December 2016 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH v3 0/6] ASoC: sun4i-codec: Add support for A31 Codec
by Chen-Yu Tsai 09 Jan '17

09 Jan '17

Hi everyone, This is v3 of my Allwinner A31 audio codec support series. These are the remaining driver patches not yet merged, with 1 new fix, and the dts file patches. The A31's internal codec is similar (in terms of DMA, interface and control layouts) to the one found in the A10/A13/A20 SoCs. However it has more external inputs and outputs, the mixer controls are now stereo (left/right separated), and it also has some audio processing features (not supported yet). Changes since v2: - Added patch """ASoC: sun4i-codec: Add "Right Mixer" to "Line Out Mono Diff." route""" which fixes line out when all the sources have their left channel muted. - Dropped unused codec_pa_pin from A31 hummingbird dts patch. In the past we always provided a pinmux setting for GPIOs used, to avoid accidentally muxing a pin that was already used for GPIO, or vice versa. The pinctrl subsystem now supports exclusive use of a pin for GPIO, which the sunxi platform wants to migrate to. This drops the requirement of the explicit pinmux setting for GPIOs. - Added Maxime's Acked-by for the two existing patches from v2. After these are merged I plan to submit patches to support the audio codec found in Allwinner's A23 and H3 SoCs. Regards ChenYu Chen-Yu Tsai (6): ASoC: sun4i-codec: Add support for optional reset control to quirks ASoC: sun4i-codec: Add support for A31 ADC capture path ASoC: sun4i-codec: Add "Right Mixer" to "Line Out Mono Diff." route ARM: dts: sun6i: Add audio codec device node ARM: dts: sun6i: hummingbird: Enable internal audio codec ARM: dts: sun6i: sina31s: Enable internal audio codec arch/arm/boot/dts/sun6i-a31-hummingbird.dts | 13 ++++ arch/arm/boot/dts/sun6i-a31.dtsi | 13 ++++ arch/arm/boot/dts/sun6i-a31s-sina31s.dts | 8 +++ sound/soc/sunxi/sun4i-codec.c | 94 ++++++++++++++++++++++++++++- 4 files changed, 127 insertions(+), 1 deletion(-) -- 2.10.2

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[alsa-devel] [PATCH] ASoC: nau8825: Add Class G function options
by Ben Zhang 09 Jan '17

09 Jan '17

Add a DAPM_MUX for userspace to control the Class G function: "Switching": Use +-1.8V or +-0.9V supply, based on signal amplitude. "1.8V": Always use +-1.8V. The default is "Switching" which is the existing behavior without this patch, so no effect on current users. On some devices, the switching may cause a clicking noise at low volume. "1.8V" eliminates the noise at the expense of a few additional mW. Signed-off-by: Ben Zhang <benzh(a)chromium.org> --- sound/soc/codecs/nau8825.c | 24 +++++++++++++++++++++--- 1 file changed, 21 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/nau8825.c b/sound/soc/codecs/nau8825.c index efe3a44..ddd15a0 100644 --- a/sound/soc/codecs/nau8825.c +++ b/sound/soc/codecs/nau8825.c @@ -1062,6 +1062,10 @@ static const char * const nau8825_dac_src[] = { "DACL", "DACR", }; +static const char * const nau8825_class_g_src[] = { + "Switching", "1.8V" +}; + static SOC_ENUM_SINGLE_DECL( nau8825_dacl_enum, NAU8825_REG_DACL_CTRL, NAU8825_DACL_CH_SEL_SFT, nau8825_dac_src); @@ -1070,12 +1074,17 @@ static SOC_ENUM_SINGLE_DECL( nau8825_dacr_enum, NAU8825_REG_DACR_CTRL, NAU8825_DACR_CH_SEL_SFT, nau8825_dac_src); +static SOC_ENUM_SINGLE_DECL( + nau8825_class_g_enum, SND_SOC_NOPM, 0, nau8825_class_g_src); + static const struct snd_kcontrol_new nau8825_dacl_mux = SOC_DAPM_ENUM("DACL Source", nau8825_dacl_enum); static const struct snd_kcontrol_new nau8825_dacr_mux = SOC_DAPM_ENUM("DACR Source", nau8825_dacr_enum); +static const struct snd_kcontrol_new nau8825_class_g_mux = + SOC_DAPM_ENUM("Class G Source", nau8825_class_g_enum); static const struct snd_soc_dapm_widget nau8825_dapm_widgets[] = { SND_SOC_DAPM_AIF_OUT("AIFTX", "Capture", 0, NAU8825_REG_I2S_PCM_CTRL2, @@ -1154,9 +1163,16 @@ static const struct snd_soc_dapm_widget nau8825_dapm_widgets[] = { SND_SOC_DAPM_PGA_S("HP Boost Driver", 9, NAU8825_REG_BOOST, 9, 1, NULL, 0), - /* Class G operation control*/ - SND_SOC_DAPM_PGA_S("Class G", 10, + /* Class G operation control. "Class G" MUX options: + * "Switching": Use +-1.8V or +-0.9V supply, based on signal amplitude. + * "1.8V": Always use +-1.8V. + * The PGA_S widget is used to ensure the Class G control bit is set at + * the correct time w.r.t the other widgets. + */ + SND_SOC_DAPM_PGA_S("Class G Driver", 10, NAU8825_REG_CLASSG_CTRL, 0, 0, NULL, 0), + SND_SOC_DAPM_MUX("Class G", SND_SOC_NOPM, 0, 0, + &nau8825_class_g_mux), SND_SOC_DAPM_OUTPUT("HPOL"), SND_SOC_DAPM_OUTPUT("HPOR"), @@ -1197,7 +1213,9 @@ static const struct snd_soc_dapm_route nau8825_dapm_routes[] = { {"HPOR Pulldown", NULL, "Output DACR"}, {"HP Boost Driver", NULL, "HPOL Pulldown"}, {"HP Boost Driver", NULL, "HPOR Pulldown"}, - {"Class G", NULL, "HP Boost Driver"}, + {"Class G Driver", NULL, "HP Boost Driver"}, + {"Class G", "1.8V", "HP Boost Driver"}, + {"Class G", "Switching", "Class G Driver"}, {"HPOL", NULL, "Class G"}, {"HPOR", NULL, "Class G"}, }; -- 2.8.0.rc3.226.g39d4020

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[alsa-devel] [PATCH] Revert "ASoC: core: replace aux_comp_list to component_dev_list"
by Nikita Yushchenko 09 Jan '17

09 Jan '17

This reverts commit 1a653aa44725668590b36bbe2d7fe4736a69f055 ("ASoC: core: replace aux_comp_list to component_dev_list"). That commit tries to remove card->aux_comp_list, using flagged entries in card->component_dev_list instead. However, components are added to card->component_dev_list in soc_probe_component(), which is called for aux devices by soc_probe_aux_devices(). Before that commit, it traversed card->aux_comp_list and found aux devices added by soc_bind_aux_dev(). After that commit, it traverses card->component_dev_list and finds no aux devices, because soc_probe_component() was not yet called for them. Thus all aux devices are lost and any setup that needs them no longer works. Signed-off-by: Nikita Yushchenko <nikita.yoush(a)cogentembedded.com> --- Fixed lost line in commit message ;) include/sound/soc.h | 4 +++- sound/soc/soc-core.c | 17 +++++------------ 2 files changed, 8 insertions(+), 13 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 2b502f6cc6d0..8172a512632a 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -809,10 +809,10 @@ struct snd_soc_component { unsigned int ignore_pmdown_time:1; /* pmdown_time is ignored at stop */ unsigned int registered_as_component:1; - unsigned int auxiliary:1; /* for auxiliary component of the card */ unsigned int suspended:1; /* is in suspend PM state */ struct list_head list; + struct list_head list_aux; /* for auxiliary component of the card */ struct list_head card_list; struct snd_soc_dai_driver *dai_drv; @@ -1152,6 +1152,7 @@ struct snd_soc_card { */ struct snd_soc_aux_dev *aux_dev; int num_aux_devs; + struct list_head aux_comp_list; const struct snd_kcontrol_new *controls; int num_controls; @@ -1547,6 +1548,7 @@ static inline void snd_soc_initialize_card_lists(struct snd_soc_card *card) INIT_LIST_HEAD(&card->widgets); INIT_LIST_HEAD(&card->paths); INIT_LIST_HEAD(&card->dapm_list); + INIT_LIST_HEAD(&card->aux_comp_list); INIT_LIST_HEAD(&card->component_dev_list); } diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index f1901bb1466e..15657a65e4ca 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1747,8 +1747,7 @@ static int soc_bind_aux_dev(struct snd_soc_card *card, int num) } component->init = aux_dev->init; - component->auxiliary = 1; - + list_add(&component->list_aux, &card->aux_comp_list); return 0; err_defer: @@ -1764,10 +1763,7 @@ static int soc_probe_aux_devices(struct snd_soc_card *card) for (order = SND_SOC_COMP_ORDER_FIRST; order <= SND_SOC_COMP_ORDER_LAST; order++) { - list_for_each_entry(comp, &card->component_dev_list, card_list) { - if (!comp->auxiliary) - continue; - + list_for_each_entry(comp, &card->aux_comp_list, list_aux) { if (comp->driver->probe_order == order) { ret = soc_probe_component(card, comp); if (ret < 0) { @@ -1791,14 +1787,11 @@ static void soc_remove_aux_devices(struct snd_soc_card *card) for (order = SND_SOC_COMP_ORDER_FIRST; order <= SND_SOC_COMP_ORDER_LAST; order++) { list_for_each_entry_safe(comp, _comp, - &card->component_dev_list, card_list) { - - if (!comp->auxiliary) - continue; - + &card->aux_comp_list, list_aux) { if (comp->driver->remove_order == order) { soc_remove_component(comp); - comp->auxiliary = 0; + /* remove it from the card's aux_comp_list */ + list_del(&comp->list_aux); } } } -- 2.1.4

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[alsa-devel] Alsamixer: missing Speakers entry in analog output choice (chipset AV200)
by Germano Massullo 08 Jan '17

08 Jan '17

I have a Asus Xonar Essence ST soundboard (Chip: AV200) on Fedora 23 kernel 4.4.5-300.fc23.x86_64 The audiocard has 3 different outputs - Speakers - Headphones - FP Headphones that have mutual exclusion, so that when you are using one of them, you cannot use the others. Sometimes the Speakers output disappears from alsamixer. By restarting the computer, the system gets back to its normal behaviour. I attach 3 logs taken when the problem was occurring. The problem happened also in past years, but in last a 1-2 years it happened less frequently

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[alsa-devel] [PATCH 1/7] ASoC: wm9705: Remove unused DAI ID defines
by Lars-Peter Clausen 06 Jan '17

06 Jan '17

The DAI ID defines are back from the time when DAIs were referenced by a numerical ID. These days a string is used for matching instead and the defines are unused. The last user of these defines was removed in commit f0fba2ad1b6b ("ASoC: multi-component - ASoC Multi-Component Support"). So remove the defines as well. This also means the wm9705.h file no longer has any content and can be removed. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/codecs/wm9705.c | 2 -- sound/soc/codecs/wm9705.h | 11 ----------- sound/soc/pxa/e740_wm9705.c | 1 - sound/soc/pxa/e750_wm9705.c | 1 - 4 files changed, 15 deletions(-) delete mode 100644 sound/soc/codecs/wm9705.h diff --git a/sound/soc/codecs/wm9705.c b/sound/soc/codecs/wm9705.c index dcdd055..266b6ec 100644 --- a/sound/soc/codecs/wm9705.c +++ b/sound/soc/codecs/wm9705.c @@ -20,8 +20,6 @@ #include <sound/initval.h> #include <sound/soc.h> -#include "wm9705.h" - #define WM9705_VENDOR_ID 0x574d4c05 #define WM9705_VENDOR_ID_MASK 0xffffffff diff --git a/sound/soc/codecs/wm9705.h b/sound/soc/codecs/wm9705.h deleted file mode 100644 index 23ea9ce..0000000 --- a/sound/soc/codecs/wm9705.h +++ /dev/null @@ -1,11 +0,0 @@ -/* - * wm9705.h -- WM9705 Soc Audio driver - */ - -#ifndef _WM9705_H -#define _WM9705_H - -#define WM9705_DAI_AC97_HIFI 0 -#define WM9705_DAI_AC97_AUX 1 - -#endif diff --git a/sound/soc/pxa/e740_wm9705.c b/sound/soc/pxa/e740_wm9705.c index 1de8765..086c37a 100644 --- a/sound/soc/pxa/e740_wm9705.c +++ b/sound/soc/pxa/e740_wm9705.c @@ -22,7 +22,6 @@ #include <asm/mach-types.h> -#include "../codecs/wm9705.h" #include "pxa2xx-ac97.h" diff --git a/sound/soc/pxa/e750_wm9705.c b/sound/soc/pxa/e750_wm9705.c index b7eb7cd..7823278 100644 --- a/sound/soc/pxa/e750_wm9705.c +++ b/sound/soc/pxa/e750_wm9705.c @@ -22,7 +22,6 @@ #include <asm/mach-types.h> -#include "../codecs/wm9705.h" #include "pxa2xx-ac97.h" static int e750_spk_amp_event(struct snd_soc_dapm_widget *w, -- 2.1.4

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[alsa-devel] [PATCH v3 00/13] ASoC: Intel: Skylake: Driver updates
by jeeja.kp＠intel.com 06 Jan '17

06 Jan '17

From: Jeeja KP <jeeja.kp(a)intel.com> This patch series includes driver updates for the handling scenarios during System S0/S3 transitions when audio stream is stopped during system. o move DMA configuration in DSP Gateway Module widget event handler to ensure DMA is configured after resume. o Enable SSP MLCK using Supply widget. This will ensure the MCLK is enabled when the widget is power on. o Check DMA decouple register state before coupling/decoupling the DMA. o In case of HDMI pass-through pipe, reset only the FE Pipe when stream in in XRUN state. o Resume the stream from HW renderer position. o HDMI Codec pin and converter state need to be restored when stream is resumed. Changes in v2: - Addressed Mark's comment on patch 1, split this into 2 as this had additional changes other than configuring the DMA. - Addressed Takashi's comment on the usage of snd_hdac_updatel() Changes in v3: - Addressed Takashi's comment on superfluous shit used in reading the value of PPCTL register and also cleaned the condition for updating the reg value. Jeeja KP (13): ASoC: Intel: Skylake: Update link_index and format in pipe params ASoC: Intel: Skylake: Add helper function to setup host/link dma ASoC: Intel: Skylake: Configure DMA in PRE_PMD handler of Mixer ASoC: Intel: Skylake: Removed unused skl_get_format() ALSA: hda: check stream decoupled register state ASoC: Intel: Skylake: Add set_tristate DAI ops to enable SSP MCLK ASoC: Intel: Skylake: Remove unused SSP BE prepare DAI ops ASoC: Intel: Skylake: Add supply widget as non DSP widget ASoC: Intel: Skylake: Add supply widget in skl_nau_max machine ASoC: Intel: Skylake: Add supply widget in bxt_da_max machine ASoC: Intel: Skylake: Don't reset pass-through pipe in BE prepare ASoC: Intel: Skylake: set the resume point to LPIB ASoC: hdac_hdmi: Enable pin and converter in prepare sound/hda/ext/hdac_ext_stream.c | 15 +- sound/soc/codecs/hdac_hdmi.c | 28 ++-- sound/soc/intel/boards/bxt_da7219_max98357a.c | 34 ++++ sound/soc/intel/boards/skl_nau88l25_max98357a.c | 34 ++++ sound/soc/intel/skylake/skl-pcm.c | 210 +++++++++++++----------- sound/soc/intel/skylake/skl-topology.c | 27 ++- sound/soc/intel/skylake/skl-topology.h | 6 + 7 files changed, 240 insertions(+), 114 deletions(-) -- 2.5.0

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[alsa-devel] [PATCH] ASoC: pcm5102a: Provide a menu selection text
by Jörg Krause 06 Jan '17

06 Jan '17

Provide a menu selection text so that users can enable, disable or mark it as module in menuconfig. Signed-off-by: Jörg Krause <joerg.krause(a)embedded.rocks> --- sound/soc/codecs/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 084c2aa..9ee9591 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -624,7 +624,7 @@ config SND_SOC_PCM3168A_SPI select REGMAP_SPI config SND_SOC_PCM5102A - tristate + tristate "Texas Instruments PCM5102A CODEC" config SND_SOC_PCM512x tristate -- 2.10.2

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Re: [alsa-devel] Stream format conversion in SoC driver
by Clemens Ladisch 05 Jan '17

05 Jan '17

Giedrius Trainavičius wrote: >It uses I2S for sending / receiving audio. However, the master clock on our >sound card is generated by the ADC, which supports only 24 bit data, so >getting other sample formats is not possible. The I2S sample format need not be the same as what the DMA controller reads/writes from/to memory. >A question on 'plug' - it accepts 'nearest' as the argument for format, >rate, what happens in case the slave pcm stream supports the exact same >format as requested by user software? The plug plugin instantiates the actual conversion plugins only as needed; with no conversion, it vanishes entirely. Regards, Clemens

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[alsa-devel] [PATCH v6 0/3] Add platform clock for BayTrail platforms
by Irina Tirdea 05 Jan '17

05 Jan '17

These patches specifically enable the audio MCLK required by Baytrail CR devices. It is the remaining part of a bigger set of patches (already merged in Mark's tree) that enable sound for Baytrail CR devices (especially Asus T100TAF) [1]. They include the clock driver and enabling the clock in the pmc_atom code (along with moving of the non-architectural pmc_atom driver code into drivers/platform/x86). [1] http://mailman.alsa-project.org/pipermail/alsa-devel/2016-August/111704.html Changes from v5: - fix build error reported by kbuild test robot - split the clk driver code from x86 platform changes Changes from v4: - move the pmc_atom driver from arch/x86/platform/atom to drivers/platform/x86 Changes from v3: - replace devm_kzalloc with devm_kcalloc - add x86 architecture maintainers Changes from v2: - move clk platform data structures to a separate include file - store clk_hw pointer for the fixed rate clocks Changes from v1: - register the clk device as child of pmc device - pass iomem pointer from pmc driver to clk driver to avoid using pmc_atom_read()/write() and use readl/writel API instead - use devm_clk_hw_register/clkdev_hw_create instead of clk_register/clkdev_create Irina Tirdea (3): clk: x86: Add Atom PMC platform clocks arch/x86/platform/atom: Move pmc_atom to drivers/platform/x86 platform/x86: Enable Atom PMC platform clocks arch/x86/Kconfig | 4 - arch/x86/platform/atom/Makefile | 1 - drivers/acpi/acpi_lpss.c | 2 +- drivers/clk/x86/Makefile | 3 + drivers/clk/x86/clk-byt-plt.c | 380 +++++++++++++++++++++ drivers/platform/x86/Kconfig | 5 + drivers/platform/x86/Makefile | 1 + .../atom => drivers/platform/x86}/pmc_atom.c | 81 ++++- include/linux/platform_data/x86/clk-byt-plt.h | 31 ++ .../linux/platform_data/x86}/pmc_atom.h | 3 + 10 files changed, 500 insertions(+), 11 deletions(-) create mode 100644 drivers/clk/x86/clk-byt-plt.c rename {arch/x86/platform/atom => drivers/platform/x86}/pmc_atom.c (88%) create mode 100644 include/linux/platform_data/x86/clk-byt-plt.h rename {arch/x86/include/asm => include/linux/platform_data/x86}/pmc_atom.h (98%) -- 1.9.1

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[alsa-devel] [PATCH] drm: sti: allow audio playback on HDMI even if disabled.
by Arnaud Pouliquen 04 Jan '17

04 Jan '17

This fix allows to play audio while HDMI is disconnected. When HDMI is disable, audio configuration is stored and samples are dropped (by HDMI IP). When HDMI is enabled, audio HDMI configuration is applied and samples are outputted on HDMI wire. Signed-off-by: Arnaud Pouliquen <arnaud.pouliquen(a)st.com> --- drivers/gpu/drm/sti/sti_hdmi.c | 205 ++++++++++++++++++++--------------------- 1 file changed, 101 insertions(+), 104 deletions(-) diff --git a/drivers/gpu/drm/sti/sti_hdmi.c b/drivers/gpu/drm/sti/sti_hdmi.c index 376b076..9c0025e 100644 --- a/drivers/gpu/drm/sti/sti_hdmi.c +++ b/drivers/gpu/drm/sti/sti_hdmi.c @@ -788,6 +788,95 @@ static void sti_hdmi_disable(struct drm_bridge *bridge) hdmi->enabled = false; } +/** + * sti_hdmi_audio_get_non_coherent_n() - get N parameter for non-coherent + * clocks. None-coherent clocks means that audio and TMDS clocks have not the + * same source (drifts between clocks). In this case assumption is that CTS is + * automatically calculated by hardware. + * + * @audio_fs: audio frame clock frequency in Hz + * + * Values computed are based on table described in HDMI specification 1.4b + * + * Returns n value. + */ +static int sti_hdmi_audio_get_non_coherent_n(unsigned int audio_fs) +{ + unsigned int n; + + switch (audio_fs) { + case 32000: + n = 4096; + break; + case 44100: + n = 6272; + break; + case 48000: + n = 6144; + break; + case 88200: + n = 6272 * 2; + break; + case 96000: + n = 6144 * 2; + break; + case 176400: + n = 6272 * 4; + break; + case 192000: + n = 6144 * 4; + break; + default: + /* Not pre-defined, recommended value: 128 * fs / 1000 */ + n = (audio_fs * 128) / 1000; + } + + return n; +} + +static int hdmi_audio_configure(struct sti_hdmi *hdmi) +{ + int audio_cfg, n; + struct hdmi_audio_params *params = &hdmi->audio; + struct hdmi_audio_infoframe *info = &params->cea; + + DRM_DEBUG_DRIVER("\n"); + + if (!hdmi->enabled) + return 0; + + /* update N parameter */ + n = sti_hdmi_audio_get_non_coherent_n(params->sample_rate); + + DRM_DEBUG_DRIVER("Audio rate = %d Hz, TMDS clock = %d Hz, n = %d\n", + params->sample_rate, hdmi->mode.clock * 1000, n); + hdmi_write(hdmi, n, HDMI_AUDN); + + /* update HDMI registers according to configuration */ + audio_cfg = HDMI_AUD_CFG_SPDIF_DIV_2 | HDMI_AUD_CFG_DTS_INVALID | + HDMI_AUD_CFG_ONE_BIT_INVALID; + + switch (info->channels) { + case 8: + audio_cfg |= HDMI_AUD_CFG_CH78_VALID; + case 6: + audio_cfg |= HDMI_AUD_CFG_CH56_VALID; + case 4: + audio_cfg |= HDMI_AUD_CFG_CH34_VALID | HDMI_AUD_CFG_8CH; + case 2: + audio_cfg |= HDMI_AUD_CFG_CH12_VALID; + break; + default: + DRM_ERROR("ERROR: Unsupported number of channels (%d)!\n", + info->channels); + return -EINVAL; + } + + hdmi_write(hdmi, audio_cfg, HDMI_AUDIO_CFG); + + return hdmi_audio_infoframe_config(hdmi); +} + static void sti_hdmi_pre_enable(struct drm_bridge *bridge) { struct sti_hdmi *hdmi = bridge->driver_private; @@ -826,9 +915,12 @@ static void sti_hdmi_pre_enable(struct drm_bridge *bridge) if (hdmi_avi_infoframe_config(hdmi)) DRM_ERROR("Unable to configure AVI infoframe\n"); - /* Program AUDIO infoframe */ - if (hdmi_audio_infoframe_config(hdmi)) - DRM_ERROR("Unable to configure AUDIO infoframe\n"); + if (hdmi->audio.enabled) { + if (hdmi_audio_configure(hdmi)) + DRM_ERROR("Unable to configure audio\n"); + } else { + hdmi_audio_infoframe_config(hdmi); + } /* Program VS infoframe */ if (hdmi_vendor_infoframe_config(hdmi)) @@ -1078,97 +1170,6 @@ static struct drm_encoder *sti_hdmi_find_encoder(struct drm_device *dev) return NULL; } -/** - * sti_hdmi_audio_get_non_coherent_n() - get N parameter for non-coherent - * clocks. None-coherent clocks means that audio and TMDS clocks have not the - * same source (drifts between clocks). In this case assumption is that CTS is - * automatically calculated by hardware. - * - * @audio_fs: audio frame clock frequency in Hz - * - * Values computed are based on table described in HDMI specification 1.4b - * - * Returns n value. - */ -static int sti_hdmi_audio_get_non_coherent_n(unsigned int audio_fs) -{ - unsigned int n; - - switch (audio_fs) { - case 32000: - n = 4096; - break; - case 44100: - n = 6272; - break; - case 48000: - n = 6144; - break; - case 88200: - n = 6272 * 2; - break; - case 96000: - n = 6144 * 2; - break; - case 176400: - n = 6272 * 4; - break; - case 192000: - n = 6144 * 4; - break; - default: - /* Not pre-defined, recommended value: 128 * fs / 1000 */ - n = (audio_fs * 128) / 1000; - } - - return n; -} - -static int hdmi_audio_configure(struct sti_hdmi *hdmi, - struct hdmi_audio_params *params) -{ - int audio_cfg, n; - struct hdmi_audio_infoframe *info = &params->cea; - - DRM_DEBUG_DRIVER("\n"); - - if (!hdmi->enabled) - return 0; - - /* update N parameter */ - n = sti_hdmi_audio_get_non_coherent_n(params->sample_rate); - - DRM_DEBUG_DRIVER("Audio rate = %d Hz, TMDS clock = %d Hz, n = %d\n", - params->sample_rate, hdmi->mode.clock * 1000, n); - hdmi_write(hdmi, n, HDMI_AUDN); - - /* update HDMI registers according to configuration */ - audio_cfg = HDMI_AUD_CFG_SPDIF_DIV_2 | HDMI_AUD_CFG_DTS_INVALID | - HDMI_AUD_CFG_ONE_BIT_INVALID; - - switch (info->channels) { - case 8: - audio_cfg |= HDMI_AUD_CFG_CH78_VALID; - case 6: - audio_cfg |= HDMI_AUD_CFG_CH56_VALID; - case 4: - audio_cfg |= HDMI_AUD_CFG_CH34_VALID | HDMI_AUD_CFG_8CH; - case 2: - audio_cfg |= HDMI_AUD_CFG_CH12_VALID; - break; - default: - DRM_ERROR("ERROR: Unsupported number of channels (%d)!\n", - info->channels); - return -EINVAL; - } - - hdmi_write(hdmi, audio_cfg, HDMI_AUDIO_CFG); - - hdmi->audio = *params; - - return hdmi_audio_infoframe_config(hdmi); -} - static void hdmi_audio_shutdown(struct device *dev, void *data) { struct sti_hdmi *hdmi = dev_get_drvdata(dev); @@ -1192,17 +1193,9 @@ static int hdmi_audio_hw_params(struct device *dev, { struct sti_hdmi *hdmi = dev_get_drvdata(dev); int ret; - struct hdmi_audio_params audio = { - .sample_width = params->sample_width, - .sample_rate = params->sample_rate, - .cea = params->cea, - }; DRM_DEBUG_DRIVER("\n"); - if (!hdmi->enabled) - return 0; - if ((daifmt->fmt != HDMI_I2S) || daifmt->bit_clk_inv || daifmt->frame_clk_inv || daifmt->bit_clk_master || daifmt->frame_clk_master) { @@ -1213,9 +1206,13 @@ static int hdmi_audio_hw_params(struct device *dev, return -EINVAL; } - audio.enabled = true; + hdmi->audio.sample_width = params->sample_width; + hdmi->audio.sample_rate = params->sample_rate; + hdmi->audio.cea = params->cea; + + hdmi->audio.enabled = true; - ret = hdmi_audio_configure(hdmi, &audio); + ret = hdmi_audio_configure(hdmi); if (ret < 0) return ret; -- 1.9.1

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