September 2017 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] Asoc: rockchip: Init dapm routes dynamically
by Jeffy Chen 18 Sep '17

18 Sep '17

Currently we are using a fixed list of dapm routes. Init dapm routes dynamically when parsing dailinks, since we are supporting optional codecs. Signed-off-by: Jeffy Chen <jeffy.chen(a)rock-chips.com> --- sound/soc/rockchip/rk3399_gru_sound.c | 88 +++++++++++++++++++++++++++++------ 1 file changed, 74 insertions(+), 14 deletions(-) diff --git a/sound/soc/rockchip/rk3399_gru_sound.c b/sound/soc/rockchip/rk3399_gru_sound.c index 0513fe480353..c537781f8054 100644 --- a/sound/soc/rockchip/rk3399_gru_sound.c +++ b/sound/soc/rockchip/rk3399_gru_sound.c @@ -47,18 +47,7 @@ static const struct snd_soc_dapm_widget rockchip_dapm_widgets[] = { SND_SOC_DAPM_SPK("Speakers", NULL), SND_SOC_DAPM_MIC("Headset Mic", NULL), SND_SOC_DAPM_MIC("Int Mic", NULL), -}; - -static const struct snd_soc_dapm_route rockchip_dapm_routes[] = { - /* Input Lines */ - {"MIC", NULL, "Headset Mic"}, - {"DMIC1L", NULL, "Int Mic"}, - {"DMIC1R", NULL, "Int Mic"}, - - /* Output Lines */ - {"Headphones", NULL, "HPL"}, - {"Headphones", NULL, "HPR"}, - {"Speakers", NULL, "Speaker"}, + SND_SOC_DAPM_LINE("HDMI", NULL), }; static const struct snd_kcontrol_new rockchip_controls[] = { @@ -66,6 +55,7 @@ static const struct snd_kcontrol_new rockchip_controls[] = { SOC_DAPM_PIN_SWITCH("Speakers"), SOC_DAPM_PIN_SWITCH("Headset Mic"), SOC_DAPM_PIN_SWITCH("Int Mic"), + SOC_DAPM_PIN_SWITCH("HDMI"), }; static int rockchip_sound_max98357a_hw_params(struct snd_pcm_substream *substream, @@ -314,8 +304,6 @@ static struct snd_soc_card rockchip_sound_card = { .owner = THIS_MODULE, .dapm_widgets = rockchip_dapm_widgets, .num_dapm_widgets = ARRAY_SIZE(rockchip_dapm_widgets), - .dapm_routes = rockchip_dapm_routes, - .num_dapm_routes = ARRAY_SIZE(rockchip_dapm_routes), .controls = rockchip_controls, .num_controls = ARRAY_SIZE(rockchip_controls), }; @@ -391,6 +379,65 @@ static const struct snd_soc_dai_link rockchip_dais[] = { }, }; +static const struct snd_soc_dapm_route rockchip_sound_cdndp_routes[] = { + /* Output */ + {"HDMI", NULL, "TX"}, +}; + +static const struct snd_soc_dapm_route rockchip_sound_da7219_routes[] = { + /* Output */ + {"Headphones", NULL, "HPL"}, + {"Headphones", NULL, "HPR"}, + + /* Input */ + {"MIC", NULL, "Headset Mic"}, +}; + +static const struct snd_soc_dapm_route rockchip_sound_dmic_routes[] = { + /* Input */ + {"Dmic", NULL, "Int Mic"}, +}; + +static const struct snd_soc_dapm_route rockchip_sound_max98357a_routes[] = { + /* Output */ + {"Speakers", NULL, "Speaker"}, +}; + +static const struct snd_soc_dapm_route rockchip_sound_rt5514_routes[] = { + /* Input */ + {"DMIC1L", NULL, "Int Mic"}, + {"DMIC1R", NULL, "Int Mic"}, +}; + +struct rockchip_sound_route { + const struct snd_soc_dapm_route *routes; + int num_routes; +}; + +static const struct rockchip_sound_route rockchip_routes[] = { + [DAILINK_CDNDP] = { + .routes = rockchip_sound_cdndp_routes, + .num_routes = ARRAY_SIZE(rockchip_sound_cdndp_routes), + }, + [DAILINK_DA7219] = { + .routes = rockchip_sound_da7219_routes, + .num_routes = ARRAY_SIZE(rockchip_sound_da7219_routes), + }, + [DAILINK_DMIC] = { + .routes = rockchip_sound_dmic_routes, + .num_routes = ARRAY_SIZE(rockchip_sound_dmic_routes), + }, + [DAILINK_MAX98357A] = { + .routes = rockchip_sound_max98357a_routes, + .num_routes = ARRAY_SIZE(rockchip_sound_max98357a_routes), + }, + [DAILINK_RT5514] = { + .routes = rockchip_sound_rt5514_routes, + .num_routes = ARRAY_SIZE(rockchip_sound_rt5514_routes), + }, + [DAILINK_RT5514_DSP] = {}, +}; + static int rockchip_sound_codec_node_match(struct device_node *np_codec) { int i; @@ -408,6 +455,7 @@ static int rockchip_sound_of_parse_dais(struct device *dev, struct device_node *np_cpu, *np_cpu0, *np_cpu1; struct device_node *np_codec; struct snd_soc_dai_link *dai; + struct snd_soc_dapm_route *routes; int i, index; card->dai_link = devm_kzalloc(dev, sizeof(rockchip_dais), @@ -415,9 +463,16 @@ static int rockchip_sound_of_parse_dais(struct device *dev, if (!card->dai_link) return -ENOMEM; + routes = devm_kzalloc(dev, sizeof(rockchip_routes), + GFP_KERNEL); + if (!routes) + return -ENOMEM; + card->dapm_routes = routes; + np_cpu0 = of_parse_phandle(dev->of_node, "rockchip,cpu", 0); np_cpu1 = of_parse_phandle(dev->of_node, "rockchip,cpu", 1); + card->num_dapm_routes = 0; card->num_links = 0; for (i = 0; i < ARRAY_SIZE(rockchip_dais); i++) { np_codec = of_parse_phandle(dev->of_node, @@ -445,6 +500,11 @@ static int rockchip_sound_of_parse_dais(struct device *dev, dai->codec_of_node = np_codec; dai->platform_of_node = np_cpu; dai->cpu_of_node = np_cpu; + + memcpy(routes + card->num_dapm_routes, + rockchip_routes[index].routes, + rockchip_routes[index].num_routes * sizeof(*routes)); + card->num_dapm_routes += rockchip_routes[index].num_routes; } return 0; -- 2.11.0

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[alsa-devel] [PATCH] sound: tegra20-spdif: remove "default m"
by Michał Mirosław 17 Sep '17

17 Sep '17

Default != n for drivers is frowned upon. This is the only one among tegra sound drivers that was different. Signed-off-by: Michał Mirosław <mirq-linux(a)rere.qmqm.pl> --- sound/soc/tegra/Kconfig | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/tegra/Kconfig b/sound/soc/tegra/Kconfig index 6875fc39a575..6c893c677110 100644 --- a/sound/soc/tegra/Kconfig +++ b/sound/soc/tegra/Kconfig @@ -38,7 +38,6 @@ config SND_SOC_TEGRA20_I2S config SND_SOC_TEGRA20_SPDIF tristate "Tegra20 SPDIF interface" depends on SND_SOC_TEGRA - default m help Say Y or M if you want to add support for the Tegra20 SPDIF interface. You will also need to select the individual machine drivers to support -- 2.11.0

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[alsa-devel] DICE Stereo-192-DSD-DAC issues since 2016-02-08 / kernel >= ~4.6
by Melvin Vermeeren 16 Sep '17

16 Sep '17

Hi all, Sometime in 2016, I believe with kernel ~4.6, my DAC started having weird behaviour. I found some mails about heavy firewire/DICE dev at the time so I sticked with 4.4 LTS for a while. I decided to do some proper research now and found out some details which hopefully will help solving the problem. The hardware, problem is with DAC only: https://www.siig.com/it-products/firewire/firewire-400/pcie/dp-firewire-2-p… https://mytekdigital.com/hifi/products/stereo-192dsd-dac/ https://us.focusrite.com/firewire-audio-interfaces/saffire-pro-24 The Saffire works perfectly for all modules I tried. I believe my DAC's firmware may have some quirks with control/notification messages. Behaviour comparison -------------------- Old modules (2eb65d67afbf9364b525b657f1475d1a2cbc27de): * Playback is instant, but DAC very rarely misses the lock. Need to restart the stream (stop, play MPD or restart jack) in that case. DAC's display still displays the samplerate change correctly, so probably firmware inconsistency. * Songs with different sample rate (e.g. 44100 -> 96000) switch instantly, unless DAC misses lock like above. DAC's display updates sample rate in 1-5 sec or so. * Sound itself is perfect, no cracks, pops or other issues. * Stop ALSA stream and DAC will indicate "No Lock", but procfs says locked. * dmesg is silent about firewire/dice/dac. New modules (4.1.1.12 and 4.13): * Takes ~2 sec before playback starts, but always works. * When MPD changes music track to track with different format it will always hang ~2 sec and fail to change format anyway. * Sound has very consistent popping in it. Very noticeable on higher frequencies, some timing issue? Besides that no other issues. * Power cycle the DAC and it will be "unlocked". After that the first stream opened will perma-lock the format. Changing to other frequency will not work. * Stop ALSA stream and DAC will indicate "No Lock", but procfs says locked. * dmesg shows only: "snd_dice fw1.0: Invalid CIP header for AMDTP: 00000000:00000000" probably when it's trying to change frequency. No other messages. I attached /proc/asound/DAC/dice output for old and new when idle, 44100 and 96000. global/notification and sample/lock change but that's basically it. Module details -------------- To test modules on my current kernel, which is 4.11.12-1-rt13-rt, I git checkout desired version of /sound/firewire and make those modules. After that I insmod snd-firewire-lib and snd-dice. The "old" modules refer to: 2eb65d67afbf9364b525b657f1475d1a2cbc27de ALSA: dice: expand timeout to wait for Dice notification These work like kernel ~4.4 did on the new kernel, just needed to include <linux/sched/signal.h> to get it to compile. New modules are as shipped as in kernel 4.11.12, might contain realtime patches. Also tried with mainline kernel 4.13's modules with same results. The commit that breaks it for my DAC is the one directly after "old" when it comes to the DICE module: 0d5ee195b1091e6ebcc121c091ce8646e95a48b6 ALSA: dice: limit to current sampling transfer frequency I don't notice any difference between the commit that "breaks" and current stable (4.13) when it comes to behaviour. Suspected cause --------------- I haven't read too much of the source code and all the development over time, so this is just a guess. Before the module would probably just send the audio stream and hope it works whereas the newer versions try to do it properly, perhaps violating some standard. The fact that according to procfs DAC's status is locked even when not playing might be why the current module will refuse to play at any different frequency. At least, that is what I think the description of 0d5ee195b1091e6ebcc121c091ce8646e95a48b6 is describing. So probably some quirk in my DAC's firmware is causing this. I remember that the Windows drivers always locked even without playback. Changing frequency from the driver software caused DAC display to update instantly with the lock changing shortly after. Though I believe Focusrite's software did the same thing for the mic interface. lspci -vv of my firewire card ----------------------------- Doubt this is related, but just in case. 03:00.0 FireWire (IEEE 1394): Texas Instruments XIO2200A IEEE-1394a-2000 Controller (PHY/Link) (rev 01) (prog-if 10 [OHCI]) Subsystem: AFAVLAB Technology Inc Device 0000 Control: I/O- Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- DisINTx- Status: Cap+ 66MHz+ UDF- FastB2B- ParErr- DEVSEL=medium >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 32 (500ns min, 1000ns max), Cache Line Size: 64 bytes Interrupt: pin A routed to IRQ 24 Region 0: Memory at fb704000 (32-bit, non-prefetchable) [size=2K] Region 1: Memory at fb700000 (32-bit, non-prefetchable) [size=16K] Capabilities: [44] Power Management version 2 Flags: PMEClk- DSI- D1+ D2+ AuxCurrent=0mA PME(D0+,D1+,D2+,D3hot+,D3cold-) Status: D0 NoSoftRst- PME-Enable- DSel=0 DScale=0 PME+ Kernel driver in use: firewire_ohci Kernel modules: firewire_ohci P.S. Seems my DAC has dedicated stereo channels for DSD128 (probably also for DSD64) besides the PCM stereo channels. Any idea if/how that may work? Melvin Vermeeren.

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[alsa-devel] [PATCH] ASoC: rt5514: expose Hotword Model control
by Hsin-Yu Chao 15 Sep '17

15 Sep '17

This change exposes mixer control 'Hotword Model' for switching hotword model in runtime. This new function requires updated firmware to load hotword model at address 0x4ff80000. Signed-off-by: Hsin-Yu Chao <hychao(a)chromium.org> --- sound/soc/codecs/rt5514.c | 52 +++++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/rt5514.h | 3 +++ 2 files changed, 55 insertions(+) diff --git a/sound/soc/codecs/rt5514.c b/sound/soc/codecs/rt5514.c index a2722177470e..7482df8ff737 100644 --- a/sound/soc/codecs/rt5514.c +++ b/sound/soc/codecs/rt5514.c @@ -299,6 +299,7 @@ static int rt5514_dsp_voice_wake_up_put(struct snd_kcontrol *kcontrol, struct rt5514_priv *rt5514 = snd_soc_component_get_drvdata(component); struct snd_soc_codec *codec = rt5514->codec; const struct firmware *fw = NULL; + int ret = 0; if (ucontrol->value.integer.value[0] == rt5514->dsp_enabled) return 0; @@ -335,6 +336,27 @@ static int rt5514_dsp_voice_wake_up_put(struct snd_kcontrol *kcontrol, fw = NULL; } + if (rt5514->model_buf && rt5514->model_len) { + ret = rt5514_spi_burst_write(0x4ff80000, + rt5514->model_buf, + ((rt5514->model_len / 8) + 1) * 8); + if (ret) { + dev_err(codec->dev, + "Model load failed %d\n", ret); + return ret; + } + } else { + request_firmware(&fw, RT5514_FIRMWARE3, + codec->dev); + if (fw) { + rt5514_spi_burst_write(0x4ff80000, + fw->data, + ((fw->size/8)+1)*8); + release_firmware(fw); + fw = NULL; + } + } + /* DSP run */ regmap_write(rt5514->i2c_regmap, 0x18002f00, 0x00055148); @@ -349,6 +371,34 @@ static int rt5514_dsp_voice_wake_up_put(struct snd_kcontrol *kcontrol, return 0; } +static int rt5514_hotword_model_put(struct snd_kcontrol *kcontrol, + const unsigned int __user *bytes, unsigned int size) +{ + struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); + struct rt5514_priv *rt5514 = snd_soc_component_get_drvdata(component); + struct snd_soc_codec *codec = rt5514->codec; + int ret = 0; + + if (rt5514->model_buf || rt5514->model_len < size) { + if (rt5514->model_buf) + devm_kfree(codec->dev, rt5514->model_buf); + rt5514->model_buf = devm_kmalloc(codec->dev, size, GFP_KERNEL); + if (!rt5514->model_buf) { + ret = -ENOMEM; + goto done; + } + } + + /* Skips the TLV header. */ + bytes += 2; + + if (copy_from_user(rt5514->model_buf, bytes, size)) + ret = -EFAULT; +done: + rt5514->model_len = (ret ? 0 : size); + return ret; +} + static const struct snd_kcontrol_new rt5514_snd_controls[] = { SOC_DOUBLE_TLV("MIC Boost Volume", RT5514_ANA_CTRL_MICBST, RT5514_SEL_BSTL_SFT, RT5514_SEL_BSTR_SFT, 8, 0, bst_tlv), @@ -360,6 +410,8 @@ static const struct snd_kcontrol_new rt5514_snd_controls[] = { adc_vol_tlv), SOC_SINGLE_EXT("DSP Voice Wake Up", SND_SOC_NOPM, 0, 1, 0, rt5514_dsp_voice_wake_up_get, rt5514_dsp_voice_wake_up_put), + SND_SOC_BYTES_TLV("Hotword Model", 0x8504, + NULL, rt5514_hotword_model_put), }; /* ADC Mixer*/ diff --git a/sound/soc/codecs/rt5514.h b/sound/soc/codecs/rt5514.h index 02bc212a86d9..07475a9918d2 100644 --- a/sound/soc/codecs/rt5514.h +++ b/sound/soc/codecs/rt5514.h @@ -236,6 +236,7 @@ #define RT5514_FIRMWARE1 "rt5514_dsp_fw1.bin" #define RT5514_FIRMWARE2 "rt5514_dsp_fw2.bin" +#define RT5514_FIRMWARE3 "rt5514_dsp_fw3.bin" /* System Clock Source */ enum { @@ -262,6 +263,8 @@ struct rt5514_priv { int pll_in; int pll_out; int dsp_enabled; + u8 *model_buf; + unsigned int model_len; }; #endif /* __RT5514_H__ */ -- 2.12.2

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[alsa-devel] Applied "ASoC: rt5514: Revert Hotword Model control" to the asoc tree
by Mark Brown 15 Sep '17

15 Sep '17

The patch ASoC: rt5514: Revert Hotword Model control has been applied to the asoc tree at git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 37128514d708457fc060a2309aaa78b057a00043 Mon Sep 17 00:00:00 2001 From: Hsin-Yu Chao <hychao(a)chromium.org> Date: Fri, 15 Sep 2017 12:32:00 +0800 Subject: [PATCH] ASoC: rt5514: Revert Hotword Model control This reverts commit eb33869c7206 ("ASoC: rt5514: Guard Hotword Model bytes loading") and commit d18420b0a0b8 ("ASoC: rt5514: expose Hotword Model control") It is discouraged to use SND_SOC_BYTES_TLV to load arbitrary bytes from userspace to driver. Removing the 'Hotword Model' control until we have a good way to verify the content of hotword model blobs. Signed-off-by: Hsin-Yu Chao <hychao(a)chromium.org> Reviewed-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/rt5514.c | 63 ----------------------------------------------- sound/soc/codecs/rt5514.h | 3 --- 2 files changed, 66 deletions(-) diff --git a/sound/soc/codecs/rt5514.c b/sound/soc/codecs/rt5514.c index 0945d212b8dc..d7956ababd11 100644 --- a/sound/soc/codecs/rt5514.c +++ b/sound/soc/codecs/rt5514.c @@ -338,39 +338,6 @@ static int rt5514_dsp_voice_wake_up_put(struct snd_kcontrol *kcontrol, fw = NULL; } - if (rt5514->model_buf && rt5514->model_len) { -#if IS_ENABLED(CONFIG_SND_SOC_RT5514_SPI) - int ret; - - ret = rt5514_spi_burst_write(0x4ff80000, - rt5514->model_buf, - ((rt5514->model_len / 8) + 1) * 8); - if (ret) { - dev_err(codec->dev, - "Model load failed %d\n", ret); - return ret; - } -#else - dev_err(codec->dev, - "No SPI driver for loading firmware\n"); -#endif - } else { - request_firmware(&fw, RT5514_FIRMWARE3, - codec->dev); - if (fw) { -#if IS_ENABLED(CONFIG_SND_SOC_RT5514_SPI) - rt5514_spi_burst_write(0x4ff80000, - fw->data, - ((fw->size/8)+1)*8); -#else - dev_err(codec->dev, - "No SPI driver to load fw\n"); -#endif - release_firmware(fw); - fw = NULL; - } - } - /* DSP run */ regmap_write(rt5514->i2c_regmap, 0x18002f00, 0x00055148); @@ -385,34 +352,6 @@ static int rt5514_dsp_voice_wake_up_put(struct snd_kcontrol *kcontrol, return 0; } -static int rt5514_hotword_model_put(struct snd_kcontrol *kcontrol, - const unsigned int __user *bytes, unsigned int size) -{ - struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); - struct rt5514_priv *rt5514 = snd_soc_component_get_drvdata(component); - struct snd_soc_codec *codec = rt5514->codec; - int ret = 0; - - if (rt5514->model_buf || rt5514->model_len < size) { - if (rt5514->model_buf) - devm_kfree(codec->dev, rt5514->model_buf); - rt5514->model_buf = devm_kmalloc(codec->dev, size, GFP_KERNEL); - if (!rt5514->model_buf) { - ret = -ENOMEM; - goto done; - } - } - - /* Skips the TLV header. */ - bytes += 2; - - if (copy_from_user(rt5514->model_buf, bytes, size)) - ret = -EFAULT; -done: - rt5514->model_len = (ret ? 0 : size); - return ret; -} - static const struct snd_kcontrol_new rt5514_snd_controls[] = { SOC_DOUBLE_TLV("MIC Boost Volume", RT5514_ANA_CTRL_MICBST, RT5514_SEL_BSTL_SFT, RT5514_SEL_BSTR_SFT, 8, 0, bst_tlv), @@ -424,8 +363,6 @@ static const struct snd_kcontrol_new rt5514_snd_controls[] = { adc_vol_tlv), SOC_SINGLE_EXT("DSP Voice Wake Up", SND_SOC_NOPM, 0, 1, 0, rt5514_dsp_voice_wake_up_get, rt5514_dsp_voice_wake_up_put), - SND_SOC_BYTES_TLV("Hotword Model", 0x8504, - NULL, rt5514_hotword_model_put), }; /* ADC Mixer*/ diff --git a/sound/soc/codecs/rt5514.h b/sound/soc/codecs/rt5514.h index 803311cb7e2a..2dc40e6d8b3f 100644 --- a/sound/soc/codecs/rt5514.h +++ b/sound/soc/codecs/rt5514.h @@ -255,7 +255,6 @@ #define RT5514_FIRMWARE1 "rt5514_dsp_fw1.bin" #define RT5514_FIRMWARE2 "rt5514_dsp_fw2.bin" -#define RT5514_FIRMWARE3 "rt5514_dsp_fw3.bin" /* System Clock Source */ enum { @@ -282,8 +281,6 @@ struct rt5514_priv { int pll_in; int pll_out; int dsp_enabled; - u8 *model_buf; - unsigned int model_len; }; #endif /* __RT5514_H__ */ -- 2.13.2

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[alsa-devel] [PATCH] ASoC: rt5514: Revert Hotword Model control
by Hsin-Yu Chao 15 Sep '17

15 Sep '17

This reverts commit eb33869c7206 ("ASoC: rt5514: Guard Hotword Model bytes loading") and commit d18420b0a0b8 ("ASoC: rt5514: expose Hotword Model control") It is discouraged to use SND_SOC_BYTES_TLV to load arbitrary bytes from userspace to driver. Removing the 'Hotword Model' control until we have a good way to verify the content of hotword model blobs. Signed-off-by: Hsin-Yu Chao <hychao(a)chromium.org> --- sound/soc/codecs/rt5514.c | 63 ----------------------------------------------- sound/soc/codecs/rt5514.h | 3 --- 2 files changed, 66 deletions(-) diff --git a/sound/soc/codecs/rt5514.c b/sound/soc/codecs/rt5514.c index 250581082e9d..a2722177470e 100644 --- a/sound/soc/codecs/rt5514.c +++ b/sound/soc/codecs/rt5514.c @@ -335,39 +335,6 @@ static int rt5514_dsp_voice_wake_up_put(struct snd_kcontrol *kcontrol, fw = NULL; } - if (rt5514->model_buf && rt5514->model_len) { -#if IS_ENABLED(CONFIG_SND_SOC_RT5514_SPI) - int ret; - - ret = rt5514_spi_burst_write(0x4ff80000, - rt5514->model_buf, - ((rt5514->model_len / 8) + 1) * 8); - if (ret) { - dev_err(codec->dev, - "Model load failed %d\n", ret); - return ret; - } -#else - dev_err(codec->dev, - "No SPI driver for loading firmware\n"); -#endif - } else { - request_firmware(&fw, RT5514_FIRMWARE3, - codec->dev); - if (fw) { -#if IS_ENABLED(CONFIG_SND_SOC_RT5514_SPI) - rt5514_spi_burst_write(0x4ff80000, - fw->data, - ((fw->size/8)+1)*8); -#else - dev_err(codec->dev, - "No SPI driver to load fw\n"); -#endif - release_firmware(fw); - fw = NULL; - } - } - /* DSP run */ regmap_write(rt5514->i2c_regmap, 0x18002f00, 0x00055148); @@ -382,34 +349,6 @@ static int rt5514_dsp_voice_wake_up_put(struct snd_kcontrol *kcontrol, return 0; } -static int rt5514_hotword_model_put(struct snd_kcontrol *kcontrol, - const unsigned int __user *bytes, unsigned int size) -{ - struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); - struct rt5514_priv *rt5514 = snd_soc_component_get_drvdata(component); - struct snd_soc_codec *codec = rt5514->codec; - int ret = 0; - - if (rt5514->model_buf || rt5514->model_len < size) { - if (rt5514->model_buf) - devm_kfree(codec->dev, rt5514->model_buf); - rt5514->model_buf = devm_kmalloc(codec->dev, size, GFP_KERNEL); - if (!rt5514->model_buf) { - ret = -ENOMEM; - goto done; - } - } - - /* Skips the TLV header. */ - bytes += 2; - - if (copy_from_user(rt5514->model_buf, bytes, size)) - ret = -EFAULT; -done: - rt5514->model_len = (ret ? 0 : size); - return ret; -} - static const struct snd_kcontrol_new rt5514_snd_controls[] = { SOC_DOUBLE_TLV("MIC Boost Volume", RT5514_ANA_CTRL_MICBST, RT5514_SEL_BSTL_SFT, RT5514_SEL_BSTR_SFT, 8, 0, bst_tlv), @@ -421,8 +360,6 @@ static const struct snd_kcontrol_new rt5514_snd_controls[] = { adc_vol_tlv), SOC_SINGLE_EXT("DSP Voice Wake Up", SND_SOC_NOPM, 0, 1, 0, rt5514_dsp_voice_wake_up_get, rt5514_dsp_voice_wake_up_put), - SND_SOC_BYTES_TLV("Hotword Model", 0x8504, - NULL, rt5514_hotword_model_put), }; /* ADC Mixer*/ diff --git a/sound/soc/codecs/rt5514.h b/sound/soc/codecs/rt5514.h index 07475a9918d2..02bc212a86d9 100644 --- a/sound/soc/codecs/rt5514.h +++ b/sound/soc/codecs/rt5514.h @@ -236,7 +236,6 @@ #define RT5514_FIRMWARE1 "rt5514_dsp_fw1.bin" #define RT5514_FIRMWARE2 "rt5514_dsp_fw2.bin" -#define RT5514_FIRMWARE3 "rt5514_dsp_fw3.bin" /* System Clock Source */ enum { @@ -263,8 +262,6 @@ struct rt5514_priv { int pll_in; int pll_out; int dsp_enabled; - u8 *model_buf; - unsigned int model_len; }; #endif /* __RT5514_H__ */ -- 2.12.2

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Re: [alsa-devel] [PATCH v2] ASoC: rt5663: Add the function of impedance sensing
by Mark Brown 15 Sep '17

15 Sep '17

On Fri, Sep 15, 2017 at 02:20:09AM +0000, Oder Chiou wrote: > Sorry for my poor explanation. That's OK. > The impedance table should like the following setting in the DTS. > < 0 300 7 0xffd160 0xffd1c0 0xff8a10 0xff8ab0 > 301 65535 4 0xffe470 0xffe470 0xffb8e0 0xffb8e0> > The first and second column are defined for the impedance range. If the > detected impedance value is in the range, then the volume value of the third > column will be set to codec. In our codec design, each volume value should > compensate different DC offset to avoid the pop sound, and it is also > different between headphone and headset. In the example, the > "realtek,impedance_sensing_num" is 2. It means that there are 2 ranges of > impedance in the impedance sensing function, and it also can be defined the > device whether can support this function or not. This stuff looks fairly good, can you put it in the binding document please?

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[alsa-devel] Need help in getting ALSA information
by Nagendra Kumar 15 Sep '17

15 Sep '17

Hi, My project needs ALSA device driver integration. Can you please help in in getting:1. Demo application(may be sound recorder and player) using ALSA device driver. 2. in knowing function call traces about packets coming from sound card and being delivering to demo(during recording).3. in knowing function call traces about packets going from application to ALSA till sound card (during playing sound). I just want to understand the function/call flow from/to application to/from sound card. My project have requirement to change the packet flow and add some code and test in the packet flow path like delay the sound packet, modify the sound packet from app till sound card and vice versa. Please kindly provide some pointers. Thanks-Nagendra

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[alsa-devel] [PATCH] ASoC: add null_snd_soc_ops and reduce NULL ops check
by Kuninori Morimoto 15 Sep '17

15 Sep '17

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Double NULL pointer check for ops and ops->func is difficult to read and might be forget to check it if new func was add. This patch adds new null_snd_soc_ops and use it if rtd->dai_link didn't have it to avoid NULL ops, and reduces ops NULL check. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/soc-core.c | 5 +++++ sound/soc/soc-pcm.c | 16 ++++++++-------- 2 files changed, 13 insertions(+), 8 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index f8696dc..100be5c 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -614,6 +614,8 @@ struct snd_pcm_substream *snd_soc_get_dai_substream(struct snd_soc_card *card, } EXPORT_SYMBOL_GPL(snd_soc_get_dai_substream); +static const struct snd_soc_ops null_snd_soc_ops; + static struct snd_soc_pcm_runtime *soc_new_pcm_runtime( struct snd_soc_card *card, struct snd_soc_dai_link *dai_link) { @@ -626,6 +628,9 @@ static struct snd_soc_pcm_runtime *soc_new_pcm_runtime( INIT_LIST_HEAD(&rtd->component_list); rtd->card = card; rtd->dai_link = dai_link; + if (!rtd->dai_link->ops) + rtd->dai_link->ops = &null_snd_soc_ops; + rtd->codec_dais = kzalloc(sizeof(struct snd_soc_dai *) * dai_link->num_codecs, GFP_KERNEL); diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index e906b7c..0d6f913 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -543,7 +543,7 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) codec_dai->rx_mask = 0; } - if (rtd->dai_link->ops && rtd->dai_link->ops->startup) { + if (rtd->dai_link->ops->startup) { ret = rtd->dai_link->ops->startup(substream); if (ret < 0) { pr_err("ASoC: %s startup failed: %d\n", @@ -617,7 +617,7 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) return 0; config_err: - if (rtd->dai_link->ops && rtd->dai_link->ops->shutdown) + if (rtd->dai_link->ops->shutdown) rtd->dai_link->ops->shutdown(substream); machine_err: @@ -739,7 +739,7 @@ static int soc_pcm_close(struct snd_pcm_substream *substream) codec_dai->driver->ops->shutdown(substream, codec_dai); } - if (rtd->dai_link->ops && rtd->dai_link->ops->shutdown) + if (rtd->dai_link->ops->shutdown) rtd->dai_link->ops->shutdown(substream); if (platform && platform->driver->ops && platform->driver->ops->close) @@ -814,7 +814,7 @@ static int soc_pcm_prepare(struct snd_pcm_substream *substream) mutex_lock_nested(&rtd->pcm_mutex, rtd->pcm_subclass); - if (rtd->dai_link->ops && rtd->dai_link->ops->prepare) { + if (rtd->dai_link->ops->prepare) { ret = rtd->dai_link->ops->prepare(substream); if (ret < 0) { dev_err(rtd->card->dev, "ASoC: machine prepare error:" @@ -944,7 +944,7 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream, if (ret) goto out; - if (rtd->dai_link->ops && rtd->dai_link->ops->hw_params) { + if (rtd->dai_link->ops->hw_params) { ret = rtd->dai_link->ops->hw_params(substream, params); if (ret < 0) { dev_err(rtd->card->dev, "ASoC: machine hw_params" @@ -1074,7 +1074,7 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream, codec_dai->rate = 0; } - if (rtd->dai_link->ops && rtd->dai_link->ops->hw_free) + if (rtd->dai_link->ops->hw_free) rtd->dai_link->ops->hw_free(substream); mutex_unlock(&rtd->pcm_mutex); @@ -1122,7 +1122,7 @@ static int soc_pcm_hw_free(struct snd_pcm_substream *substream) } /* free any machine hw params */ - if (rtd->dai_link->ops && rtd->dai_link->ops->hw_free) + if (rtd->dai_link->ops->hw_free) rtd->dai_link->ops->hw_free(substream); /* free any DMA resources */ @@ -1206,7 +1206,7 @@ static int soc_pcm_trigger(struct snd_pcm_substream *substream, int cmd) return ret; } - if (rtd->dai_link->ops && rtd->dai_link->ops->trigger) { + if (rtd->dai_link->ops->trigger) { ret = rtd->dai_link->ops->trigger(substream, cmd); if (ret < 0) return ret; -- 1.9.1

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[alsa-devel] [PATCH] ASoC: rt5645: Make a few struct const
by Jean Delvare 15 Sep '17

15 Sep '17

These dmi_system_id structures and associated platform data are never modified so they can be marked const. Signed-off-by: Jean Delvare <jdelvare(a)suse.de> Cc: Bard Liao <bardliao(a)realtek.com> Cc: Oder Chiou <oder_chiou(a)realtek.com> Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> -- sound/soc/codecs/rt5645.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) --- linux-4.14-rc0.orig/sound/soc/codecs/rt5645.c 2017-09-14 11:08:35.328003671 +0200 +++ linux-4.14-rc0/sound/soc/codecs/rt5645.c 2017-09-14 11:38:34.014328130 +0200 @@ -3637,14 +3637,14 @@ static const struct dmi_system_id dmi_pl {} }; -static struct rt5645_platform_data general_platform_data2 = { +static const struct rt5645_platform_data general_platform_data2 = { .dmic1_data_pin = RT5645_DMIC_DATA_IN2N, .dmic2_data_pin = RT5645_DMIC2_DISABLE, .jd_mode = 3, .inv_jd1_1 = true, }; -static struct dmi_system_id dmi_platform_asus_t100ha[] = { +static const struct dmi_system_id dmi_platform_asus_t100ha[] = { { .ident = "ASUS T100HAN", .matches = { @@ -3655,11 +3655,11 @@ static struct dmi_system_id dmi_platform { } }; -static struct rt5645_platform_data minix_z83_4_platform_data = { +static const struct rt5645_platform_data minix_z83_4_platform_data = { .jd_mode = 3, }; -static struct dmi_system_id dmi_platform_minix_z83_4[] = { +static const struct dmi_system_id dmi_platform_minix_z83_4[] = { { .ident = "MINIX Z83-4", .matches = { -- Jean Delvare SUSE L3 Support

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