July 2015 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH v3] ASoC: wm8960: update pll and clock setting function
by Zidan Wang 10 Aug '15

10 Aug '15

Add sysclk auto mode. When it's sysclk auto mode, if the MCLK is available for clock configure, using MCLK to provide sysclk directly, otherwise, search a available pll out frequcncy and set pll. Configure clock in hw_params may cause problems when using bypass style paths without hw_params in machine driver getting called. So add configure clock to set_bias_level. Signed-off-by: Zidan Wang <zidan.wang(a)freescale.com> --- sound/soc/codecs/wm8960.c | 217 +++++++++++++++++++++++++++++++++++++--------- sound/soc/codecs/wm8960.h | 1 + 2 files changed, 177 insertions(+), 41 deletions(-) diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c index 94c5c46..e9f8e41 100644 --- a/sound/soc/codecs/wm8960.c +++ b/sound/soc/codecs/wm8960.c @@ -48,6 +48,9 @@ #define WM8960_DISOP 0x40 #define WM8960_DRES_MASK 0x30 +static bool is_pll_freq_available(unsigned int source, unsigned int target); +static int wm8960_set_pll(struct snd_soc_codec *codec, + unsigned int freq_in, unsigned int freq_out); /* * wm8960 register cache * We can't read the WM8960 register space when we are @@ -126,9 +129,12 @@ struct wm8960_priv { struct snd_soc_dapm_widget *rout1; struct snd_soc_dapm_widget *out3; bool deemph; - int playback_fs; + int lrclk; int bclk; int sysclk; + int clk_id; + int freq_in; + bool is_stream_in_use[2]; struct wm8960_data pdata; }; @@ -164,8 +170,8 @@ static int wm8960_set_deemph(struct snd_soc_codec *codec) if (wm8960->deemph) { best = 1; for (i = 2; i < ARRAY_SIZE(deemph_settings); i++) { - if (abs(deemph_settings[i] - wm8960->playback_fs) < - abs(deemph_settings[best] - wm8960->playback_fs)) + if (abs(deemph_settings[i] - wm8960->lrclk) < + abs(deemph_settings[best] - wm8960->lrclk)) best = i; } @@ -565,6 +571,9 @@ static struct { { 8000, 5 }, }; +/* -1 for reserved value */ +static const int sysclk_divs[] = { 1, -1, 2, -1 }; + /* Multiply 256 for internal 256 div */ static const int dac_divs[] = { 256, 384, 512, 768, 1024, 1408, 1536 }; @@ -574,61 +583,109 @@ static const int bclk_divs[] = { 120, 160, 220, 240, 320, 320, 320 }; -static void wm8960_configure_clocking(struct snd_soc_codec *codec, - bool tx, int lrclk) +static int wm8960_configure_clocking(struct snd_soc_codec *codec) { struct wm8960_priv *wm8960 = snd_soc_codec_get_drvdata(codec); + int sysclk, bclk, lrclk, freq_out, freq_in; u16 iface1 = snd_soc_read(codec, WM8960_IFACE1); - u16 iface2 = snd_soc_read(codec, WM8960_IFACE2); - u32 sysclk; - int i, j; + int i, j, k; if (!(iface1 & (1<<6))) { dev_dbg(codec->dev, "Codec is slave mode, no need to configure clock\n"); - return; + return 0; + } + + if (wm8960->clk_id != WM8960_SYSCLK_MCLK && !wm8960->freq_in) { + dev_err(codec->dev, "No MCLK configured\n"); + return -EINVAL; } - if (!wm8960->sysclk) { - dev_dbg(codec->dev, "No SYSCLK configured\n"); - return; + freq_in = wm8960->freq_in; + bclk = wm8960->bclk; + lrclk = wm8960->lrclk; + /* + * If it's sysclk auto mode, check if the MCLK can provide sysclk or + * not. If MCLK can provide sysclk, using MCLK to provide sysclk + * directly. Otherwise, auto select a available pll out frequency + * and set PLL. + */ + if (wm8960->clk_id == WM8960_SYSCLK_AUTO) { + /* disable the PLL and using MCLK to provide sysclk */ + wm8960_set_pll(codec, 0, 0); + freq_out = freq_in; + } else if (wm8960->sysclk) { + freq_out = wm8960->sysclk; + } else { + dev_err(codec->dev, "No SYSCLK configured\n"); + return -EINVAL; } - if (!wm8960->bclk || !lrclk) { - dev_dbg(codec->dev, "No audio clocks configured\n"); - return; + /* check if the sysclk frequency is available. */ + for (i = 0; i < ARRAY_SIZE(sysclk_divs); ++i) { + if (sysclk_divs[i] == -1) + continue; + sysclk = freq_out / sysclk_divs[i]; + for (j = 0; j < ARRAY_SIZE(dac_divs); ++j) { + if (sysclk == dac_divs[j] * lrclk) { + for (k = 0; k < ARRAY_SIZE(bclk_divs); ++k) + if (sysclk == bclk * bclk_divs[k] / 10) + break; + if (k != ARRAY_SIZE(bclk_divs)) + break; + } + } + if (j != ARRAY_SIZE(dac_divs)) + break; } - for (i = 0; i < ARRAY_SIZE(dac_divs); ++i) { - if (wm8960->sysclk == lrclk * dac_divs[i]) { - for (j = 0; j < ARRAY_SIZE(bclk_divs); ++j) { - sysclk = wm8960->bclk * bclk_divs[j] / 10; - if (wm8960->sysclk == sysclk) + if (i != ARRAY_SIZE(sysclk_divs)) { + goto configure_clock; + } else if (wm8960->clk_id != WM8960_SYSCLK_AUTO) { + dev_err(codec->dev, "failed to configure clock\n"); + return -EINVAL; + } + /* get a available pll out frequency and set pll */ + for (i = 0; i < ARRAY_SIZE(sysclk_divs); ++i) { + if (sysclk_divs[i] == -1) + continue; + for (j = 0; j < ARRAY_SIZE(dac_divs); ++j) { + sysclk = lrclk * dac_divs[j]; + freq_out = sysclk * sysclk_divs[i]; + + for (k = 0; k < ARRAY_SIZE(bclk_divs); ++k) { + if (sysclk == bclk * bclk_divs[k] / 10 && + is_pll_freq_available(freq_in, freq_out)) { + wm8960_set_pll(codec, freq_in, freq_out); break; + } else { + continue; + } } - if(j != ARRAY_SIZE(bclk_divs)) + if (k != ARRAY_SIZE(bclk_divs)) break; } + if (j != ARRAY_SIZE(dac_divs)) + break; } - if (i == ARRAY_SIZE(dac_divs)) { - dev_err(codec->dev, "Unsupported sysclk %d\n", wm8960->sysclk); - return; + if (i == ARRAY_SIZE(sysclk_divs)) { + dev_err(codec->dev, "failed to configure clock\n"); + return -EINVAL; } - /* - * configure frame clock. If ADCLRC configure as GPIO pin, DACLRC - * pin is used as a frame clock for ADCs and DACs. - */ - if (iface2 & (1<<6)) - snd_soc_update_bits(codec, WM8960_CLOCK1, 0x7 << 3, i << 3); - else if (tx) - snd_soc_update_bits(codec, WM8960_CLOCK1, 0x7 << 3, i << 3); - else if (!tx) - snd_soc_update_bits(codec, WM8960_CLOCK1, 0x7 << 6, i << 6); +configure_clock: + /* configure sysclk clock */ + snd_soc_update_bits(codec, WM8960_CLOCK1, 3 << 1, i << 1); + + /* configure frame clock */ + snd_soc_update_bits(codec, WM8960_CLOCK1, 0x7 << 3, j << 3); + snd_soc_update_bits(codec, WM8960_CLOCK1, 0x7 << 6, j << 6); /* configure bit clock */ - snd_soc_update_bits(codec, WM8960_CLOCK2, 0xf, j); + snd_soc_update_bits(codec, WM8960_CLOCK2, 0xf, k); + + return 0; } static int wm8960_hw_params(struct snd_pcm_substream *substream, @@ -667,9 +724,9 @@ static int wm8960_hw_params(struct snd_pcm_substream *substream, return -EINVAL; } + wm8960->lrclk = params_rate(params); /* Update filters for the new rate */ - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - wm8960->playback_fs = params_rate(params); + if (tx) { wm8960_set_deemph(codec); } else { for (i = 0; i < ARRAY_SIZE(alc_rates); i++) @@ -682,7 +739,23 @@ static int wm8960_hw_params(struct snd_pcm_substream *substream, /* set iface */ snd_soc_write(codec, WM8960_IFACE1, iface); - wm8960_configure_clocking(codec, tx, params_rate(params)); + wm8960->is_stream_in_use[tx] = true; + + if (codec->dapm.bias_level == SND_SOC_BIAS_ON && + !wm8960->is_stream_in_use[!tx]) + return wm8960_configure_clocking(codec); + + return 0; +} + +static int wm8960_hw_free(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct wm8960_priv *wm8960 = snd_soc_codec_get_drvdata(codec); + bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + + wm8960->is_stream_in_use[tx] = false; return 0; } @@ -702,6 +775,7 @@ static int wm8960_set_bias_level_out3(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { struct wm8960_priv *wm8960 = snd_soc_codec_get_drvdata(codec); + u16 pm2 = snd_soc_read(codec, WM8960_POWER2); int ret; switch (level) { @@ -711,6 +785,9 @@ static int wm8960_set_bias_level_out3(struct snd_soc_codec *codec, case SND_SOC_BIAS_PREPARE: switch (snd_soc_codec_get_bias_level(codec)) { case SND_SOC_BIAS_STANDBY: + ret = wm8960_configure_clocking(codec); + if (ret) + return ret; if (!IS_ERR(wm8960->mclk)) { ret = clk_prepare_enable(wm8960->mclk); if (ret) { @@ -726,6 +803,13 @@ static int wm8960_set_bias_level_out3(struct snd_soc_codec *codec, break; case SND_SOC_BIAS_ON: + /* + * If it's sysclk auto mode, and the pll is enabled, + * disable the pll + */ + if (wm8960->clk_id == WM8960_SYSCLK_AUTO && (pm2 & 0x1)) + wm8960_set_pll(codec, 0, 0); + if (!IS_ERR(wm8960->mclk)) clk_disable_unprepare(wm8960->mclk); break; @@ -780,6 +864,7 @@ static int wm8960_set_bias_level_capless(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { struct wm8960_priv *wm8960 = snd_soc_codec_get_drvdata(codec); + u16 pm2 = snd_soc_read(codec, WM8960_POWER2); int reg, ret; switch (level) { @@ -822,6 +907,10 @@ static int wm8960_set_bias_level_capless(struct snd_soc_codec *codec, msleep(100); + ret = wm8960_configure_clocking(codec); + if (ret) + return ret; + if (!IS_ERR(wm8960->mclk)) { ret = clk_prepare_enable(wm8960->mclk); if (ret) { @@ -834,6 +923,13 @@ static int wm8960_set_bias_level_capless(struct snd_soc_codec *codec, break; case SND_SOC_BIAS_ON: + /* + * If it's sysclk auto mode, and the pll is enabled, + * disable the pll + */ + if (wm8960->clk_id == WM8960_SYSCLK_AUTO && (pm2 & 0x1)) + wm8960_set_pll(codec, 0, 0); + if (!IS_ERR(wm8960->mclk)) clk_disable_unprepare(wm8960->mclk); @@ -892,6 +988,28 @@ struct _pll_div { u32 k:24; }; +static bool is_pll_freq_available(unsigned int source, unsigned int target) +{ + unsigned int Ndiv; + + if (source == 0 || target == 0) + return false; + + /* Scale up target to PLL operating frequency */ + target *= 4; + Ndiv = target / source; + + if (Ndiv < 6) { + source >>= 1; + Ndiv = target / source; + } + + if ((Ndiv < 6) || (Ndiv > 12)) + return false; + + return true; +} + /* The size in bits of the pll divide multiplied by 10 * to allow rounding later */ #define FIXED_PLL_SIZE ((1 << 24) * 10) @@ -943,10 +1061,9 @@ static int pll_factors(unsigned int source, unsigned int target, return 0; } -static int wm8960_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id, - int source, unsigned int freq_in, unsigned int freq_out) +static int wm8960_set_pll(struct snd_soc_codec *codec, + unsigned int freq_in, unsigned int freq_out) { - struct snd_soc_codec *codec = codec_dai->codec; u16 reg; static struct _pll_div pll_div; int ret; @@ -986,6 +1103,20 @@ static int wm8960_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id, return 0; } +static int wm8960_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id, + int source, unsigned int freq_in, unsigned int freq_out) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct wm8960_priv *wm8960 = snd_soc_codec_get_drvdata(codec); + + wm8960->freq_in = freq_in; + + if (pll_id == WM8960_SYSCLK_AUTO) + return 0; + + return wm8960_set_pll(codec, freq_in, freq_out); +} + static int wm8960_set_dai_clkdiv(struct snd_soc_dai *codec_dai, int div_id, int div) { @@ -1043,11 +1174,14 @@ static int wm8960_set_dai_sysclk(struct snd_soc_dai *dai, int clk_id, snd_soc_update_bits(codec, WM8960_CLOCK1, 0x1, WM8960_SYSCLK_PLL); break; + case WM8960_SYSCLK_AUTO: + break; default: return -EINVAL; } wm8960->sysclk = freq; + wm8960->clk_id = clk_id; return 0; } @@ -1060,6 +1194,7 @@ static int wm8960_set_dai_sysclk(struct snd_soc_dai *dai, int clk_id, static const struct snd_soc_dai_ops wm8960_dai_ops = { .hw_params = wm8960_hw_params, + .hw_free = wm8960_hw_free, .digital_mute = wm8960_mute, .set_fmt = wm8960_set_dai_fmt, .set_clkdiv = wm8960_set_dai_clkdiv, diff --git a/sound/soc/codecs/wm8960.h b/sound/soc/codecs/wm8960.h index 2d8163d..ab3220d 100644 --- a/sound/soc/codecs/wm8960.h +++ b/sound/soc/codecs/wm8960.h @@ -82,6 +82,7 @@ #define WM8960_SYSCLK_MCLK (0 << 0) #define WM8960_SYSCLK_PLL (1 << 0) +#define WM8960_SYSCLK_AUTO (2 << 0) #define WM8960_DAC_DIV_1 (0 << 3) #define WM8960_DAC_DIV_1_5 (1 << 3) -- 1.9.1

2 5

[alsa-devel] RME MADIFX mainline inclusion
by Adrian Knoth 10 Aug '15

10 Aug '15

Hi Takashi! Back in the days (late 2012 I guess), I wrote the MADIFX driver based on the existing RME hdspm.c. I never had physical access to this 1399 Euro audio interface, only the OSX driver and a shell login to one of IOSONO's machines which temporarily featured such a card. I know there are other users like University of Washington (DXARTS) and probably some more (the usual WFS suspects). Long story short: I don't have such a card and since I no longer work in Linux audio, I guess I won't have access in the foreseeable future. That said, I think the driver should be part of the mainline kernel, so it gets all the love the other drivers get (static code analysis, API changes etc.) I think I've ported everything you guys have done to hdspm: I've read the commitlog for hdspm and applied those changes to the madifx driver. Code is here: https://github.com/adiknoth/madifx/commits/tokernel Status: * PCM playback/capture working (SS and DS tested, QS untested) * MIDI working * All card settings working (e.g. TX64, SMUX, AESpro, WC-Term, WC-singlespeed...) * Slave Mode/External clock selection working * Mirror-MADI1-to-Out2+3 maybe working (untested) * Redundancy mode maybe working (untested) * ioctls implemented (see ioctl.c) * Static mixer working (fixed 1:1 mapping) * DSP NOT working. RME doesn't intend to release any information regarding the DSP. * Adjustable mixer NOT working (needs new userspace tools) * Levelmetering NOT working (maybe wrong, needs new userspace tools) I have no idea who could write the new mixer code. Apparently, demand is low, maybe zero. checkpatch reports four warnings of which one is a false-positive, the other one doesn't make sense and only two are real but somewhat ugly to fix (line over 80 with really long array indices). In its current state, can I send you three patches (Kconfig, madifx.h, madifx.c) that add said driver to sound/pci/rme9652? Cheers

6 16

[alsa-devel] DP MST audio support
by Dave Airlie 10 Aug '15

10 Aug '15

Hi, So I have a branch that makes no sound, http://cgit.freedesktop.org/~airlied/linux/log/?h=dp-mst-audio and I'm not sure where I need to turn to next, The Intel docs I've read are kinda vague, assuming you know lots of things I clearly don't. so in theory my branch, sets up the SDP stream to the monitor in the payload creation, enables the codec in the intel GPU driver, and passes the ELD to the audio driver. The audio driver uses the device list to get the presence/valid bits per device, and manages to retrieve the ELD. I even create ELD files in /proc/asound/HDMI/ that have sensible values in them So it looks like I'm just missing some routing somewhere, most likely in the audio driver, then again I could be missing a lot more than that. Just looking for any ideas or knowledge people may have locked in their brains or inside their firewalls. Dave.

4 6

[alsa-devel] [PATCH 0/4] Add SPDIF support for rockchip
by Sjoerd Simons 07 Aug '15

07 Aug '15

This patchset adds support for the Rockchip SPDIF transceiver as present on RK3066, RK3188 and RK3288 boards and enables it on a Radxa rock pro. Tested on a Radxa Rock Pro board. Sjoerd Simons (4): ASoC: dt-bindings: add rockchip tranceiver bindings ASoc: rockchip: Add rockchip SPDIF transceiver driver ARM: dts: rockchip: Add SPDIF transceiver for RK3188 ARM: dts: rockchip: Add SPDIF optical out on Radxa Rock .../devicetree/bindings/sound/rockchip-spdif.txt | 41 +++ arch/arm/boot/dts/rk3188-radxarock.dts | 19 ++ arch/arm/boot/dts/rk3188.dtsi | 22 ++ sound/soc/rockchip/Kconfig | 9 + sound/soc/rockchip/Makefile | 3 + sound/soc/rockchip/rockchip_spdif.c | 375 +++++++++++++++++++++ sound/soc/rockchip/rockchip_spdif.h | 63 ++++ 7 files changed, 532 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/rockchip-spdif.txt create mode 100644 sound/soc/rockchip/rockchip_spdif.c create mode 100644 sound/soc/rockchip/rockchip_spdif.h -- 2.4.6

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[alsa-devel] [PATCH] ASoC: wm8741: Simplify wm8741_set_dai_sysclk implementation
by Axel Lin 06 Aug '15

06 Aug '15

Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/codecs/wm8741.c | 41 ++++++++++++++--------------------------- 1 file changed, 14 insertions(+), 27 deletions(-) diff --git a/sound/soc/codecs/wm8741.c b/sound/soc/codecs/wm8741.c index 430fa7d..9723e44 100644 --- a/sound/soc/codecs/wm8741.c +++ b/sound/soc/codecs/wm8741.c @@ -278,51 +278,38 @@ static int wm8741_set_dai_sysclk(struct snd_soc_dai *codec_dai, switch (freq) { case 0: wm8741->sysclk_constraints = NULL; - wm8741->sysclk = freq; - return 0; - + break; case 11289600: wm8741->sysclk_constraints = &constraints_11289; - wm8741->sysclk = freq; - return 0; - + break; case 12288000: wm8741->sysclk_constraints = &constraints_12288; - wm8741->sysclk = freq; - return 0; - + break; case 16384000: wm8741->sysclk_constraints = &constraints_16384; - wm8741->sysclk = freq; - return 0; - + break; case 16934400: wm8741->sysclk_constraints = &constraints_16934; - wm8741->sysclk = freq; - return 0; - + break; case 18432000: wm8741->sysclk_constraints = &constraints_18432; - wm8741->sysclk = freq; - return 0; - + break; case 22579200: case 33868800: wm8741->sysclk_constraints = &constraints_22579; - wm8741->sysclk = freq; - return 0; - + break; case 24576000: wm8741->sysclk_constraints = &constraints_24576; - wm8741->sysclk = freq; - return 0; - + break; case 36864000: wm8741->sysclk_constraints = &constraints_36864; - wm8741->sysclk = freq; - return 0; + break; + default: + return -EINVAL; } - return -EINVAL; + + wm8741->sysclk = freq; + return 0; } static int wm8741_set_dai_fmt(struct snd_soc_dai *codec_dai, -- 2.1.0

3 2

[alsa-devel] query on upstream ASoC driver split across SOUND and DRM subsystems
by Bayyavarapu, Maruthi 06 Aug '15

06 Aug '15

Hi, I want to send AMD ASoC driver for review. This driver has a dependency with AMD's 'DRM' subsystem components, which are not upstreamed yet. Please suggest, whether this ASoC driver can be upstreamed via 'DRM' tree or any alternate approach is preferred. The link to ASoC driver is : http://cgit.freedesktop.org/~agd5f/linux/commit/?h=acp-wip&id=2b67a44150988… ASoC commits and dependencies are currently in 'acp-wip' branch of : git://people.freedesktop.org/~agd5f/linux Thanks, Maruthi

3 2

Re: [alsa-devel] When select 2.1 profile in pulseaudio, how do pulseaudio select notebook 's internal subwoofer or 5.1 external speaker ?
by Raymond Yau 05 Aug '15

05 Aug '15

> > > > Do pulseaudio 2.1 profile select the internal subwoofer or your 5.1 > > external speaker ? > > My external speaker (i.e not subwoofer but center). It always was so. > https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/commit/sound/p… +/* default channel maps for 2.1 speakers; + * since HD-audio supports only stereo, odd number channels are omitted + */ +const struct snd_pcm_chmap_elem snd_pcm_2_1_chmaps[] = { + { .channels = 2, + .map = { SNDRV_CHMAP_FL, SNDRV_CHMAP_FR } }, + { .channels = 4, + .map = { SNDRV_CHMAP_FL, SNDRV_CHMAP_FR, + SNDRV_CHMAP_LFE, SNDRV_CHMAP_LFE } }, + { } +}; The default 2.1 channel map for internal subwoofer does not has 5.1 channel map But your notebook have three jacks at ext front which support 5.1 In your case,there is no default 2.1 channel map when the current model has three speaker pins after remove the not exist internal surround speakers, [ALC882_FIXUP_ACER_ASPIRE_4930G] = { - .type = HDA_FIXUP_PINS, - .v.pins = (const struct hda_pintbl[]) { - { 0x16, 0x99130111 }, /* CLFE speaker */ - { 0x17, 0x99130112 }, /* surround speaker */ - { } - }, you need to use the left channel version of 2.1 channel and 5.1 channel map +static const struct snd_pcm_chmap_elem acer_pcm_chmaps[] = { + { .channels = 2, + .map = { SNDRV_CHMAP_FL, SNDRV_CHMAP_FR } }, + { .channels = 4, + .map = { SNDRV_CHMAP_FL, SNDRV_CHMAP_FR, + SNDRV_CHMAP_LFE, SNDRV_CHMAP_NA } }, /* LFE only on left */ + { .channels = 4, + .map = { SNDRV_CHMAP_FL, SNDRV_CHMAP_FR, + SNDRV_CHMAP_RL, SNDRV_CHMAP_RR } }, + { .channels = 6, + .map = { SNDRV_CHMAP_FL, SNDRV_CHMAP_FR, + SNDRV_CHMAP_RL, SNDRV_CHMAP_RR, + SNDRV_CHMAP_FC, SNDRV_CHMAP_LFE } }, + { } +}; http://git.alsa-project.org/?p=alsa-lib.git;a=commitdiff;h=48f1b308cc66152e… <http://git.alsa-project.org/?p=alsa-lib.git;a=commitdiff;h=48f1b308cc66152e…> conf: Allow 2.1 surround to use different number of channels This way, cards that support LFE on four channels (e g laptop with internal subwoofer) can do that, and other cards on a six channel setup can use that as well. which channel map is used by alsa lib when you open surround21 pcm device since your notebook will have both channel maps ?

1 1

[alsa-devel] [PATCH v14 0/3] ASoC: tda998x: add a codec to the HDMI transmitter
by Jean-Francois Moine 03 Aug '15

03 Aug '15

This patch series asks for Russell King's patch http://article.gmane.org/gmane.linux.ports.arm.kernel/411574 "drm/edid: add function to help find SADs" to be applied. v14: (Mark Brown) - protect the CODEC from edid change - change some structure names - remove the static pointer to the HDMI driver functions - get the ELD from a pointer in the device private area - prevent audio from streaming many sources at the same time - build the CODEC DAI's from two different samples - use EXPORT_SYMBOL_GPL for exported functions - simplify exchanges between audio and video by using common data in the device private area v13: - rebase on 4.2.0-rc3 - remove Russell's patches v12: - use Russell King's DRM ELD helper (Mark Brown) v11: - reduce the patch series to adding the tda998x codec only v10: - add the generic dt-card - define the audio ports from a DT graph of ports (Russell King) - reuse HDMI constants (Andrew Jackson - Jyri Sarha) - alloc rate_constraints in codec (Jyri Sarha) - fix bad number of channels (Jyri Sarha) - correct codec generation from config (Russell King - Jyri Sarha) - no module init/exit (Russell King) v9: - back to a TDA998x specific CODEC - more comments - change magic values to constants v8: - change some comments about the patches v7: - remove the change of the K predivider (Jyri Sarha) - add S24_3LE and S32_LE as possible audio formats (Jyri Sarha) - don't move the struct priv2 definition and use the slave encoder private data as the device private data (Russell King) - remove the useless request_module (Russell King/Mark Brown) - don't lock the HDMI module (Russell King) - use platform_device_unregister to remove the codec (Russell King) v6: - extend the HDMI CODEC instead of using a specific CODEC v5: - use the TDA998x private data instead of a specific area for the CODEC interface - the CODEC is TDA998x specific (Mark Brown) v4: - remove all the TDA998x specific stuff from the CODEC - move the EDID scan from the CODEC to the TDA998x - move the CODEC to sound/soc (Mark Brown) - update the audio_sample_rate from the EDID (Andrew Jackson) v3: fix bad rate (Andrew Jackson) v2: check double stream start (Mark Brown) Jean-Francois Moine (3): drm/i2c: tda998x: Add support of a DT graph of ports drm/i2c: tda998x: Change drvdata for audio extension ASoC: tda998x: add a codec to the HDMI transmitter .../devicetree/bindings/drm/i2c/tda998x.txt | 51 +++++++ drivers/gpu/drm/i2c/tda998x_drv.c | 163 +++++++++++++++++++-- include/sound/tda998x.h | 18 +++ sound/soc/codecs/Kconfig | 6 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/tda998x.c | 141 ++++++++++++++++++ 6 files changed, 370 insertions(+), 11 deletions(-) create mode 100644 include/sound/tda998x.h create mode 100644 sound/soc/codecs/tda998x.c -- 2.4.6

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[alsa-devel] Beaglebone black : Audio cap & Linux version 4.1.3-ti-r6.2
by Micka 03 Aug '15

03 Aug '15

HI everyone, I'm trying to test the sound output with the driver tlv320aic3104. I'm stuck somewhere and I can't figure it out ! I Put a lot of printk and I don't know where to put other printk. Do you know who call the function soc_pcm_prepare ? 693 <http://lxr.free-electrons.com/source/sound/soc/soc-pcm.c#L693> */**694 <http://lxr.free-electrons.com/source/sound/soc/soc-pcm.c#L694> * * Called by ALSA when the PCM substream is prepared, can set format, sample*695 <http://lxr.free-electrons.com/source/sound/soc/soc-pcm.c#L695> * * rate, etc. This function is non atomic and can be called multiple times,*696 <http://lxr.free-electrons.com/source/sound/soc/soc-pcm.c#L696> * * it can refer to the runtime info.*697 <http://lxr.free-electrons.com/source/sound/soc/soc-pcm.c#L697> * */*698 <http://lxr.free-electrons.com/source/sound/soc/soc-pcm.c#L698> static int soc_pcm_prepare <http://lxr.free-electrons.com/ident?i=soc_pcm_prepare>(struct snd_pcm_substream <http://lxr.free-electrons.com/ident?i=snd_pcm_substream> *substream) Looks like that it's ALSA, but I can't find where ? I tried to find a line in the kernel calling the function: rtd->ops <http://lxr.free-electrons.com/ident?i=ops>.prepare(.... ) can't find it .... my trace : [ 550.269286] ASoC: tlv320aic3x-hifi <-> 48038000.mcasp info: [ 550.269363] ASoC: rate mask 0x7fe [ 550.269398] ASoC: min ch 2 max ch 2 [ 550.269431] ASoC: min rate 8000 max rate 96000 [ 550.275809] tlv aic3x_hw_params [ 550.275862] snd_soc_read: snd soc component read: ret 0 reg 9 val 64 [ 550.275934] snd_soc_component_write reg 9 val 64 [ 550.275987] snd_soc_component_write reg 101 val 0 [ 550.276030] snd_soc_component_write reg 7 val 10 [ 550.276063] snd_soc_component_write reg 2 val 0 [ 550.289260] snd_soc_component_write reg 11 val 1 [ 550.289339] snd_soc_component_write reg 4 val 32 [ 550.289375] snd_soc_component_write reg 5 val 30 [ 550.289407] snd_soc_component_write reg 6 val 0 [ 550.289952] soc_pcm_prepare [ 550.290003] snd_soc_component_write reg 10 val 0 [ 550.290043] soc_dapm_stream_event 1 [ 550.290107] soc_dapm_stream_event: dapm_power_widgets:start dapm_power_widgets 1 [ 550.290150] trace_snd_soc_dapm_start [ 550.290176] dapm reset [ 550.290226] trace_snd_soc_dapm_widget_power [ 550.290257] trace_snd_soc_dapm_widget_power [ 550.290284] trace_snd_soc_dapm_widget_power [ 550.290311] trace_snd_soc_dapm_widget_power [ 550.290339] trace_snd_soc_dapm_widget_power [ 550.290370] trace_snd_soc_dapm_widget_power [ 550.290395] trace_snd_soc_dapm_widget_power [ 550.290423] trace_snd_soc_dapm_widget_power [ 550.290448] trace_snd_soc_dapm_widget_power [ 550.290477] trace_snd_soc_dapm_widget_power [ 550.290503] trace_snd_soc_dapm_widget_power [ 550.290531] trace_snd_soc_dapm_widget_power [ 550.290557] trace_snd_soc_dapm_widget_power [ 550.290582] trace_snd_soc_dapm_widget_power [ 550.290608] trace_snd_soc_dapm_widget_power [ 550.290634] trace_snd_soc_dapm_widget_power [ 550.290659] trace_snd_soc_dapm_widget_power [ 550.290686] trace_snd_soc_dapm_widget_power [ 550.290712] trace_snd_soc_dapm_widget_power [ 550.290738] trace_snd_soc_dapm_widget_power [ 550.290765] trace_snd_soc_dapm_widget_power [ 550.290791] trace_snd_soc_dapm_widget_power [ 550.290817] trace_snd_soc_dapm_widget_power [ 550.290842] trace_snd_soc_dapm_widget_power [ 550.290879] dap set bias to off [ 550.290910] dapm_pre_sequence_async off [ 550.290937] snd soc dapm set bias level 1 [ 550.290962] trace_snd_soc_bias_level_start 1 [ 550.290988] trace_snd_soc_bias_level_done 1 [ 550.291011] dapm_pre_sequence_async onsnd soc dapm set bias level 2 [ 550.291047] trace_snd_soc_bias_level_start 2 [ 550.291073] trace_snd_soc_bias_level_done 2 [ 550.291318] dapm_pre_sequence_async off [ 550.291354] snd soc dapm set bias level 1 [ 550.291381] trace_snd_soc_bias_level_start 1 [ 550.291405] soc set bias level [ 550.291432] tlv set bias level 1 [ 550.291456] tlv standby [ 550.291480] tlv set power [ 550.300504] snd_soc_read: snd soc component read: ret 0 reg 5 val 30 [ 550.300583] snd_soc_read: snd soc component read: ret 0 reg 6 val 0 [ 550.300631] snd_soc_component_write reg 5 val 30 [ 550.301103] snd_soc_component_write reg 6 val 0 [ 550.301577] trace_snd_soc_bias_level_done 1 [ 550.301611] dapm_pre_sequence_async onsnd soc dapm set bias level 2 [ 550.301650] trace_snd_soc_bias_level_start 2 [ 550.301675] soc set bias level [ 550.301701] tlv set bias level 2 [ 550.301724] tlv prepare [ 550.301757] trace_snd_soc_bias_level_done 2 [ 550.301864] dapm event WILL_PMU [ 550.301897] dapm event WILL_PMU [ 550.301923] dapm event WILL_PMU [ 550.301948] dapm event WILL_PMU [ 550.301972] dapm event WILL_PMU [ 550.301997] dapm event WILL_PMU [ 550.302021] dapm event WILL_PMU [ 550.302045] dapm event WILL_PMU [ 550.302069] dapm event WILL_PMU [ 550.302094] dapm event WILL_PMU [ 550.302119] dapm event WILL_PMU [ 550.302143] dapm event WILL_PMU [ 550.302168] dapm event WILL_PMU [ 550.302192] dapm event WILL_PMU [ 550.302216] dapm event WILL_PMU [ 550.302240] dapm event WILL_PMU [ 550.302265] dapm event WILL_PMU [ 550.302289] dapm event WILL_PMU [ 550.302313] dapm event WILL_PMU [ 550.302337] dapm event WILL_PMU [ 550.302362] dapm event WILL_PMU [ 550.302386] dapm event WILL_PMU [ 550.302410] dapm event WILL_PMU [ 550.302435] dapm event WILL_PMU [ 550.302460] dapm sq run 1 [ 550.302489] dapm sq run 1 [ 550.302523] dapm event PRE_PMU [ 550.302550] dapm event PRE_PMD [ 550.302575] dapm event PRE_PMU [ 550.302599] dapm event PRE_PMD [ 550.302623] dapm event PRE_PMU [ 550.302647] dapm event PRE_PMD [ 550.302671] dapm event PRE_PMU [ 550.302696] dapm event PRE_PMD [ 550.302720] dapm event PRE_PMU [ 550.302744] dapm event PRE_PMD [ 550.302769] dapm event PRE_PMU [ 550.302793] dapm event PRE_PMD [ 550.302817] dapm event POST_PMU [ 550.302842] dapm event POST_PMD [ 550.302867] dapm event POST_PMU [ 550.302891] dapm event POST_PMD [ 550.302915] dapm event POST_PMU [ 550.302939] dapm event POST_PMD [ 550.302964] dapm event POST_PMU [ 550.302988] dapm event POST_PMD [ 550.303012] dapm event POST_PMU [ 550.303036] dapm event POST_PMD [ 550.303060] dapm event POST_PMU [ 550.303084] dapm event POST_PMD [ 550.303110] dapm event PRE_PMU [ 550.303134] dapm event PRE_PMD [ 550.303159] dapm event POST_PMU [ 550.303183] dapm event POST_PMD [ 550.303209] dapm event PRE_PMU [ 550.303234] dapm event PRE_PMD [ 550.303258] dapm event PRE_PMU [ 550.303282] dapm event PRE_PMD [ 550.303306] dapm event PRE_PMU [ 550.303331] dapm event PRE_PMD [ 550.303355] dapm event PRE_PMU [ 550.303379] dapm event PRE_PMD [ 550.303403] dapm event POST_PMU [ 550.303427] dapm event POST_PMD [ 550.303451] dapm event POST_PMU [ 550.303475] dapm event POST_PMD [ 550.303499] dapm event POST_PMU [ 550.303524] dapm event POST_PMD [ 550.303548] dapm event POST_PMU [ 550.303572] dapm event POST_PMD [ 550.303598] dapm event PRE_PMU [ 550.303622] dapm event PRE_PMD [ 550.303647] dapm event PRE_PMU [ 550.303671] dapm event PRE_PMD [ 550.304162] dapm event POST_PMU [ 550.304199] dapm event POST_PMD [ 550.304226] dapm event POST_PMU [ 550.304250] dapm event POST_PMD [ 550.304278] dapm event PRE_PMU [ 550.304303] dapm event PRE_PMD [ 550.304328] dapm event PRE_PMU [ 550.304352] dapm event PRE_PMD [ 550.304377] dapm event PRE_PMU [ 550.304401] dapm event PRE_PMD [ 550.304426] dapm event PRE_PMU [ 550.304450] dapm event PRE_PMD [ 550.304474] dapm event POST_PMU [ 550.304499] dapm event POST_PMD [ 550.304523] dapm event POST_PMU [ 550.304547] dapm event POST_PMD [ 550.304572] dapm event POST_PMU [ 550.304596] dapm event POST_PMD [ 550.304620] dapm event POST_PMU [ 550.304644] dapm event POST_PMD [ 550.304670] dapm event PRE_PMU [ 550.304695] dapm event PRE_PMD [ 550.305156] dapm event POST_PMU [ 550.305191] dapm event POST_PMD [ 550.305219] dapm event PRE_PMU [ 550.305245] dapm event PRE_PMD [ 550.305696] dapm event POST_PMU [ 550.305730] dapm event POST_PMD [ 550.305757] dapm event PRE_PMU [ 550.305782] dapm event PRE_PMD [ 550.306275] dapm event POST_PMU [ 550.306312] dapm event POST_PMD [ 550.306341] dapm event PRE_PMU [ 550.306366] dapm event PRE_PMD [ 550.306849] dapm event POST_PMU [ 550.306885] dapm event POST_PMD [ 550.306914] dapm event PRE_PMU [ 550.306939] dapm event PRE_PMD [ 550.317926] dapm event POST_PMU [ 550.317958] dapm event POST_PMD [ 550.317977] dapm event PRE_PMU [ 550.317991] dapm event PRE_PMD [ 550.318401] dapm event POST_PMU [ 550.318422] dapm event POST_PMD [ 550.318439] dapm event PRE_PMU [ 550.318454] dapm event PRE_PMD [ 550.318467] dapm event POST_PMU [ 550.318480] dapm event POST_PMD [ 550.318496] dapm_power_widgets run all the bias changes in parrallel [ 550.321470] dapm_post_sequence_async prepare 1 [ 550.321498] snd soc dapm set bias level 3 [ 550.321513] trace_snd_soc_bias_level_start 3 [ 550.321527] soc set bias level [ 550.321543] tlv set bias level 3 [ 550.321555] tlv bias on [ 550.321570] trace_snd_soc_bias_level_done 3 [ 550.321649] dapm_power_widgets dapm_power_widgets [ 550.321666] DAPM sequencing finished [ 550.321680] trace_snd_soc_dapm_done [ 550.321695] pcm prepare call snd_soc_dai_digital_mute [ 550.321711] snd_soc_dai_digital_mute 0 0 [ 550.321725] tlv aic3x_mute [ 550.321739] snd_soc_read: snd soc component read: ret 0 reg 43 val 168 [ 550.321772] snd_soc_read: snd soc component read: ret 0 reg 44 val 168 [ 550.321798] snd_soc_component_write reg 43 val 40 [ 550.322224] snd_soc_component_write reg 44 val 40 [ 550.322630] pcm prepare call snd_soc_dai_digital_mute for cpu_dai [ 550.322651] snd_soc_dai_digital_mute 0 0 > >

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[alsa-devel] [PATCH] ALSA: hda - Fix a wrong busy check in alt PCM open
by Takashi Iwai 03 Aug '15

03 Aug '15

Currently, the alt PCM open callback returns -EBUSY when an independent HP is turned off, supposing that it conflicts with the main PCM. However, obviously, this check is wrong when the independent HP itself isn't enabled but the alt PCM was explicitly created via alc_dac_nid by a codec driver. Reported-and-tested-by: Kailang Yang <kailang(a)realtek.com> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_generic.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index ac0db1679f09..b077bb644434 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -5175,7 +5175,7 @@ static int alt_playback_pcm_open(struct hda_pcm_stream *hinfo, int err = 0; mutex_lock(&spec->pcm_mutex); - if (!spec->indep_hp_enabled) + if (spec->indep_hp && !spec->indep_hp_enabled) err = -EBUSY; else spec->active_streams |= 1 << STREAM_INDEP_HP; -- 2.4.5

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