR_ERR(es8316->mclk) == -EPROBE_DEFER) - return -EPROBE_DEFER; + es8316->mclk = devm_clk_get_optional(component->dev, "mclk"); if (IS_ERR(es8316->mclk)) { - dev_err(component->dev, "clock is invalid, ignored\n"); - es8316->mclk = NULL; + dev_err(component->dev, "unable to get mclk\n"); + return PTR_ERR(es8316->mclk); } + if (!es8316->mclk) + dev_warn(component->dev, "assuming static mclk\n"); - if (es8316->mclk) { - ret = clk_prepare_enable(es8316->mclk); - if (ret) { - dev_err(component->dev, "unable to enable clock\n"); - return ret; - } + ret = clk_prepare_enable(es8316->mclk); + if (ret) { + dev_err(component->dev, "unable to enable mclk\n"); + return ret; } /* Reset codec and enable current state machine */ @@ -754,8 +750,7 @@ static void es8316_remove(struct snd_soc_component *component) { struct es8316_priv *es8316 = snd_soc_component_get_drvdata(component); - if (es8316->mclk) - clk_disable_unprepare(es8316->mclk); + clk_disable_unprepare(es8316->mclk); } static const struct snd_soc_component_driver soc_component_dev_es8316 = { -- 2.23.0.rc1

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[alsa-devel] [RFC PATCH] ASoC: ams-delta: Take control over audio mute GPIO pins
by Janusz Krzysztofik 09 Sep '19

09 Sep '19

Since commit 1137ceee76ba ("ARM: OMAP1: ams-delta: Don't request unused GPIOs"), on-board audio has appeared muted. It has been discovered that believed to be unused GPIO pins "hookflash1" and "hookflash2" need to be set low for audible sound in handsfree and handset mode respectively. According to Amstrad E3 wiki, the purpose of both pins hasn't been clearly identified. Original Amstrad software used to produce a high pulse on them when the phone was taken off hook or recall was pressed. With the current findings, we can assume the pins provide a kind of audio mute function, separately for handset and handsfree operation modes. Commit 2afdb4c41d78 ("ARM: OMAP1: ams-delta: Fix audio permanently muted") attempted to fix the issue temporarily by hogging the GPIO pin "hookflash1" renamed to "audio_mute", however the fix occurred incomplete as it restored audible sound only for handsfree mode. Stop hogging that pin, rename the pins to "handsfree_mute" and "handset_mute" respectively and implement appropriate DAPM event callbacks for "Speaker" and "Earpiece" DAPM widgets. Fixes: 1137ceee76ba ("ARM: OMAP1: ams-delta: Don't request unused GPIOs") Signed-off-by: Janusz Krzysztofik <jmkrzyszt(a)gmail.com> --- arch/arm/mach-omap1/board-ams-delta.c | 10 +++++---- sound/soc/ti/ams-delta.c | 31 +++++++++++++++++++++++++-- 2 files changed, 35 insertions(+), 6 deletions(-) diff --git a/arch/arm/mach-omap1/board-ams-delta.c b/arch/arm/mach-omap1/board-ams-delta.c index e47a6fbcfd6e..a2aa7a12b374 100644 --- a/arch/arm/mach-omap1/board-ams-delta.c +++ b/arch/arm/mach-omap1/board-ams-delta.c @@ -246,8 +246,8 @@ static struct platform_device latch2_gpio_device = { #define LATCH2_PIN_SCARD_CMDVCC 11 #define LATCH2_PIN_MODEM_NRESET 12 #define LATCH2_PIN_MODEM_CODEC 13 -#define LATCH2_PIN_AUDIO_MUTE 14 -#define LATCH2_PIN_HOOKFLASH 15 +#define LATCH2_PIN_HANDSFREE_MUTE 14 +#define LATCH2_PIN_HANDSET_MUTE 15 static struct regulator_consumer_supply modem_nreset_consumers[] = { REGULATOR_SUPPLY("RESET#", "serial8250.1"), @@ -476,6 +476,10 @@ static struct gpiod_lookup_table ams_delta_audio_gpio_table = { "hook_switch", 0), GPIO_LOOKUP(LATCH2_LABEL, LATCH2_PIN_MODEM_CODEC, "modem_codec", 0), + GPIO_LOOKUP(LATCH2_LABEL, LATCH2_PIN_HANDSFREE_MUTE, + "handsfree_mute", 0), + GPIO_LOOKUP(LATCH2_LABEL, LATCH2_PIN_HANDSET_MUTE, + "handset_mute", 0), { }, }, }; @@ -590,8 +594,6 @@ static int gpiochip_match_by_label(struct gpio_chip *chip, void *data) static struct gpiod_hog ams_delta_gpio_hogs[] = { GPIO_HOG(LATCH2_LABEL, LATCH2_PIN_KEYBRD_DATAOUT, "keybrd_dataout", GPIO_ACTIVE_HIGH, GPIOD_OUT_LOW), - GPIO_HOG(LATCH2_LABEL, LATCH2_PIN_AUDIO_MUTE, "audio_mute", - GPIO_ACTIVE_HIGH, GPIOD_OUT_LOW), {}, }; diff --git a/sound/soc/ti/ams-delta.c b/sound/soc/ti/ams-delta.c index dee8fc70a64f..8e2fb81ad05c 100644 --- a/sound/soc/ti/ams-delta.c +++ b/sound/soc/ti/ams-delta.c @@ -23,14 +23,31 @@ #include "omap-mcbsp.h" #include "../codecs/cx20442.h" +static struct gpio_desc *handset_mute; +static struct gpio_desc *handsfree_mute; + +static int ams_delta_event_handset(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *k, int event) +{ + gpiod_set_value_cansleep(handset_mute, !SND_SOC_DAPM_EVENT_ON(event)); + return 0; +} + +static int ams_delta_event_handsfree(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *k, int event) +{ + gpiod_set_value_cansleep(handsfree_mute, !SND_SOC_DAPM_EVENT_ON(event)); + return 0; +} + /* Board specific DAPM widgets */ static const struct snd_soc_dapm_widget ams_delta_dapm_widgets[] = { /* Handset */ SND_SOC_DAPM_MIC("Mouthpiece", NULL), - SND_SOC_DAPM_HP("Earpiece", NULL), + SND_SOC_DAPM_HP("Earpiece", ams_delta_event_handset), /* Handsfree/Speakerphone */ SND_SOC_DAPM_MIC("Microphone", NULL), - SND_SOC_DAPM_SPK("Speaker", NULL), + SND_SOC_DAPM_SPK("Speaker", ams_delta_event_handsfree), }; /* How they are connected to codec pins */ @@ -542,6 +559,16 @@ static int ams_delta_probe(struct platform_device *pdev) card->dev = &pdev->dev; + handset_mute = devm_gpiod_get(card->dev, "handset_mute", + GPIOD_OUT_HIGH); + if (IS_ERR(handset_mute)) + return PTR_ERR(handset_mute); + + handsfree_mute = devm_gpiod_get(card->dev, "handsfree_mute", + GPIOD_OUT_HIGH); + if (IS_ERR(handsfree_mute)) + return PTR_ERR(handsfree_mute); + ret = snd_soc_register_card(card); if (ret) { dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", ret); -- 2.21.0

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[alsa-devel] [PATCH] ASoC: rt1011: make array pd static const, makes object smaller
by Colin King 09 Sep '19

09 Sep '19

From: Colin Ian King <colin.king(a)canonical.com> Don't populate the array pd on the stack but instead make it static const. Makes the object code smaller by 100 bytes. Before: text data bss dec hex filename 51463 13016 128 64607 fc5f sound/soc/codecs/rt1011.o After: text data bss dec hex filename 51299 13080 128 64507 fbfb sound/soc/codecs/rt1011.o (gcc version 9.2.1, amd64) Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/codecs/rt1011.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt1011.c b/sound/soc/codecs/rt1011.c index f9235764ad32..be1e276e3631 100644 --- a/sound/soc/codecs/rt1011.c +++ b/sound/soc/codecs/rt1011.c @@ -1519,7 +1519,8 @@ static const struct snd_soc_dapm_route rt1011_dapm_routes[] = { static int rt1011_get_clk_info(int sclk, int rate) { - int i, pd[] = {1, 2, 3, 4, 6, 8, 12, 16}; + int i; + static const int pd[] = {1, 2, 3, 4, 6, 8, 12, 16}; if (sclk <= 0 || rate <= 0) return -EINVAL; -- 2.20.1

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[alsa-devel] [PATCH] ASoC: codecs: ad193x: make two arrays static const, makes object smaller
by Colin King 09 Sep '19

09 Sep '19

From: Colin Ian King <colin.king(a)canonical.com> Don't populate the arrays on the stack but instead make them static const. Makes the object code smaller by 37 bytes. Before: text data bss dec hex filename 16253 7200 0 23453 5b9d sound/soc/codecs/ad193x.o After: text data bss dec hex filename 16056 7360 0 23416 5b78 sound/soc/codecs/ad193x.o (gcc version 9.2.1, amd64) Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/codecs/ad193x.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/ad193x.c b/sound/soc/codecs/ad193x.c index fb04c9379b71..980e024a5720 100644 --- a/sound/soc/codecs/ad193x.c +++ b/sound/soc/codecs/ad193x.c @@ -416,7 +416,7 @@ static struct snd_soc_dai_driver ad193x_no_adc_dai = { /* codec register values to set after reset */ static void ad193x_reg_default_init(struct ad193x_priv *ad193x) { - const struct reg_sequence reg_init[] = { + static const struct reg_sequence reg_init[] = { { 0, 0x99 }, /* PLL_CLK_CTRL0: pll input: mclki/xi 12.288Mhz */ { 1, 0x04 }, /* PLL_CLK_CTRL1: no on-chip Vref */ { 2, 0x40 }, /* DAC_CTRL0: TDM mode */ @@ -432,7 +432,7 @@ static void ad193x_reg_default_init(struct ad193x_priv *ad193x) { 12, 0x00 }, /* DAC_L4_VOL: no attenuation */ { 13, 0x00 }, /* DAC_R4_VOL: no attenuation */ }; - const struct reg_sequence reg_adc_init[] = { + static const struct reg_sequence reg_adc_init[] = { { 14, 0x03 }, /* ADC_CTRL0: high-pass filter enable */ { 15, 0x43 }, /* ADC_CTRL1: sata delay=1, adc aux mode */ { 16, 0x00 }, /* ADC_CTRL2: reset */ -- 2.20.1

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[alsa-devel] [PATCH] ASoC: rt1308: make array pd static const, makes object smaller
by Colin King 09 Sep '19

09 Sep '19

From: Colin Ian King <colin.king(a)canonical.com> Don't populate the array pd on the stack but instead make it static const. Makes the object code smaller by 82 bytes. Before: text data bss dec hex filename 26548 7288 64 33900 846c sound/soc/codecs/rt1308.o After: text data bss dec hex filename 26370 7384 64 33818 841a sound/soc/codecs/rt1308.o (gcc version 9.2.1, amd64) Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/codecs/rt1308.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt1308.c b/sound/soc/codecs/rt1308.c index 6f2ee6809dbb..b75931a69a1c 100644 --- a/sound/soc/codecs/rt1308.c +++ b/sound/soc/codecs/rt1308.c @@ -436,7 +436,8 @@ static const struct snd_soc_dapm_route rt1308_dapm_routes[] = { static int rt1308_get_clk_info(int sclk, int rate) { - int i, pd[] = {1, 2, 3, 4, 6, 8, 12, 16}; + int i; + static const int pd[] = {1, 2, 3, 4, 6, 8, 12, 16}; if (sclk <= 0 || rate <= 0) return -EINVAL; -- 2.20.1

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[alsa-devel] [PATCH] ASoC: rt1305: make array pd static const, makes object smaller
by Colin King 09 Sep '19

09 Sep '19

From: Colin Ian King <colin.king(a)canonical.com> Don't populate the array pd on the stack but instead make it static const. Makes the object code smaller by 93 bytes. Before: text data bss dec hex filename 38961 9784 64 48809 bea9 sound/soc/codecs/rt1305.o After: text data bss dec hex filename 38804 9848 64 48716 be4c sound/soc/codecs/rt1305.o (gcc version 9.2.1, amd64) Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/codecs/rt1305.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt1305.c b/sound/soc/codecs/rt1305.c index 9909369483f0..e27742abfa76 100644 --- a/sound/soc/codecs/rt1305.c +++ b/sound/soc/codecs/rt1305.c @@ -608,7 +608,8 @@ static const struct snd_soc_dapm_route rt1305_dapm_routes[] = { static int rt1305_get_clk_info(int sclk, int rate) { - int i, pd[] = {1, 2, 3, 4, 6, 8, 12, 16}; + int i; + static const int pd[] = {1, 2, 3, 4, 6, 8, 12, 16}; if (sclk <= 0 || rate <= 0) return -EINVAL; -- 2.20.1

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[alsa-devel] [PATCH resend 00/11] ASoC: soc-core cleanup
by Kuninori Morimoto 09 Sep '19

09 Sep '19

Hi Mark These are resend of soc-core cleanup patches which are not yet accepted/reviewed. These are based on latest mark/for-v5.4 branch. Kuninori Morimoto (11): ASoC: soc-core: add comment to jack at soc_remove_component() ASoC: soc-core: self contained soc_probe_link_components() ASoC: soc-core: self contained soc_remove_link_components() ASoC: soc-core: self contained soc_remove_link_dais() ASoC: soc-core: move soc_probe_dai() next to soc_remove_dai() ASoC: soc-core: add new soc_link_init() ASoC: soc-core: self contained soc_probe_link_dais() ASoC: soc-core: move soc_probe_link_dais() next to soc_remove_link_dais() ASoC: soc-core: self contained soc_bind_aux_dev() ASoC: soc-core: add soc_unbind_aux_dev() ASoC: soc-core: self contained soc_unbind_aux_dev() sound/soc/soc-core.c | 264 +++++++++++++++++++++++++++------------------------ 1 file changed, 139 insertions(+), 125 deletions(-) -- 2.7.4

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[alsa-devel] [PATCH v2] drm: bridge/dw_hdmi: add audio sample channel status setting
by Cheng-Yi Chiang 09 Sep '19

09 Sep '19

From: Yakir Yang <ykk(a)rock-chips.com> When transmitting IEC60985 linear PCM audio, we configure the Aduio Sample Channel Status information of all the channel status bits in the IEC60958 frame. Refer to 60958-3 page 10 for frequency, original frequency, and wordlength setting. This fix the issue that audio does not come out on some monitors (e.g. LG 22CV241) Note that these registers are only for interfaces: I2S audio interface, General Purpose Audio (GPA), or AHB audio DMA (AHBAUDDMA). For S/PDIF interface this information comes from the stream. Currently this function dw_hdmi_set_channel_status is only called from dw-hdmi-i2s-audio in I2S setup. Signed-off-by: Yakir Yang <ykk(a)rock-chips.com> Signed-off-by: Cheng-Yi Chiang <cychiang(a)chromium.org> --- Original patch by Yakir Yang is at https://lore.kernel.org/patchwork/patch/539653/ Change from v1 to v2: 1. Remove the version check because this will only be called by dw-hdmi-i2s-audio, and the registers are available in I2S setup. 2. Set these registers in dw_hdmi_i2s_hw_params. 3. Fix the sample width setting so it can use 16 or 24 bits. .../drm/bridge/synopsys/dw-hdmi-i2s-audio.c | 1 + drivers/gpu/drm/bridge/synopsys/dw-hdmi.c | 70 +++++++++++++++++++ drivers/gpu/drm/bridge/synopsys/dw-hdmi.h | 20 ++++++ include/drm/bridge/dw_hdmi.h | 2 + 4 files changed, 93 insertions(+) diff --git a/drivers/gpu/drm/bridge/synopsys/dw-hdmi-i2s-audio.c b/drivers/gpu/drm/bridge/synopsys/dw-hdmi-i2s-audio.c index 34d8e837555f..b801a28b0f17 100644 --- a/drivers/gpu/drm/bridge/synopsys/dw-hdmi-i2s-audio.c +++ b/drivers/gpu/drm/bridge/synopsys/dw-hdmi-i2s-audio.c @@ -102,6 +102,7 @@ static int dw_hdmi_i2s_hw_params(struct device *dev, void *data, } dw_hdmi_set_sample_rate(hdmi, hparms->sample_rate); + dw_hdmi_set_channel_status(hdmi, hparms->sample_width); dw_hdmi_set_channel_count(hdmi, hparms->channels); dw_hdmi_set_channel_allocation(hdmi, hparms->cea.channel_allocation); diff --git a/drivers/gpu/drm/bridge/synopsys/dw-hdmi.c b/drivers/gpu/drm/bridge/synopsys/dw-hdmi.c index bd65d0479683..d1daa369c8ae 100644 --- a/drivers/gpu/drm/bridge/synopsys/dw-hdmi.c +++ b/drivers/gpu/drm/bridge/synopsys/dw-hdmi.c @@ -582,6 +582,76 @@ static unsigned int hdmi_compute_n(unsigned int freq, unsigned long pixel_clk) return n; } +/* + * When transmitting IEC60958 linear PCM audio, these registers allow to + * configure the channel status information of all the channel status + * bits in the IEC60958 frame. For the moment this configuration is only + * used when the I2S audio interface, General Purpose Audio (GPA), + * or AHB audio DMA (AHBAUDDMA) interface is active + * (for S/PDIF interface this information comes from the stream). + */ +void dw_hdmi_set_channel_status(struct dw_hdmi *hdmi, + unsigned int sample_width) +{ + u8 aud_schnl_samplerate; + u8 aud_schnl_8; + u8 word_length_bits; + + switch (hdmi->sample_rate) { + case 32000: + aud_schnl_samplerate = HDMI_FC_AUDSCHNLS7_SMPRATE_32K; + break; + case 44100: + aud_schnl_samplerate = HDMI_FC_AUDSCHNLS7_SMPRATE_44K1; + break; + case 48000: + aud_schnl_samplerate = HDMI_FC_AUDSCHNLS7_SMPRATE_48K; + break; + case 88200: + aud_schnl_samplerate = HDMI_FC_AUDSCHNLS7_SMPRATE_88K2; + break; + case 96000: + aud_schnl_samplerate = HDMI_FC_AUDSCHNLS7_SMPRATE_96K; + break; + case 176400: + aud_schnl_samplerate = HDMI_FC_AUDSCHNLS7_SMPRATE_176K4; + break; + case 192000: + aud_schnl_samplerate = HDMI_FC_AUDSCHNLS7_SMPRATE_192K; + break; + case 768000: + aud_schnl_samplerate = HDMI_FC_AUDSCHNLS7_SMPRATE_768K; + break; + default: + dev_warn(hdmi->dev, "Unsupported audio sample rate (%u)\n", + hdmi->sample_rate); + return; + } + + /* set channel status register */ + hdmi_modb(hdmi, aud_schnl_samplerate, HDMI_FC_AUDSCHNLS7_SMPRATE_MASK, + HDMI_FC_AUDSCHNLS7); + + /* + * Set original frequency to be the same as frequency. + * Use one-complement value as stated in IEC60958-3 page 13. + */ + aud_schnl_8 = (~aud_schnl_samplerate) << + HDMI_FC_AUDSCHNLS8_ORIGSAMPFREQ_OFFSET; + + /* + * Refer to IEC60958-3 page 12. We can accept 16 bits or 24 bits. + * Otherwise, set the register to 0t o indicate using default value. + */ + word_length_bits = (sample_width == 16) ? 0x2 : + ((sample_width == 24) ? 0xb : 0); + + aud_schnl_8 |= word_length_bits << HDMI_FC_AUDSCHNLS8_WORDLEGNTH_OFFSET; + + hdmi_writeb(hdmi, aud_schnl_8, HDMI_FC_AUDSCHNLS8); +} +EXPORT_SYMBOL_GPL(dw_hdmi_set_channel_status); + static void hdmi_set_clk_regenerator(struct dw_hdmi *hdmi, unsigned long pixel_clk, unsigned int sample_rate) { diff --git a/drivers/gpu/drm/bridge/synopsys/dw-hdmi.h b/drivers/gpu/drm/bridge/synopsys/dw-hdmi.h index 6988f12d89d9..619ebc1c8354 100644 --- a/drivers/gpu/drm/bridge/synopsys/dw-hdmi.h +++ b/drivers/gpu/drm/bridge/synopsys/dw-hdmi.h @@ -158,6 +158,17 @@ #define HDMI_FC_SPDDEVICEINF 0x1062 #define HDMI_FC_AUDSCONF 0x1063 #define HDMI_FC_AUDSSTAT 0x1064 +#define HDMI_FC_AUDSV 0x1065 +#define HDMI_FC_AUDSU 0x1066 +#define HDMI_FC_AUDSCHNLS0 0x1067 +#define HDMI_FC_AUDSCHNLS1 0x1068 +#define HDMI_FC_AUDSCHNLS2 0x1069 +#define HDMI_FC_AUDSCHNLS3 0x106a +#define HDMI_FC_AUDSCHNLS4 0x106b +#define HDMI_FC_AUDSCHNLS5 0x106c +#define HDMI_FC_AUDSCHNLS6 0x106d +#define HDMI_FC_AUDSCHNLS7 0x106e +#define HDMI_FC_AUDSCHNLS8 0x106f #define HDMI_FC_DATACH0FILL 0x1070 #define HDMI_FC_DATACH1FILL 0x1071 #define HDMI_FC_DATACH2FILL 0x1072 @@ -706,6 +717,15 @@ enum { /* HDMI_FC_AUDSCHNLS7 field values */ HDMI_FC_AUDSCHNLS7_ACCURACY_OFFSET = 4, HDMI_FC_AUDSCHNLS7_ACCURACY_MASK = 0x30, + HDMI_FC_AUDSCHNLS7_SMPRATE_MASK = 0x0f, + HDMI_FC_AUDSCHNLS7_SMPRATE_192K = 0xe, + HDMI_FC_AUDSCHNLS7_SMPRATE_176K4 = 0xc, + HDMI_FC_AUDSCHNLS7_SMPRATE_96K = 0xa, + HDMI_FC_AUDSCHNLS7_SMPRATE_768K = 0x9, + HDMI_FC_AUDSCHNLS7_SMPRATE_88K2 = 0x8, + HDMI_FC_AUDSCHNLS7_SMPRATE_32K = 0x3, + HDMI_FC_AUDSCHNLS7_SMPRATE_48K = 0x2, + HDMI_FC_AUDSCHNLS7_SMPRATE_44K1 = 0x0, /* HDMI_FC_AUDSCHNLS8 field values */ HDMI_FC_AUDSCHNLS8_ORIGSAMPFREQ_MASK = 0xf0, diff --git a/include/drm/bridge/dw_hdmi.h b/include/drm/bridge/dw_hdmi.h index cf528c289857..12144d2f80f4 100644 --- a/include/drm/bridge/dw_hdmi.h +++ b/include/drm/bridge/dw_hdmi.h @@ -156,6 +156,8 @@ void dw_hdmi_setup_rx_sense(struct dw_hdmi *hdmi, bool hpd, bool rx_sense); void dw_hdmi_set_sample_rate(struct dw_hdmi *hdmi, unsigned int rate); void dw_hdmi_set_channel_count(struct dw_hdmi *hdmi, unsigned int cnt); +void dw_hdmi_set_channel_status(struct dw_hdmi *hdmi, + unsigned int sample_width); void dw_hdmi_set_channel_allocation(struct dw_hdmi *hdmi, unsigned int ca); void dw_hdmi_audio_enable(struct dw_hdmi *hdmi); void dw_hdmi_audio_disable(struct dw_hdmi *hdmi); -- 2.23.0.187.g17f5b7556c-goog

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